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@@TJ-hs1qm agreed, although just a heads up if you're not aware that may help, you can insert melodyne on the audio channel you want to automate the pitch of, then add pitch as an envelope and draw your automation, becomes quick once used to it. be aware you need to add it as an insert, rather than the 'edit with melodyne' auto option in order for the automatable pitch envelope to be available
Most plugins are prefader. So it's important to use the mixtool, gain tool, etc to control gain that goes into plugins. The fader only changes the output volume. It doesn't affect plugins
Great point. On my last project I was hearing some noise that I traced back to overloading an amp sim. There was a compressor and the fader after that keeping the final volume okay but it was clipping in the amp plugin.
The Mixtool is nice in a demonstration like this for visual representation. But don't forget that it's much easier to use the trim knob at the top of the channel for any necessary gain adjustments. No need to add plugins for this job. Before I start a mix, I always get my input levels in the same ballpark using the trim knob. Makes keeping levels under control way easier. One of the main reasons I switched to Studio One. Great feature.
What I learnt, is use Hornet VU Mk4 or LU MK2 (both of them combined costs around USD 5 only). Honestly, the easiest workflow ever. Put VU MK4 as the first plugin and set it to -18dBFS RMS values so that VU MK4 will make sure the signal coming into VU will not leave exceeding -18dBFS RMS so that you can use your analog modeled plugin with ease. Now, I put LU MK2 as the last plugin and I set it to -23 LUFS (for percussive sounds Momentary and for other sounds Short term). So now all of my channel are at equal loudness. Then I use fader and do a anchor style mixing.
(Re-posted from a reply to a comment below for public amusement...) Welp, I learned Gain Staging in the late 80s. Both in the Studio and in Live Sound. Not sure when the fancy label came about, but it was long before TH-cam. I suspect, as with most things in Audio, the term comes from the broader Electronics field. The capacitors, resistors, transistors, transformers, tubes, diodes, integrated circuits, discrete operational amplifiers, etc. don’t really care what signal is being passed through them, or whether or not when patched into a transducer at the end of a long series of components, it generates a “sound” that subjectively sounds “good” to any one human ear, or billions across generations. Each component simply says “If you Don’t put this into me, I won’t do anything. If you put this to this in, I’ll put this out. However, if you put This in, I’ll blow up!” All those metal boxes, faraday cages really, whether shaped like a desktop to be sat at, or maybe a standardized 19” width of various unit height to be racked together with other metal boxes, they all had (have) some cocktail or electronic wizardry inside. But luckily, the electronic engineers who concocted these works of art to help other artists produce their own art, thought “ hey, maybe the end users will want to “interconnect” two or two-hundred of these boxes together, in unpredictable ways. And we already know that every component, in every circuit, in every box has a ‘not-enough, too-much, just-right’ pain threshold. So we better train these guys how to do that, as well as give them some basic information about each box, and give em a bunch of different ‘controls’ and relevant ‘meters’ on the outside of the box, so they don’t get themselves into too much trouble. We know these music guys are hopeless, but the broadcast and live sound guys seem to really want to learn this stuff. If they do, we electronic engineers should call them audio engineers.” Bill Putnam, Colin Sanders, Saul Walker, David Dearden, Paul Wolf and the King of Kings himself, now 94-years young, Sir Rupert Neve, all spent their lives on a quest, in search of “The Perfect Circuit”. Billions of records, and broadcasts, and live shows have passed through circuits that were the product of their productivity. Whilst the code-writers have done some amazing things, 64-bit floating point math being one of them, there might be a reason, other than nostalgia, that every major code-for-audio company is desperately trying to reproduce in code, all of the sonic qualities of analog circuits. And hint, the unlimited dynamic range, whilst useful in the background, has little to do with the price of tea in China. Two other observations: (I really am trying not to go too deep) A) The two main tools of the audio engineer remain: EQ & Dynamics (specifically Compressors). Both are used ultimately to control, and primarily REDUCE...Dynamic Range. A compressors government name is ‘Dynamic Range Compressor’. A limiter puts a ceiling on how high a signal can go. Raise the fader all you want, and your signal will either get a haircut or get beheaded. Gates and Expanders work on the bottom end of a signals Dynamic Range, usually to cut it off at the ankles.... or knees. EQs, formerly and formally known as Equalizers, seem to only exist within the Frequency arena of a signal, but no... EQs are Static Dynamic Range controllers, divided into 2 or more frequency bands. Imagine a 1/3-octave Graphic Equalizer: on it you will find 31 little tiny faders, each of which does the same thing as the 100mm faders on a mixing console, except they only boast or cut a specific frequency band within the signal they receive, regardless of that signals input level, aka static. There’s no minimum threshold. In “Subtractive EQing” you run a signal through the machine, whichever frequencies you don’t want, cut em, anything popping out, tuck it back in. You whittle and carve away at whatever walked through the door, trimming the fat, nipping and tucking like either a butcher or a cosmetic surgeon. In the end, she will hopefully be a more beautiful version of herself (disclaimer: beauty is in the ear of the beholder), and when she’s back out on the street, all will be stunned and none will be the wiser. Or....You can go nuts and while she’s unconscious on the operating table, aka mixing console, start injecting her with 1000-CCs of this, and two-pounds of that, you can use those scalpels to open her up and start Implanting foreign objects into her body with no regard for her natural beauty or health or what she looked like before your hands all over her. Now she’s Jessica Rabbit, and trying get her fit in the mix is like trying to walk into a party with her in a hot red mini-skirt without being seen, or in this case heard. Whether you’re producing a feature film with an ensemble cast of glamorous Hollywood stars living their best selves, in the best wardrobe, hair, and makeup money can buy, or you’re making a low budget porno movie with impossibly endowed actors that can’t act (I could go on), you’re still going to package this hot mess into a box of some sort, and that medium of consumption will have dynamic range limitations. Whatever sounds you have coming out of your speakers, the loud sounds are gonna have a relationship with the quiet sounds and with the silence. Ultimately, the Dynamic Range of your Mix, turns out is a big part of the quality of that mix, both Objectively now measured in LUFS, and Subjectively now measured in Downloads. B) It’s ironic to me that the same people excited about unlimited dynamic range, are the same ones most likely to be on the enemy side of the loudness wars! They seem to want to both put in sounds with +/-96dB of epileptic convulsions, and THEN run limiters at full throttle until their waveforms look like their DAWs are literally shitting bricks! For some strange reason, the antiquated products of those aforementioned ancient men, now command more capital to acquire than when they were first released (Rupert was famously amused by the prices people were paying for 1073s, which he considered the best he could do at the time with limited resources). I’m not sure what they’d say about 64-bit Floating Point or unlimited Dynamic Range, though we know what they tried to do with +/-15, 16, 18, 24, 40, even 60v Voltage Rails and >+28dBu Input/Output Levels. I’m also not sure if they’d be angered or amused, but I think they’d all be a little relieved to hear that: Gain Staging seems to be a “thing” now. (Please don’t take offense to this. I’m only being facetious and smart-assey in hopes that others who stubble upon it may be mildly amused enough to be slightly informed. I’m sure you’ve produced vastly more and better actual music than I, as music has always been my friend-with-benefits, never my wife. And...my name is also Gary. And the international board of Garys clearly stipulates that one Gary cannot be angry with another Gary. My fathers name is also Gary, making me Gary Jr. And actually, if you are mad at another Gary it should be him, he’s the one that put a soldering iron in my hands at 5-ish, and he’s also the one that taught each the dark arts of talkin’ shit! Garys of the world unite! VU Meters FOREVER!!!)
Great video! Love that song and all your others also. Especially your song "Amen" I listen to it often to bring me peace. Thanks for all you do! You are the reason I chose Studio One and I am so happy I did!
This is so great Joe, specially when creating automation, when I need to turn the overall track volume up or down for gain staging without messing up the automation. Definitely this is something I can do on every mix. Thank you, Joe and Presonus!
I use the Mixtool plugin all over my mixes! In all the templates that I use, every track has a Mixtool plugin ready to go. That's a great point about making sure not to clip when recording. But the good news is, when recording in a digital environment like Studio One, you can record as quietly as you want and add as much gain as you need later. I find that especially handy when I'm acting as my own tracking engineer. I always make sure I have tons of headroom when recording, then I don't need to worry about any clipping and I can add gain as needed during mixing.
Within reason, but yes. There is still a noise floor, if you get too close to it, you’ll be bringing it up too. So not as quietly as you want, but you never need to get near the ceiling. Peaks at -10dBFS is a safe rule of thumb. I track at -6 to -3 peaks, but I’m a tracking engineer, and there’s a lot of tools at my fingertips.
@@henryijeoma Short Answer: I’m pretty DAW Agnostic, but I really like Studio One. Unnecessarily Long Answer: I track many different things these days so it totally depends on the project. If tracking a Live show, probably it’ll be mixed on a digital console and the multitracks coming from there to whatever makes sense with that ecosystem. Midas/Behringer, vs A&H, vs Presonus Studio Live. I really like the handoff from Presonus Studio Live to Studio One, it flows from the stage to the studio pretty seamlessly and That’s what I’d probably recommend for a band that’s just starting to invest in gear. (The Behringer stuff is better bang for buck, but slightly less intuitive.) If tracking in a studio, the studio is usually built around something…often Pro Tools. If recording Location Sound (dialog) I’m on a field recorder (Sound Devices or Zoom). If doing ADR I’m probably in Davinci Resolve/Fairlight. And if I’m mixing and have free reign, I’m Hybrid on Reaper/CSI/and Analog. So again, I don’t know that Studio One is the “best” at doing any single function, but I do think it’s the “most” complete, intuitive, jack of all trades, get in and get to work, jump right in DAW today. (It’s also, for me been super stable. And flexible with MIDI control. And quick with Drag-n-drop everywhere and single actions rather than pro tools taking five steps) More than you asked I know. But I’m a sound guy!
thx joe for that, i think this could forever end the discussion about subtractive and additive eq also, because - verifying the outcome of both in the way you did here with the gain, will proof also that both are exactly the same inside the digital world, the only thing about deciding to add or to sub with eq, is the mindset of what to archive in specific way's, like filtering low end out of a reverb for example. in fact, it would be same to boost the rest and gaining down...
I always skip his videos, ok not "Always" like yourself I use PreSonus for live gear. "Instruments" mics. I just got Studio 1.6 pro, so I'm spending quite a bit of time with you Joe. The couple of songs that are on my TH-cam channel through an old church mixing board into Tascam DP-03. No editing,click, tracking myself. I got a degree in audio engineering 20 years ago. Decided to mic up again a year or two ago. Getting back into it. Thanks for your time and explaining these. ✌️
I usually create a folder and put all my tracks into it and assign that folder to a VCA channel. Then I just adjust the VCA fader when I want all tracks to go up or down evenly. This tends to work well with any tracks that also have automation on them seeing as it moves that automation data along with the fader.
The one place I tend to run into trouble is the tip of the mix-buss where the sun of all channels creeps up and, by default, there’s no meter there. I have even used a master VCA to pull all channels down. Because I tend to have an ssl compressor at the top of my mix-buss, I know when I’m hitting it too hard. While the right thing to do is proper gain staging, it’s good to know that a mix tool can fix the mistake. The VCA requires a lot of thought to not mess up FX balance.
Who here started having a small panic attack about the condition of their monitors due to the rumble just before Joe Gilder said, "Sorry my kids are going nuts upstairs"? Me, that's who.
Well done, excellent demonstration and explanation (on the difference between 64bit floating point summing engine clipping and input A/D clipping, too). 👍👍👍
WARNING!! Yes, DAWs use 32-64 bit Floating Point Math now, especially in their Mix Engines. Yes, this provides nearly unlimited dynamic range in between your converters. But, this is only true if you Do Not use any Analog Modeled code, or use any actual Analog gear. Part of what brought many to Studio One is its integration between the DAW and the Analog world. - Presonus Mix Engine FX, Console Shaper, CTC-1, Softube Tape all analog rules - Presonus Pipeline XT to insert say 64 pieces of analog gear through two Presonus Quantum 4848s, better gain stage - Presonus Fat Channel plug-ins, guess what That’s all before you touch your Universal Audio, Waves, Softube, Plug-in Alliance, Slate, Tokyo Dawn, etc, etc plug-in folders. Gain Staging is critical going into your A/D converters. Gain Staging is critical going out of your D/A converters. I suppose you could generate Everything inside the box. And only use digital plugins with no nonlinearities. And do No Dynamics Processing. Or you could just learn your Gain Staging, and Level Matching too while you’re at it, just in case you wanna spread your wings a little.
Hi Joe, Nice video and like you I tend to pull things down. Just like in the days of analogue. One thing I'd like to point out, which you likely already know is that whether you use 32bit floating point or as you demonstrated 64bit floating point you will get the same result. You can't clip 32bit floating point either. It's all in the floating point. In fact this is one of the ways that is used to make mp3 audio files sound so loud. They use 32bit to their advantage and simply crank it up. As long as the file remains 32bit float, you're alright. Try making a CD using those files and you have a problem. I had a guy purchase some online mp3 files and wanted to create a CD for his girlfriend and I had to do a whole lot of things in order to be able to do just that. Again thanks for the video. Thanks to both yourself and Gregor for all the Studio One videos the two of you create. Love Studio One and always learning something from you two.
Who would dislike this and why? I don't understand sad people with sad life's ...if they would at least leave a comment about why they dislike we might learn something else.
Its so hard to do that if you’ve been mixing on analog consoles for years. So you drive the inserts (plugins) your pre-fader gain and pull down the output to prevent clipping. So give yourself lots of headroom at every stage. So that on all your busses you might just need to pull the fader down 1 to 3db. That saves you more time than relying on digital tech because you could clip at any stage and you’d be looking for it like a needle in a haystack. If you get it right, the only thing you need to do is pull down the master buss 1 db before you export it to file.
You're only playing one channel into the Mix Buss. If you had multiple channels and one of them was clipping, you couldn't turn down the Mix Buss and keep the relative volume level of the various channels to keep your "static" mix the same. You have to fix the channel that is clipping in that case.
Actually makes sense too, because you are basically overflowing the space a 64bit float can handle on the computer, and if some developer didn't just put a check there to ignore pushing the value any further under the hood than the maximum allowed numeric space the entire program would crash, so technically it wouldn't clip your output either, but that's besides the point :p But fun though, I never actually thought of that, so let me officially forgive you for that weird attempt at explaining 64 bit floating point ;) Honestly, I wish I could spend more time working in Studio One than I have to spend at work...
Joe, your videos have convinced me to make the switch to Studio One... I was totally skeptical because im shallow and i didnt think it looked "cool"... i've been a Logic user for 10+ years now and i happened to buy a Presonus Quantum interface for the latency and i'll be damned if you havent won me around with all these features!!
@@henryijeoma I've never looked back. I'm on Artist v6 now. Had some early issues with crashing plugins but sorted it out by regularly refreshing plugin list. Still on Artist though as the upgrade to pro is probably more money than it's worth for my workflow.
Correct me if I'm wrong here Joe, but I don't think you even need the trim plugin. On bus channels, you can literally just turn down the fader and the clipping will stop.
In the 90s, nobody needed to make these funny experiments. All recordings were made trough analog gear, analog mixing console and the print to on multitrack tape, to get mixing back again in the analog mixing console with some analog processor... and? That's are the sound that ALL PRODUCERS LOVE and want to sounds now... yeah, in the cold and sterile digital world of 64bits, digital cliping and bla blablaa... 80s and 90s = Get set, tart recording, mixing, mastering and go to Billboards... Now are soooo many steps to get a poor digital master... Thanks.
This was a bug in Studio One 4 earlier versions. Although the channel meter was -14Db, ( I think because I have selected oversampling in one ampsim plugin) there were intersample peaks not cached by channels meter or any (I had used several additional) meters and the mixbus clipped pretty badly. It took me an hour to find the cause. At last, I had put true peak limiter to every single channel and removed one by one until that guitar channel mentioned above and I then only could see the channel clipping on limiters meter. I am happy they fixed it and I am happy that it won't be a problem now. It had driven me nuts back then.
Thanks! To address the end of your enlightening, timesaving video (perhaps someone's mentioned it earlier, anyway) - I believe that if you record at 32-bit (also floating point) instead of at 24-bit, you can actually distort/ overload when recording, then just turn the gain/volume down afterwards, and you'll be just fine. Overload gone. My default. Happy 2021!
@PreSonus when are you going to fix 666db bug that so many of us have experienced, and made tickets for it? What happens is, you lose your audio and your Master Bus pegs out and freezes at 666db, the only remedy to force and close the session then restart. The worst part, you never know when it will occur so you might loose your work. There is a forum discussing this issue since 2015 with S1 version 3, some of us upgraded to the latest version and still have issues! And is 666db freeze up intentional? Why 666db? It's about time someone taking this issue seriously, Thanks
As long as internally is used floating point data representation and summing, there is no reason to clip bc floating point has literally infinite level (internally, not output to DAC, it's still limited at 0dB). The Single or Double float is only about dynamic range - single has 144dB (24 bit mantissa x 6dB/bit), double has 318dB of dynamic range (53 bits of mantissa). The problem with signal level can occur with plugins using some threshold (limiter, compressor, saturation & Co.), they needs to be driven below 0dB as usual. And last, the VST3 defines processing with double precision (and also needs to be implemented in the plugin, but it's not mandatory), but VST2 is single only (!!) (DAW will convert double to single before routing into VST2). So always try to use VST3 to get the best sound quality.
Hmm, are you sure VST2 is single presicion? I seem to remember from the old Steinberg SDK that the processing function accepted 2 *in and 2 *out both as double, but I might be remembering wrong. For sure I have never worked with the VST3 API... Anyway, already starting to doubt now, they could have been float, but I thought double...
Interesting. However, I was held to the belief that gain staging is dealt with at the source, i.e; mics, keys, etc., and that's why I came here...looking for gain staging at the creation of the tracks, not at the mixing stage.
Great video Joe! Thank you!! How does the 64 bit handle noise floor? So if we recorded at a lower level can we use gain to bring the level back up at 64 bit without introducing crazy noise into the audio?
That answer is NO. Your question is exactly correct, the converters still have a noise floor. Capture signal, near the noise, and your recording will have a low signal to noise ratio. Track with your peaks around -10 to -6dBfs, averages around -20 to -18, and you should have both ample headroom and a healthy SNR. But again, if you track Super Low, then boost later, you’re boosting Signal AND Noise.
@@G_handle What if the gap between peaks and averages is larger (say peak at 0 and average at -20)? Would this reflect poor mic-ing technique? Maybe a limiter/compressor would be needed? I'm newer to recording/mixing (sorry if this is basic).
@@BOBopalooza Transient sources, say snare hits, are exactly as you describe. So first rule: protect the peaks. If the peaks are clipping your A-D converters, well they're clipped. (That clipped converter transient is Now your Snare sound. You may actually Like that, music is subjective, so you are free to break the rules, but as a rule Don't clip the converters.) That leads to Headroom. In theory if you are peaking at -0.1dBFS then you're technically not clipping the converters. However, A) every snare hit isn't the same, and what happens if the drummer gets happy? Giving yourself 3, 6, 10, 12 dB of Headroom (meaning don't let peaks go above -3, -6, -10, -12dBFS) insures that you won't Accidentally clip. B) The metering you're using may not be telling you the whole truth. Transients are so fast that most meters can't keep up, both by physical limitations and design choice. There are many meters out there to check your average and peak levels, VU, PPM, ETC. and then there are "True-Peak" meters that factor in ISP (Intersample Peaks) that 'could be' in the signal, but that level of precision should probably be left for mastering, not tracking and mixing. Give yourself room to work. Now, during Tracking and Mixing, back when the Multi-Tracks and 2-Tracks were on TAPE, best practices were different. There was (and still is) a Noise "Floor" and a point where the signal Distorts or "Ceiling". With the Noise Floor, every piece of Analog equipment has one, then and now. Whatever circuitry is in your analog signal chain, before it reaches the recording medium, the noise floor is set by the "lowest common denominator" ( or the device with the highest noise floor. In a professional studio full of high end gear that usually was the Tape Machine and Tape itself, so many systems of noise reduction were employed to squeeze out a few more dB of "signal to noise ratio". But the main tactic was to record as hot as possible, placing the signal as far away from the noise as possible. So get away from the floor. But that meant getting closer to the ceiling. Now what happens at the ceiling for analog and digital are totally different. I'll limit this to Analog Tape and Digital Converters. When you hit the Ceiling of converters, they are pretty much perfectly linear up until 0dBFS and then they clip. Analog magnetic tape though Changes dramatically based upon its input signal level, and has many different artifacts at different signal levels (and tape formulations and settings and speeds and biases and...). The primary point is this: When you push a signal up into your Converters, the signal is basically flat and linear until it clips. When you push a signal up into Tape the signal goes from mostly flat-ish and pretty linear to clearly altered frequency responses, compressed dynamics, and obvious harmonic distortion nicknamed "saturation". Most Engineers don't like the sound of digital clipping. Most Engineers learned to Love the various types of artifacts that come from Analog distortion/clipping. Magnetic Tape at its absolute best reaches 90dB of dynamic range, but is mostly 80dB DR. 24-Bit Digital Audio is theoretically 144dB DR, but in practice the best converters get to about 120dB, average...say 110dB. With tape though, you would Push into the Ceiling and maybe get another 10-20dB! away from the Floor depending on how much saturation you like. With digital, there's basically nothing good at the ceiling and you should likely will keep 10dB of Headroom to make sure you never find out. (The 24-bit SSL 2+ converters I just bought for less than $250 bucks Obliterate the 16-bit Digidesign converters I first bought in 1996 for many thousands of dollars. The chips keep getting better, just like your smartphone. Tape remains the same. And BTW 16-bit has a theoretical DR of 96dB which is part of why most Analog engineers Hated the switch from recording on tape to digital at first. Many attributed the deficiencies of Those converters to Digital Audio itself. However I think today Most engineers would be happy to never touch another tape machine in their lives, except for funsies!) So, long way of saying... Just don't clip your converters, give yourself some headroom, you have Plenty of dynamic range to spare.
@@G_handle wow, thanks for the awesome response! That was a great read. Are the analog "pushing the ceiling" harmonics you describe similar to what's happening when distorting a tube guitar amp? Have you used any "tape simulation" plugins? I'm curious how close those are getting to sounding realistic. And to your snare example... it's good knowing the noise floor can accommodate an input averaging, say, -35dbs. That just sounds really quiet coming out of my monitors, haha. Is there any reason to buy a high end A/D or D/A converter these days? I'm wondering where you'd even notice a difference.
@@BOBopalooza 1) Glad Im not typing into the digital abyss. 2) Exactly. Push a 1k sine-wave into any analog circuit and at some point the circuit won't be able to handle the voltage. On an oscilloscope that perfect wave will change shape and look "distorted" hence the harmonic distortion. Basically a failure of the system, that we use to our advantage. A bug turned into a feature in code-speak. The labels or nicknames for all the different phenomena are less important than understanding that Every system has it's technically cleanest operating level, a lower level that's too much noise, and a higher level that's broken up and distorted, basically its failure point. Ideally you would take Every piece of equipment and Plugin and push signals through for yourself to see where its too noisy, cleanest, and if when it starts to distort, when is it Adding something that you like, and when is it too much. Electric guitarists in particular do this as regular course, and half the gear they use is for its desirable failures. 3) So many amazing tape plugins that I never need to see tape again. (Chow is FREE. Hornet is cheap. Softube goes on sale often.) 4) In general, track your peaks hot but with headroom. To the original comment, there still is a noise floor so you still want to stay away from it, but you don't (in digital) want to (ever) touch Zero. If you record too low, then have to boost the signal in the mix, you're boosting the noise with it. The SNR remains whatever you captured. Better to record hot, then turn down for the mix, decreasing signal and pushing the noise floor lower. Basically, in my opinion, all the modern converters from reputable manufacturers are fantastic now. If you don't know if you need to upgrade, you don't. Definitely not for Recording and Mixing. Mastering...maybe. But at the point where you need mastering converters, you should consider letting a Mastering Engineer take over. A $2,000 interface two channel interface would buy you quite a few mastered songs, and with each one comes more than just the final product. You get to see, or rather Hear what they hear.
Reaper is ahead with many things to be honest, especially when it comes to system resource demands, and yeah probably surround mix too if you need that. But to me there is a reason after trying them all for many years I have settled in Studio One. Every DAW has flaws, but Presonus has so much going for it, productivity wise, comfort wise, they listen to their users, and they don't re-invent wheels (oh hi there ARA :p). Studio One feels like home, and it is getting better and better all the time.
Nice tip and a good observation. As you said, getting it right/close to start with is probably the better bet, but if you need to fix just that little bit of clipping without trying to fix it at all of the tracks/channels seems reasonable (and of course not having to reduce anything by 96db :) ). I also thought of something while watching. I usually have it set as your example shows with all of my tracks going to busses and those busses going to the "Main" hardware output. I've seen others who add a Master Bus that then goes to Main with all of their Track Busses going to the Master Bus on the way out (which is how I used to have this setup in Cakewalk because that's just how it defaulted over the years). Since the function of a Bus vs the Main is different at some level, do you have any advice about the benefits or uselessness of adding this extra bussing layer? Example, a Bus can have the Input Gain and Polarity where the Main has to have the Channel tool added to get those features. Is there anything else "special" about the Main?
It's what I do now anyway I think you said it before? ..."just slap mixtool on the bus" ..always felt I was cheating but not anymore Thanks..The mind is a weird thing right!
A question: I've been working with the console shaper lately, and as far as I can understand using the gain/trim knob comes after it hits the console shaper (but before inserts and sends). Is that correct? If so, this will not help me to get a controlled crosstalk, would it? I mean, one channel could still be super hot into the console shaper, right? If I'm correct: Could you guys at PreSonus please move the gain control "up"?
Excellent, thank you very much. There are simple things that do not occur to us because of their simplicity, genius has two forms, simple and complicated. another thing, i don't know why when i use fl studio in vst mode in studio one 5 i can edit audio but when i export it even if i bounce internally, for some reason i don't have audio, and the same if i have a lot of tracks of a great arrangement the audios of fl studio when i export, this only when i export or i bounce internally Ctrl-B the audio is cancelled, you only hear it in midi but really there is not. thanks, if you could help me it would be a great relief.
Gain staging is a thing now because all the plugins which have been designed to emulate the gear you used back then, are designed to operate best at a certain gain level.
64-bit floating point math didn’t change the fact that Gain Staging is still The Thing. I wonder if we explained voltage rails, SNR, and headroom if the mouse clickers would spontaneously combust. As a Recording Engineer in the 70s, I’m sure you tracked to tape every channel as hot as you could before it broke up to avoid the noise floor. When I got that 2-inch with 24-tracks of hot as hell signal, the first step of the mix is to Gain Stage it at the inputs of the board to “nominal” calibrated at 0VU RMS which inside of my board is -2dBu. Then we Gain Staged to unity through every piece of equipment in the chain. Now no more tape machines, every track is recorded super low with tons of headroom, and the first step of the mix is to Gain Stage everything Up to nominal, -18dBFS, and often into a Tape Machine Emulation Plugin. Hardware inserts, and Analog Emulation plugins all over the mix, and we’re still Gain Staging and Level Matching as much as ever.
@@garyshepherd9226 Welp, I learned Gain Staging in the late 80s. Both in the Studio and in Live Sound. Not sure when the fancy label came about, but it was long before TH-cam. I suspect, as with most things in Audio, the term comes from the broader Electronics field. The capacitors, resistors, transistors, transformers, tubes, diodes, integrated circuits, discrete operational amplifiers, etc. don’t really care what signal is being passed through them, or whether or not when patched into a transducer at the end of a long series of components, it generates a “sound” that subjectively sounds “good” to any one human ear, or billions across generations. Each component simply says “If you Don’t put this into me, I won’t do anything. If you put this to this in, I’ll put this out. However, if you put This in, I’ll blow up!” All those metal boxes, faraday cages really, whether shaped like a desktop to be sat at, or maybe a standardized 19” width of various unit height to be racked together with other metal boxes, they all had (have) some cocktail or electronic wizardry inside. But luckily, the electronic engineers who concocted these works of art to help other artists produce their own art, thought “ hey, maybe the end users will want to “interconnect” two or two-hundred of these boxes together, in unpredictable ways. And we already know that every component, in every circuit, in every box has a ‘not-enough, too-much, just-right’ pain threshold. So we better train these guys how to do that, as well as give them some basic information about each box, and give em a bunch of different ‘controls’ and relevant ‘meters’ on the outside of the box, so they don’t get themselves into too much trouble. We know these music guys are hopeless, but the broadcast and live sound guys seem to really want to learn this stuff. If they do, we electronic engineers should call them audio engineers.” Bill Putnam, Colin Sanders, Saul Walker, David Dearden, Paul Wolf and the King of Kings himself, now 94-years young, Sir Rupert Neve, all spent their lives on a quest, in search of “The Perfect Circuit”. Billions of records, and broadcasts, and live shows have passed through circuits that were the product of their productivity. Whilst the code-writers have done some amazing things, 64-bit floating point math being one of them, there might be a reason, other than nostalgia, that every major code-for-audio company is desperately trying to reproduce in code, all of the sonic qualities of analog circuits. And hint, the unlimited dynamic range, whilst useful in the background, has little to do with the price of tea in China. Two other observations: (I really am trying not to go too deep) A) The two main tools of the audio engineer remain: EQ & Dynamics (specifically Compressors). Both are used ultimately to control, and primarily REDUCE...Dynamic Range. A compressors government name is ‘Dynamic Range Compressor’. A limiter puts a ceiling on how high a signal can go. Raise the fader all you want, and your signal will either get a haircut or get beheaded. Gates and Expanders work on the bottom end of a signals Dynamic Range, usually to cut it off at the ankles.... or knees. EQs, formerly and formally known as Equalizers, seem to only exist within the Frequency arena of a signal, but no... EQs are Static Dynamic Range controllers, divided into 2 or more frequency bands. Imagine a 1/3-octave Graphic Equalizer: on it you will find 31 little tiny faders, each of which does the same thing as the 100mm faders on a mixing console, except they only boast or cut a specific frequency band within the signal they receive, regardless of that signals input level, aka static. There’s no minimum threshold. In “Subtractive EQing” you run a signal through the machine, whichever frequencies you don’t want, cut em, anything popping out, tuck it back in. You whittle and carve away at whatever walked through the door, trimming the fat, nipping and tucking like either a butcher or a cosmetic surgeon. In the end, she will hopefully be a more beautiful version of herself (disclaimer: beauty is in the ear of the beholder), and when she’s back out on the street, all will be stunned and none will be the wiser. Or....You can go nuts and while she’s unconscious on the operating table, aka mixing console, start injecting her with 1000-CCs of this, and two-pounds of that, you can use those scalpels to open her up and start Implanting foreign objects into her body with no regard for her natural beauty or health or what she looked like before your hands all over her. Now she’s Jessica Rabbit, and trying get her fit in the mix is like trying to walk into a party with her in a hot red mini-skirt without being seen, or in this case heard. Whether you’re producing a feature film with an ensemble cast of glamorous Hollywood stars living their best selves, in the best wardrobe, hair, and makeup money can buy, or you’re making a low budget porno movie with impossibly endowed actors that can’t act (I could go on), you’re still going to package this hot mess into a box of some sort, and that medium of consumption will have dynamic range limitations. Whatever sounds you have coming out of your speakers, the loud sounds are gonna have a relationship with the quiet sounds and with the silence. Ultimately, the Dynamic Range of your Mix, turns out is a big part of the quality of that mix, both Objectively now measured in LUFS, and Subjectively now measured in Downloads. B) It’s ironic to me that the same people excited about unlimited dynamic range, are the same ones most likely to be on the enemy side of the loudness wars! They seem to want to both put in sounds with +/-96dB of epileptic convulsions, and THEN run limiters at full throttle until their waveforms look like their DAWs are literally shitting bricks! For some strange reason, the antiquated products of those aforementioned ancient men, now command more capital to acquire than when they were first released (Rupert was famously amused by the prices people were paying for 1073s, which he considered the best he could do at the time with limited resources). I’m not sure what they’d say about 64-bit Floating Point or unlimited Dynamic Range, though we know what they tried to do with +/-15, 16, 18, 24, 40, even 60v Voltage Rails and >+28dBu Input/Output Levels. I’m also not sure if they’d be angered or amused, but I think they’d all be a little relieved to hear that: Gain Staging seems to be a “thing” now. (Please don’t take offense to this. I’m only being facetious and smart-assey in hopes that others who stubble upon it may be mildly amused enough to be slightly informed. I’m sure you’ve produced vastly more and better actual music than I, as music has always been my friend-with-benefits, never my wife. And...my name is also Gary. And the international board of Garys clearly stipulates that one Gary cannot be angry with another Gary. My fathers name is also Gary, making me Gary Jr. And actually, if you are mad at another Gary it should be him, he’s the one that put a soldering iron in my hands at 5-ish, and he’s also the one that taught each the dark arts of talkin’ shit! Garys of the world unite! VU Meters FOREVER!!!)
@@G_handle This Gary is definitely not angry! I just worry that all the less knowledgable hobby recordists will think they have to learn Gain Staging - they don't and most will never ever need it. I know the pros getting external files to mix do need this - but it really is more a pro thing. Have a great day!
This is how I normally mix in my DAW. Clipping at the output of the individual outputs of the tracks make no difference in the sound quality, as long as you don't let your master output clip (by using a gain utility to reduce volume after). That is why I prefer using 32 (or 64) bit floating point as much as possible.
@@BuenoLaidley read my other comments in this thread, but: Just because your new digital freeway has no real speed limit anymore, doesn't mean your Analog car can suddenly do 200mph. Nor can it turn corners at high speeds. Nor is it on the road alone. If you use Anything Analog or Analog Emulated, or if you use Any processing that is Threshold based (Compressors, Limiters, Expanders, Gates, Saturation, Distortion, ...) Anything that is Dynamic, at any "Stage", you're gonna want to Gain Stage for that Stage no matter what.
@@G_handle I have been mixing for years now. I know that the level going into your plugins must be gain staged properly, but the level that is post mixer doesn't have to be below 0db.
How do I know what the sweet spot is when it comes to volume while using different plugins? What is the optimal volume/gain going into a plugin? Obviously, it's different for different plugins, but how do I know?
I've personally never worried about it, and I'm happy with the way my mixes sound. I think too many people get caught up in doing "correct" gain-staging and at the expense of learning to listen and make good decisions based on how it sounds. - Joe Gilder
Would setting the gain knob at the top of the channel to -24 be equivalent to using the Mixtool at -24 to accomplish this? Is there an advantage to doing it one way or the other?
🤦♂️ spreading the idea that you don't need to learn proper gain staging on a mix channel. It's only the first and most important thing people should do on every mix. wow
Not disrespecting the presenter but that's really old news. for example here th-cam.com/video/CGRusg9GnAg/w-d-xo.html where he also shows the behavior of analog modeled plugins.
@@technoscout yesterday you’re right. I thought it was for something else. Here’s a forum that addresses pros and cons forums.steinberg.net/t/32-bit-or-64-bit/98651/24
Joe Gilder always brings the best Value about Studio One.
Agreed!!!!!!
Presonus Studio One ☝🏾 The Best DAW Ever!!!
If they upgraded and allowed us to mix in 5.1 or 7.1 then it would be. Maybe it’s coming but I had to switch to reaper so that I can mix my show. 🥺🥺
EVER... and ever.....
what joe shows here, is not a studio one specific topic. it works for all daw's....
unless you want to automate pitch on an audio track lol
@@TJ-hs1qm agreed, although just a heads up if you're not aware that may help, you can insert melodyne on the audio channel you want to automate the pitch of, then add pitch as an envelope and draw your automation, becomes quick once used to it. be aware you need to add it as an insert, rather than the 'edit with melodyne' auto option in order for the automatable pitch envelope to be available
Most plugins are prefader. So it's important to use the mixtool, gain tool, etc to control gain that goes into plugins. The fader only changes the output volume. It doesn't affect plugins
Thank you! did not know that, but actually makes sense thinking about it. Great little nugget of info. Cheers!
Great point. On my last project I was hearing some noise that I traced back to overloading an amp sim. There was a compressor and the fader after that keeping the final volume okay but it was clipping in the amp plugin.
jhughs3 check your input/output level in the amp sim and if all is fine, add the mixtool before the chain
Just use the gain on the input controls in the studio one mixer. That is pre plugins 😉
@@AntonioRockGP That's exactly what I was about to say lol
The Mixtool is nice in a demonstration like this for visual representation. But don't forget that it's much easier to use the trim knob at the top of the channel for any necessary gain adjustments. No need to add plugins for this job. Before I start a mix, I always get my input levels in the same ballpark using the trim knob. Makes keeping levels under control way easier. One of the main reasons I switched to Studio One. Great feature.
What I learnt, is use Hornet VU Mk4 or LU MK2 (both of them combined costs around USD 5 only).
Honestly, the easiest workflow ever.
Put VU MK4 as the first plugin and set it to -18dBFS RMS values so that VU MK4 will make sure the signal coming into VU will not leave exceeding -18dBFS RMS so that you can use your analog modeled plugin with ease.
Now, I put LU MK2 as the last plugin and I set it to -23 LUFS (for percussive sounds Momentary and for other sounds Short term). So now all of my channel are at equal loudness. Then I use fader and do a anchor style mixing.
(Re-posted from a reply to a comment below for public amusement...)
Welp, I learned Gain Staging in the late 80s.
Both in the Studio and in Live Sound.
Not sure when the fancy label came about, but it was long before TH-cam.
I suspect, as with most things in Audio, the term comes from the broader Electronics field.
The capacitors, resistors, transistors, transformers, tubes, diodes, integrated circuits, discrete operational amplifiers, etc. don’t really care what signal is being passed through them, or whether or not when patched into a transducer at the end of a long series of components, it generates a “sound” that subjectively sounds “good” to any one human ear, or billions across generations.
Each component simply says “If you Don’t put this into me, I won’t do anything. If you put this to this in, I’ll put this out. However, if you put This in, I’ll blow up!”
All those metal boxes, faraday cages really, whether shaped like a desktop to be sat at, or maybe a standardized 19” width of various unit height to be racked together with other metal boxes, they all had (have) some cocktail or electronic wizardry inside. But luckily, the electronic engineers who concocted these works of art to help other artists produce their own art, thought “ hey, maybe the end users will want to “interconnect” two or two-hundred of these boxes together, in unpredictable ways. And we already know that every component, in every circuit, in every box has a ‘not-enough, too-much, just-right’ pain threshold. So we better train these guys how to do that, as well as give them some basic information about each box, and give em a bunch of different ‘controls’ and relevant ‘meters’ on the outside of the box, so they don’t get themselves into too much trouble. We know these music guys are hopeless, but the broadcast and live sound guys seem to really want to learn this stuff. If they do, we electronic engineers should call them audio engineers.”
Bill Putnam, Colin Sanders, Saul Walker, David Dearden, Paul Wolf and the King of Kings himself, now 94-years young, Sir Rupert Neve, all spent their lives on a quest, in search of “The Perfect Circuit”. Billions of records, and broadcasts, and live shows have passed through circuits that were the product of their productivity.
Whilst the code-writers have done some amazing things, 64-bit floating point math being one of them, there might be a reason, other than nostalgia, that every major code-for-audio company is desperately trying to reproduce in code, all of the sonic qualities of analog circuits. And hint, the unlimited dynamic range, whilst useful in the background, has little to do with the price of tea in China.
Two other observations: (I really am trying not to go too deep)
A) The two main tools of the audio engineer remain: EQ & Dynamics (specifically Compressors).
Both are used ultimately to control, and primarily REDUCE...Dynamic Range.
A compressors government name is ‘Dynamic Range Compressor’.
A limiter puts a ceiling on how high a signal can go. Raise the fader all you want, and your signal will either get a haircut or get beheaded.
Gates and Expanders work on the bottom end of a signals Dynamic Range, usually to cut it off at the ankles.... or knees.
EQs, formerly and formally known as Equalizers, seem to only exist within the Frequency arena of a signal, but no...
EQs are Static Dynamic Range controllers, divided into 2 or more frequency bands. Imagine a 1/3-octave Graphic Equalizer: on it you will find 31 little tiny faders, each of which does the same thing as the 100mm faders on a mixing console, except they only boast or cut a specific frequency band within the signal they receive, regardless of that signals input level, aka static. There’s no minimum threshold. In “Subtractive EQing” you run a signal through the machine, whichever frequencies you don’t want, cut em, anything popping out, tuck it back in. You whittle and carve away at whatever walked through the door, trimming the fat, nipping and tucking like either a butcher or a cosmetic surgeon. In the end, she will hopefully be a more beautiful version of herself (disclaimer: beauty is in the ear of the beholder), and when she’s back out on the street, all will be stunned and none will be the wiser. Or....You can go nuts and while she’s unconscious on the operating table, aka mixing console, start injecting her with 1000-CCs of this, and two-pounds of that, you can use those scalpels to open her up and start Implanting foreign objects into her body with no regard for her natural beauty or health or what she looked like before your hands all over her. Now she’s Jessica Rabbit, and trying get her fit in the mix is like trying to walk into a party with her in a hot red mini-skirt without being seen, or in this case heard. Whether you’re producing a feature film with an ensemble cast of glamorous Hollywood stars living their best selves, in the best wardrobe, hair, and makeup money can buy, or you’re making a low budget porno movie with impossibly endowed actors that can’t act (I could go on), you’re still going to package this hot mess into a box of some sort, and that medium of consumption will have dynamic range limitations. Whatever sounds you have coming out of your speakers, the loud sounds are gonna have a relationship with the quiet sounds and with the silence. Ultimately, the Dynamic Range of your Mix, turns out is a big part of the quality of that mix, both Objectively now measured in LUFS, and Subjectively now measured in Downloads.
B) It’s ironic to me that the same people excited about unlimited dynamic range, are the same ones most likely to be on the enemy side of the loudness wars! They seem to want to both put in sounds with +/-96dB of epileptic convulsions, and THEN run limiters at full throttle until their waveforms look like their DAWs are literally shitting bricks!
For some strange reason, the antiquated products of those aforementioned ancient men, now command more capital to acquire than when they were first released (Rupert was famously amused by the prices people were paying for 1073s, which he considered the best he could do at the time with limited resources).
I’m not sure what they’d say about 64-bit Floating Point or unlimited Dynamic Range, though we know what they tried to do with +/-15, 16, 18, 24, 40, even 60v Voltage Rails and >+28dBu Input/Output Levels.
I’m also not sure if they’d be angered or amused, but I think they’d all be a little relieved to hear that:
Gain Staging seems to be a “thing” now.
(Please don’t take offense to this. I’m only being facetious and smart-assey in hopes that others who stubble upon it may be mildly amused enough to be slightly informed. I’m sure you’ve produced vastly more and better actual music than I, as music has always been my friend-with-benefits, never my wife. And...my name is also Gary. And the international board of Garys clearly stipulates that one Gary cannot be angry with another Gary. My fathers name is also Gary, making me Gary Jr. And actually, if you are mad at another Gary it should be him, he’s the one that put a soldering iron in my hands at 5-ish, and he’s also the one that taught each the dark arts of talkin’ shit! Garys of the world unite! VU Meters FOREVER!!!)
Great video! Love that song and all your others also. Especially your song "Amen" I listen to it often to bring me peace. Thanks for all you do! You are the reason I chose Studio One and I am so happy I did!
South Africa, Limpopo is here. S1 Forever
This is so great Joe, specially when creating automation, when I need to turn the overall track volume up or down for gain staging without messing up the automation. Definitely this is something I can do on every mix. Thank you, Joe and Presonus!
I use the Mixtool plugin all over my mixes! In all the templates that I use, every track has a Mixtool plugin ready to go. That's a great point about making sure not to clip when recording. But the good news is, when recording in a digital environment like Studio One, you can record as quietly as you want and add as much gain as you need later. I find that especially handy when I'm acting as my own tracking engineer. I always make sure I have tons of headroom when recording, then I don't need to worry about any clipping and I can add gain as needed during mixing.
Within reason, but yes.
There is still a noise floor, if you get too close to it, you’ll be bringing it up too.
So not as quietly as you want, but you never need to get near the ceiling.
Peaks at -10dBFS is a safe rule of thumb.
I track at -6 to -3 peaks, but I’m a tracking engineer, and there’s a lot of tools at my fingertips.
@@G_handle exactly
@@G_handle is studio one your go-to tracking daw?
@@henryijeoma Short Answer: I’m pretty DAW Agnostic, but I really like Studio One.
Unnecessarily Long Answer: I track many different things these days so it totally depends on the project. If tracking a Live show, probably it’ll be mixed on a digital console and the multitracks coming from there to whatever makes sense with that ecosystem. Midas/Behringer, vs A&H, vs Presonus Studio Live. I really like the handoff from Presonus Studio Live to Studio One, it flows from the stage to the studio pretty seamlessly and That’s what I’d probably recommend for a band that’s just starting to invest in gear. (The Behringer stuff is better bang for buck, but slightly less intuitive.)
If tracking in a studio, the studio is usually built around something…often Pro Tools.
If recording Location Sound (dialog) I’m on a field recorder (Sound Devices or Zoom).
If doing ADR I’m probably in Davinci Resolve/Fairlight.
And if I’m mixing and have free reign, I’m Hybrid on Reaper/CSI/and Analog.
So again, I don’t know that Studio One is the “best” at doing any single function, but I do think it’s the “most” complete, intuitive, jack of all trades, get in and get to work, jump right in DAW today. (It’s also, for me been super stable. And flexible with MIDI control. And quick with Drag-n-drop everywhere and single actions rather than pro tools taking five steps)
More than you asked I know. But I’m a sound guy!
thx joe for that, i think this could forever end the discussion about subtractive and additive eq also, because - verifying the outcome of both in the way you did here with the gain, will proof also that both are exactly the same inside the digital world, the only thing about deciding to add or to sub with eq, is the mindset of what to archive in specific way's, like filtering low end out of a reverb for example. in fact, it would be same to boost the rest and gaining down...
I always skip his videos, ok not "Always" like yourself I use PreSonus for live gear. "Instruments" mics. I just got Studio 1.6 pro, so I'm spending quite a bit of time with you Joe. The couple of songs that are on my TH-cam channel through an old church mixing board into Tascam DP-03. No editing,click, tracking myself. I got a degree in audio engineering 20 years ago. Decided to mic up again a year or two ago. Getting back into it. Thanks for your time and explaining these. ✌️
I usually create a folder and put all my tracks into it and assign that folder to a VCA channel. Then I just adjust the VCA fader when I want all tracks to go up or down evenly. This tends to work well with any tracks that also have automation on them seeing as it moves that automation data along with the fader.
This is invaluable info! Great job!
Great video, i learned a lot. I know a lot about floating point and its great how Studio one uses it internally.
The one place I tend to run into trouble is the tip of the mix-buss where the sun of all channels creeps up and, by default, there’s no meter there. I have even used a master VCA to pull all channels down. Because I tend to have an ssl compressor at the top of my mix-buss, I know when I’m hitting it too hard. While the right thing to do is proper gain staging, it’s good to know that a mix tool can fix the mistake. The VCA requires a lot of thought to not mess up FX balance.
Again a great video Joe. I learn so much from you.
Who here started having a small panic attack about the condition of their monitors due to the rumble just before Joe Gilder said, "Sorry my kids are going nuts upstairs"? Me, that's who.
THIS IS WHAT I NEEDED!!!!!!!!
THANKS Joe great Info
You and Gregor are both incredible helpful on so many levels.
TOZZ
Well done, excellent demonstration and explanation (on the difference between 64bit floating point summing engine clipping and input A/D clipping, too).
👍👍👍
WARNING!!
Yes, DAWs use 32-64 bit Floating Point Math now, especially in their Mix Engines.
Yes, this provides nearly unlimited dynamic range in between your converters.
But, this is only true if you Do Not use any Analog Modeled code, or use any actual Analog gear.
Part of what brought many to Studio One is its integration between the DAW and the Analog world.
- Presonus Mix Engine FX, Console Shaper, CTC-1, Softube Tape all analog rules
- Presonus Pipeline XT to insert say 64 pieces of analog gear through two Presonus Quantum 4848s, better gain stage
- Presonus Fat Channel plug-ins, guess what
That’s all before you touch your Universal Audio, Waves, Softube, Plug-in Alliance, Slate, Tokyo Dawn, etc, etc plug-in folders.
Gain Staging is critical going into your A/D converters.
Gain Staging is critical going out of your D/A converters.
I suppose you could generate Everything inside the box.
And only use digital plugins with no nonlinearities.
And do No Dynamics Processing.
Or you could just learn your Gain Staging, and Level Matching too while you’re at it, just in case you wanna spread your wings a little.
the lightning fast inserts of Gregor's face are 💯 😄
Hi Joe,
Nice video and like you I tend to pull things down. Just like in the days of analogue. One thing I'd like to point out, which you likely already know is that whether you use 32bit floating point or as you demonstrated 64bit floating point you will get the same result. You can't clip 32bit floating point either.
It's all in the floating point. In fact this is one of the ways that is used to make mp3 audio files sound so loud. They use 32bit to their advantage and simply crank it up. As long as the file remains 32bit float, you're alright. Try making a CD using those files and you have a problem.
I had a guy purchase some online mp3 files and wanted to create a CD for his girlfriend and I had to do a whole lot of things in order to be able to do just that.
Again thanks for the video. Thanks to both yourself and Gregor for all the Studio One videos the two of you create. Love Studio One and always learning something from you two.
Who would dislike this and why? I don't understand sad people with sad life's ...if they would at least leave a comment about why they dislike we might learn something else.
Its so hard to do that if you’ve been mixing on analog consoles for years. So you drive the inserts (plugins) your pre-fader gain and pull down the output to prevent clipping. So give yourself lots of headroom at every stage. So that on all your busses you might just need to pull the fader down 1 to 3db. That saves you more time than relying on digital tech because you could clip at any stage and you’d be looking for it like a needle in a haystack. If you get it right, the only thing you need to do is pull down the master buss 1 db before you export it to file.
Great video Joe!
You're only playing one channel into the Mix Buss. If you had multiple channels and one of them was clipping, you couldn't turn down the Mix Buss and keep the relative volume level of the various channels to keep your "static" mix the same. You have to fix the channel that is clipping in that case.
Great vid thank you, starting to learn how better to use studio one.
u guys should have a gain staging preset for guitars when u play it it sets the gain staging
Actually makes sense too, because you are basically overflowing the space a 64bit float can handle on the computer, and if some developer didn't just put a check there to ignore pushing the value any further under the hood than the maximum allowed numeric space the entire program would crash, so technically it wouldn't clip your output either, but that's besides the point :p But fun though, I never actually thought of that, so let me officially forgive you for that weird attempt at explaining 64 bit floating point ;) Honestly, I wish I could spend more time working in Studio One than I have to spend at work...
Perfect practical demo. Yeehaa 😎👌
Joe, your videos have convinced me to make the switch to Studio One... I was totally skeptical because im shallow and i didnt think it looked "cool"... i've been a Logic user for 10+ years now and i happened to buy a Presonus Quantum interface for the latency and i'll be damned if you havent won me around with all these features!!
1year later. How's your Studio One experience been so far?
@@henryijeoma I've never looked back. I'm on Artist v6 now. Had some early issues with crashing plugins but sorted it out by regularly refreshing plugin list. Still on Artist though as the upgrade to pro is probably more money than it's worth for my workflow.
@@Jay836836 haha niceeeeeee
07:00 there is no metering for this crime
The Meter revolted!
thats a sweet instrumental innit
Hello Joe what microphone do you use in your you tube videos
Correct me if I'm wrong here Joe, but I don't think you even need the trim plugin. On bus channels, you can literally just turn down the fader and the clipping will stop.
This is a great way to lower things that already have volume automation so you don't have to redo the automaton
Hey man I like that track doc.
In the 90s, nobody needed to make these funny experiments. All recordings were made trough analog gear, analog mixing console and the print to on multitrack tape, to get mixing back again in the analog mixing console with some analog processor... and? That's are the sound that ALL PRODUCERS LOVE and want to sounds now... yeah, in the cold and sterile digital world of 64bits, digital cliping and bla blablaa... 80s and 90s = Get set, tart recording, mixing, mastering and go to Billboards... Now are soooo many steps to get a poor digital master... Thanks.
Excellent video very clever experiment
I was able to fix a mild clipping problem in a recorded track with the Izotop plugin. That thing is magical 😁🤘😎
It would have been nice if studio one had an auto gain match plug in.
I wish I knew it before, I would save a few weeks of my life back in the days.
Holy s#it. I love you bro! Studio one rules!❤👌
This was a bug in Studio One 4 earlier versions. Although the channel meter was -14Db, ( I think because I have selected oversampling in one ampsim plugin) there were intersample peaks not cached by channels meter or any (I had used several additional) meters and the mixbus clipped pretty badly. It took me an hour to find the cause. At last, I had put true peak limiter to every single channel and removed one by one until that guitar channel mentioned above and I then only could see the channel clipping on limiters meter. I am happy they fixed it and I am happy that it won't be a problem now. It had driven me nuts back then.
just add a pitch envelope to the audio clip and a native vocoder and studio one will be the best daw. without it, it sucks.
Mix engine tolerates extra 96 dB! Incredible! Thanks a lot!
This is the kind of thing we saw the FBI do in movies/tv in the 90s "audio magic"
Thanks! To address the end of your enlightening, timesaving video (perhaps someone's mentioned it earlier, anyway) - I believe that if you record at 32-bit (also floating point) instead of at 24-bit, you can actually distort/ overload when recording, then just turn the gain/volume down afterwards, and you'll be just fine. Overload gone. My default. Happy 2021!
Not unless you have 32-bit FP converters, which are just becoming available
@PreSonus when are you going to fix 666db bug that so many of us have experienced, and made tickets for it? What happens is, you lose your audio and your Master Bus pegs out and freezes at 666db, the only remedy to force and close the session then restart. The worst part, you never know when it will occur so you might loose your work. There is a forum discussing this issue since 2015 with S1 version 3, some of us upgraded to the latest version and still have issues! And is 666db freeze up intentional? Why 666db? It's about time someone taking this issue seriously, Thanks
With 64bit audio, you can record at a quieter input volume without adding noise.
As long as internally is used floating point data representation and summing, there is no reason to clip bc floating point has literally infinite level (internally, not output to DAC, it's still limited at 0dB). The Single or Double float is only about dynamic range - single has 144dB (24 bit mantissa x 6dB/bit), double has 318dB of dynamic range (53 bits of mantissa). The problem with signal level can occur with plugins using some threshold (limiter, compressor, saturation & Co.), they needs to be driven below 0dB as usual. And last, the VST3 defines processing with double precision (and also needs to be implemented in the plugin, but it's not mandatory), but VST2 is single only (!!) (DAW will convert double to single before routing into VST2). So always try to use VST3 to get the best sound quality.
Hmm, are you sure VST2 is single presicion? I seem to remember from the old Steinberg SDK that the processing function accepted 2 *in and 2 *out both as double, but I might be remembering wrong. For sure I have never worked with the VST3 API... Anyway, already starting to doubt now, they could have been float, but I thought double...
@@TheApeMachine Double was introduced first in VST 2.4 (activated with define "VST_2_4_EXTENSIONS"), earlier version are single only.
@@tommino8970 Ok fair enough, but yeah, around 15 years ago when I was working on plugins we were for sure using 2.4 :p
Interesting. However, I was held to the belief that gain staging is dealt with at the source, i.e; mics, keys, etc., and that's why I came here...looking for gain staging at the creation of the tracks, not at the mixing stage.
Great video Joe! Thank you!! How does the 64 bit handle noise floor? So if we recorded at a lower level can we use gain to bring the level back up at 64 bit without introducing crazy noise into the audio?
That answer is NO. Your question is exactly correct, the converters still have a noise floor. Capture signal, near the noise, and your recording will have a low signal to noise ratio. Track with your peaks around -10 to -6dBfs, averages around -20 to -18, and you should have both ample headroom and a healthy SNR.
But again, if you track Super Low, then boost later, you’re boosting Signal AND Noise.
@@G_handle What if the gap between peaks and averages is larger (say peak at 0 and average at -20)? Would this reflect poor mic-ing technique? Maybe a limiter/compressor would be needed?
I'm newer to recording/mixing (sorry if this is basic).
@@BOBopalooza Transient sources, say snare hits, are exactly as you describe. So first rule: protect the peaks. If the peaks are clipping your A-D converters, well they're clipped. (That clipped converter transient is Now your Snare sound. You may actually Like that, music is subjective, so you are free to break the rules, but as a rule Don't clip the converters.)
That leads to Headroom. In theory if you are peaking at -0.1dBFS then you're technically not clipping the converters. However, A) every snare hit isn't the same, and what happens if the drummer gets happy? Giving yourself 3, 6, 10, 12 dB of Headroom (meaning don't let peaks go above -3, -6, -10, -12dBFS) insures that you won't Accidentally clip. B) The metering you're using may not be telling you the whole truth. Transients are so fast that most meters can't keep up, both by physical limitations and design choice. There are many meters out there to check your average and peak levels, VU, PPM, ETC. and then there are "True-Peak" meters that factor in ISP (Intersample Peaks) that 'could be' in the signal, but that level of precision should probably be left for mastering, not tracking and mixing. Give yourself room to work.
Now, during Tracking and Mixing, back when the Multi-Tracks and 2-Tracks were on TAPE, best practices were different. There was (and still is) a Noise "Floor" and a point where the signal Distorts or "Ceiling".
With the Noise Floor, every piece of Analog equipment has one, then and now. Whatever circuitry is in your analog signal chain, before it reaches the recording medium, the noise floor is set by the "lowest common denominator" ( or the device with the highest noise floor. In a professional studio full of high end gear that usually was the Tape Machine and Tape itself, so many systems of noise reduction were employed to squeeze out a few more dB of "signal to noise ratio". But the main tactic was to record as hot as possible, placing the signal as far away from the noise as possible. So get away from the floor.
But that meant getting closer to the ceiling. Now what happens at the ceiling for analog and digital are totally different. I'll limit this to Analog Tape and Digital Converters. When you hit the Ceiling of converters, they are pretty much perfectly linear up until 0dBFS and then they clip. Analog magnetic tape though Changes dramatically based upon its input signal level, and has many different artifacts at different signal levels (and tape formulations and settings and speeds and biases and...).
The primary point is this: When you push a signal up into your Converters, the signal is basically flat and linear until it clips. When you push a signal up into Tape the signal goes from mostly flat-ish and pretty linear to clearly altered frequency responses, compressed dynamics, and obvious harmonic distortion nicknamed "saturation".
Most Engineers don't like the sound of digital clipping.
Most Engineers learned to Love the various types of artifacts that come from Analog distortion/clipping.
Magnetic Tape at its absolute best reaches 90dB of dynamic range, but is mostly 80dB DR.
24-Bit Digital Audio is theoretically 144dB DR, but in practice the best converters get to about 120dB, average...say 110dB.
With tape though, you would Push into the Ceiling and maybe get another 10-20dB! away from the Floor depending on how much saturation you like.
With digital, there's basically nothing good at the ceiling and you should likely will keep 10dB of Headroom to make sure you never find out.
(The 24-bit SSL 2+ converters I just bought for less than $250 bucks Obliterate the 16-bit Digidesign converters I first bought in 1996 for many thousands of dollars. The chips keep getting better, just like your smartphone. Tape remains the same. And BTW 16-bit has a theoretical DR of 96dB which is part of why most Analog engineers Hated the switch from recording on tape to digital at first. Many attributed the deficiencies of Those converters to Digital Audio itself. However I think today Most engineers would be happy to never touch another tape machine in their lives, except for funsies!)
So, long way of saying...
Just don't clip your converters, give yourself some headroom, you have Plenty of dynamic range to spare.
@@G_handle wow, thanks for the awesome response! That was a great read.
Are the analog "pushing the ceiling" harmonics you describe similar to what's happening when distorting a tube guitar amp?
Have you used any "tape simulation" plugins? I'm curious how close those are getting to sounding realistic.
And to your snare example... it's good knowing the noise floor can accommodate an input averaging, say, -35dbs. That just sounds really quiet coming out of my monitors, haha.
Is there any reason to buy a high end A/D or D/A converter these days? I'm wondering where you'd even notice a difference.
@@BOBopalooza 1) Glad Im not typing into the digital abyss.
2) Exactly. Push a 1k sine-wave into any analog circuit and at some point the circuit won't be able to handle the voltage. On an oscilloscope that perfect wave will change shape and look "distorted" hence the harmonic distortion. Basically a failure of the system, that we use to our advantage. A bug turned into a feature in code-speak. The labels or nicknames for all the different phenomena are less important than understanding that Every system has it's technically cleanest operating level, a lower level that's too much noise, and a higher level that's broken up and distorted, basically its failure point. Ideally you would take Every piece of equipment and Plugin and push signals through for yourself to see where its too noisy, cleanest, and if when it starts to distort, when is it Adding something that you like, and when is it too much. Electric guitarists in particular do this as regular course, and half the gear they use is for its desirable failures.
3) So many amazing tape plugins that I never need to see tape again. (Chow is FREE. Hornet is cheap. Softube goes on sale often.)
4) In general, track your peaks hot but with headroom. To the original comment, there still is a noise floor so you still want to stay away from it, but you don't (in digital) want to (ever) touch Zero. If you record too low, then have to boost the signal in the mix, you're boosting the noise with it. The SNR remains whatever you captured. Better to record hot, then turn down for the mix, decreasing signal and pushing the noise floor lower.
Basically, in my opinion, all the modern converters from reputable manufacturers are fantastic now. If you don't know if you need to upgrade, you don't. Definitely not for Recording and Mixing. Mastering...maybe. But at the point where you need mastering converters, you should consider letting a Mastering Engineer take over. A $2,000 interface two channel interface would buy you quite a few mastered songs, and with each one comes more than just the final product. You get to see, or rather Hear what they hear.
thanks alot :)
When will you guys upgrade and allow us to surround mix? I love the DAW but how is reaper ahead of you all with this?
Reaper is ahead with many things to be honest, especially when it comes to system resource demands, and yeah probably surround mix too if you need that. But to me there is a reason after trying them all for many years I have settled in Studio One. Every DAW has flaws, but Presonus has so much going for it, productivity wise, comfort wise, they listen to their users, and they don't re-invent wheels (oh hi there ARA :p). Studio One feels like home, and it is getting better and better all the time.
Nice tip and a good observation. As you said, getting it right/close to start with is probably the better bet, but if you need to fix just that little bit of clipping without trying to fix it at all of the tracks/channels seems reasonable (and of course not having to reduce anything by 96db :) ). I also thought of something while watching. I usually have it set as your example shows with all of my tracks going to busses and those busses going to the "Main" hardware output. I've seen others who add a Master Bus that then goes to Main with all of their Track Busses going to the Master Bus on the way out (which is how I used to have this setup in Cakewalk because that's just how it defaulted over the years). Since the function of a Bus vs the Main is different at some level, do you have any advice about the benefits or uselessness of adding this extra bussing layer? Example, a Bus can have the Input Gain and Polarity where the Main has to have the Channel tool added to get those features. Is there anything else "special" about the Main?
Must. Give. Thumbs. Up.
I was thumbs up number 500 : )
Never once called Gregor, "Greg" in my head. Not once.
This was fun 🍻
It’s okay, Joe an experiment understood by many. We’re not all Andrew Scheps! Still love ya though, Scheps! 😀
Fun experiment! 😂
Recording into a 32 bit float not integer audio interface would solve a lot of these problems .
Hey Joe.....why my songs sound great played on a speaker but when i play it on the speaker integrated on a cell phone sounds sturated?????
It's what I do now anyway I think you said it before? ..."just slap mixtool on the bus" ..always felt I was cheating but not anymore Thanks..The mind is a weird thing right!
A question: I've been working with the console shaper lately, and as far as I can understand using the gain/trim knob comes after it hits the console shaper (but before inserts and sends). Is that correct? If so, this will not help me to get a controlled crosstalk, would it? I mean, one channel could still be super hot into the console shaper, right?
If I'm correct: Could you guys at PreSonus please move the gain control "up"?
Excellent, thank you very much. There are simple things that do not occur to us because of their simplicity, genius has two forms, simple and complicated. another thing, i don't know why when i use fl studio in vst mode in studio one 5 i can edit audio but when i export it even if i bounce internally, for some reason i don't have audio, and the same if i have a lot of tracks of a great arrangement the audios of fl studio when i export, this only when i export or i bounce internally Ctrl-B the audio is cancelled, you only hear it in midi but really there is not. thanks, if you could help me it would be a great relief.
How can I get a person's tshirt?
Please forgive me but what version are you using? My Studio One Sphere does not look like what you are using please help
This is the latest version. 5.something
I was a recording engineer in the 1970's - it was all about balancing and levels. Gain Staging seems to be a "thing" now.
Gain staging is a thing now because all the plugins which have been designed to emulate the gear you used back then, are designed to operate best at a certain gain level.
64-bit floating point math didn’t change the fact that Gain Staging is still The Thing.
I wonder if we explained voltage rails, SNR, and headroom if the mouse clickers would spontaneously combust.
As a Recording Engineer in the 70s, I’m sure you tracked to tape every channel as hot as you could before it broke up to avoid the noise floor.
When I got that 2-inch with 24-tracks of hot as hell signal, the first step of the mix is to Gain Stage it at the inputs of the board to “nominal” calibrated at 0VU RMS which inside of my board is -2dBu.
Then we Gain Staged to unity through every piece of equipment in the chain.
Now no more tape machines, every track is recorded super low with tons of headroom, and the first step of the mix is to Gain Stage everything Up to nominal, -18dBFS, and often into a Tape Machine Emulation Plugin.
Hardware inserts, and Analog Emulation plugins all over the mix, and we’re still Gain Staging and Level Matching as much as ever.
@@G_handle we used our ears and watched the VU meters - it just didn't have a fancy technical sounding name like Gain Staging. That's all.
@@garyshepherd9226
Welp, I learned Gain Staging in the late 80s.
Both in the Studio and in Live Sound.
Not sure when the fancy label came about, but it was long before TH-cam.
I suspect, as with most things in Audio, the term comes from the broader Electronics field.
The capacitors, resistors, transistors, transformers, tubes, diodes, integrated circuits, discrete operational amplifiers, etc. don’t really care what signal is being passed through them, or whether or not when patched into a transducer at the end of a long series of components, it generates a “sound” that subjectively sounds “good” to any one human ear, or billions across generations.
Each component simply says “If you Don’t put this into me, I won’t do anything. If you put this to this in, I’ll put this out. However, if you put This in, I’ll blow up!”
All those metal boxes, faraday cages really, whether shaped like a desktop to be sat at, or maybe a standardized 19” width of various unit height to be racked together with other metal boxes, they all had (have) some cocktail or electronic wizardry inside. But luckily, the electronic engineers who concocted these works of art to help other artists produce their own art, thought “ hey, maybe the end users will want to “interconnect” two or two-hundred of these boxes together, in unpredictable ways. And we already know that every component, in every circuit, in every box has a ‘not-enough, too-much, just-right’ pain threshold. So we better train these guys how to do that, as well as give them some basic information about each box, and give em a bunch of different ‘controls’ and relevant ‘meters’ on the outside of the box, so they don’t get themselves into too much trouble. We know these music guys are hopeless, but the broadcast and live sound guys seem to really want to learn this stuff. If they do, we electronic engineers should call them audio engineers.”
Bill Putnam, Colin Sanders, Saul Walker, David Dearden, Paul Wolf and the King of Kings himself, now 94-years young, Sir Rupert Neve, all spent their lives on a quest, in search of “The Perfect Circuit”. Billions of records, and broadcasts, and live shows have passed through circuits that were the product of their productivity.
Whilst the code-writers have done some amazing things, 64-bit floating point math being one of them, there might be a reason, other than nostalgia, that every major code-for-audio company is desperately trying to reproduce in code, all of the sonic qualities of analog circuits. And hint, the unlimited dynamic range, whilst useful in the background, has little to do with the price of tea in China.
Two other observations: (I really am trying not to go too deep)
A) The two main tools of the audio engineer remain: EQ & Dynamics (specifically Compressors).
Both are used ultimately to control, and primarily REDUCE...Dynamic Range.
A compressors government name is ‘Dynamic Range Compressor’.
A limiter puts a ceiling on how high a signal can go. Raise the fader all you want, and your signal will either get a haircut or get beheaded.
Gates and Expanders work on the bottom end of a signals Dynamic Range, usually to cut it off at the ankles.... or knees.
EQs, formerly and formally known as Equalizers, seem to only exist within the Frequency arena of a signal, but no...
EQs are Static Dynamic Range controllers, divided into 2 or more frequency bands. Imagine a 1/3-octave Graphic Equalizer: on it you will find 31 little tiny faders, each of which does the same thing as the 100mm faders on a mixing console, except they only boast or cut a specific frequency band within the signal they receive, regardless of that signals input level, aka static. There’s no minimum threshold. In “Subtractive EQing” you run a signal through the machine, whichever frequencies you don’t want, cut em, anything popping out, tuck it back in. You whittle and carve away at whatever walked through the door, trimming the fat, nipping and tucking like either a butcher or a cosmetic surgeon. In the end, she will hopefully be a more beautiful version of herself (disclaimer: beauty is in the ear of the beholder), and when she’s back out on the street, all will be stunned and none will be the wiser. Or....You can go nuts and while she’s unconscious on the operating table, aka mixing console, start injecting her with 1000-CCs of this, and two-pounds of that, you can use those scalpels to open her up and start Implanting foreign objects into her body with no regard for her natural beauty or health or what she looked like before your hands all over her. Now she’s Jessica Rabbit, and trying get her fit in the mix is like trying to walk into a party with her in a hot red mini-skirt without being seen, or in this case heard. Whether you’re producing a feature film with an ensemble cast of glamorous Hollywood stars living their best selves, in the best wardrobe, hair, and makeup money can buy, or you’re making a low budget porno movie with impossibly endowed actors that can’t act (I could go on), you’re still going to package this hot mess into a box of some sort, and that medium of consumption will have dynamic range limitations. Whatever sounds you have coming out of your speakers, the loud sounds are gonna have a relationship with the quiet sounds and with the silence. Ultimately, the Dynamic Range of your Mix, turns out is a big part of the quality of that mix, both Objectively now measured in LUFS, and Subjectively now measured in Downloads.
B) It’s ironic to me that the same people excited about unlimited dynamic range, are the same ones most likely to be on the enemy side of the loudness wars! They seem to want to both put in sounds with +/-96dB of epileptic convulsions, and THEN run limiters at full throttle until their waveforms look like their DAWs are literally shitting bricks!
For some strange reason, the antiquated products of those aforementioned ancient men, now command more capital to acquire than when they were first released (Rupert was famously amused by the prices people were paying for 1073s, which he considered the best he could do at the time with limited resources).
I’m not sure what they’d say about 64-bit Floating Point or unlimited Dynamic Range, though we know what they tried to do with +/-15, 16, 18, 24, 40, even 60v Voltage Rails and >+28dBu Input/Output Levels.
I’m also not sure if they’d be angered or amused, but I think they’d all be a little relieved to hear that:
Gain Staging seems to be a “thing” now.
(Please don’t take offense to this. I’m only being facetious and smart-assey in hopes that others who stubble upon it may be mildly amused enough to be slightly informed. I’m sure you’ve produced vastly more and better actual music than I, as music has always been my friend-with-benefits, never my wife. And...my name is also Gary. And the international board of Garys clearly stipulates that one Gary cannot be angry with another Gary. My fathers name is also Gary, making me Gary Jr. And actually, if you are mad at another Gary it should be him, he’s the one that put a soldering iron in my hands at 5-ish, and he’s also the one that taught each the dark arts of talkin’ shit! Garys of the world unite! VU Meters FOREVER!!!)
@@G_handle This Gary is definitely not angry! I just worry that all the less knowledgable hobby recordists will think they have to learn Gain Staging - they don't and most will never ever need it. I know the pros getting external files to mix do need this - but it really is more a pro thing. Have a great day!
This is how I normally mix in my DAW. Clipping at the output of the individual outputs of the tracks make no difference in the sound quality, as long as you don't let your master output clip (by using a gain utility to reduce volume after). That is why I prefer using 32 (or 64) bit floating point as much as possible.
Normally?
Why would you do that?
@@G_handle Because you don't have to worry about each track clipping, only the master channel is importing for this.
@@BuenoLaidley read my other comments in this thread, but:
Just because your new digital freeway has no real speed limit anymore, doesn't mean your Analog car can suddenly do 200mph. Nor can it turn corners at high speeds. Nor is it on the road alone.
If you use Anything Analog or Analog Emulated, or if you use Any processing that is Threshold based (Compressors, Limiters, Expanders, Gates, Saturation, Distortion, ...) Anything that is Dynamic, at any "Stage", you're gonna want to Gain Stage for that Stage no matter what.
@@G_handle I have been mixing for years now. I know that the level going into your plugins must be gain staged properly, but the level that is post mixer doesn't have to be below 0db.
Presonus when will you add shortcuts for the input controls? :(
Thats not relly "some" forgiveness, thats 4x24=? Db worth of fogriveness.
Question so when i have all my tracks at -12dbs is there a db level that your master fader should be at b4 adding plugins to tracks
How do I know what the sweet spot is when it comes to volume while using different plugins? What is the optimal volume/gain going into a plugin? Obviously, it's different for different plugins, but how do I know?
I've personally never worried about it, and I'm happy with the way my mixes sound. I think too many people get caught up in doing "correct" gain-staging and at the expense of learning to listen and make good decisions based on how it sounds. - Joe Gilder
@@presonus Great, thank you so much! 🎸
Would setting the gain knob at the top of the channel to -24 be equivalent to using the Mixtool at -24 to accomplish this? Is there an advantage to doing it one way or the other?
Exactly the same. Though if you’re using + or - 24dB, something else is wrong.
I like joe’s music taste... he is a lot into more natural music rather than in full on EDM.
Great - SO is the best 🤗 good tip but you said it at the end - not use this for reording! 😁
Oooof, so close to first!
I've been telling this to people for years and, my god, don't they argue!
There’s a reason why they argue.
I want to see the comments from folks who left before 8:52. 😂
🤦♂️ spreading the idea that you don't need to learn proper gain staging on a mix channel. It's only the first and most important thing people should do on every mix. wow
If the end result is the same, there's no harm. 😊 - Joe Gilder
Not disrespecting the presenter but that's really old news. for example here th-cam.com/video/CGRusg9GnAg/w-d-xo.html
where he also shows the behavior of analog modeled plugins.
So should I switch to 64bit float or not?
its native in S1. You can't switch to it.
@@MellowXBrew No. Theres an option in settings.
@@technoscout yesterday you’re right. I thought it was for something else. Here’s a forum that addresses pros and cons forums.steinberg.net/t/32-bit-or-64-bit/98651/24
hi joe