Don't you love when you've been running audio systems for 40 plus years and within 5 minutes a 20 year old kid has already started to teach you something.
@@AudioUniversity yeah I was worried that I might have been peeking my Daw but my cakewalk sonar maxes out at zero on the meters. They turn red about -6 so I actually run a little in the red just like a VU which I like. Someone else was arguing with me about running my d a w m in the red and I'm like no man I can hear clipping especially digital clipping I can hear that s*** from across the block LOL
Haha! Digital clipping is hard to miss! In reality, digital signals in the DAW sound exactly the same regardless of level until they actually reach 0 dBFS and start to clip. That is, unless you’re using plugins specifically designed to for non-linearity. Of course, signal to noise ratio varies at different levels, but there’s A LOT of room for error in the 24-bit digital realm when it comes to noise.
Teaching is a gift! No matter what you do in life, use your gift kid! One thing that’s hard for us older guy’s to wrap our heads around is the amount of people that are into audio these days. It used to be a secret society but not anymore. Plus, Re- teaching the fundamentals is …. Fundamental.
This video finally answered "stupid questions" I've had for years, such as "I want playback to be louder - do I turn up the volume on the audio interface, the computer, the guitar, the DAW faders...?" I think people are too embarrassed to admit that they don't know how to properly set up everything before they even hit record, and you even gave us suggested % levels to work with. Thank you so much!!
I have watched literally every video on youtube regarding tutorials on mixong and mastering etc and I can say with full confidence that this chap is by far the best mixing and mastering teacher that I have ever come across - so happy to have found this channel today - thank you so much for the superb content!
As an industry veteran of 45 years, I feel like you glossed over the importance of setting the gain of the pre-amp correctly. The goal of a pre-amp is to take signals that are microvolts (mics and DIs) and boost them up to line level. Every source requires special attention. A vocal mic typically needs more gain than a kick drum mic. Once you've mastered pre-amp gain, everything else downstream operates at line level. For me, the pre-amp is THE most important element of gain staging.
Technically, I have a microphone that comes with line level output built-in, but considering that that's probably a pretty rare thing still, even these days, I'd still have to agree with you... I'm only a volunteer soundboard operator and have no training in the subject though, so my knowledge is almost nothing compared to all you industry veterans out there, and let me just say thank you to every one of you no matter what part of the industry you're in, because even in my volunteer status, I know a bit about how hard an A/V guy can sometimes work and STILL not be 100% sure if it's my fault or the equipment's fault, when something happens. Let's just say, "25 year old soundboard" and "channels going out", and you'll probably know exactly how bad of a time I had for the last couple of years before 2020 or so when our church got enough money to have our audio installation company we use for our technical work, actually finally get and install a new soundboard. Technically, the installer's parts source didn't have the requested soundboard available except for a wait-list or something, so the church had to purchase it from sweetwater themselves instead, to meet the installer's timeframe for fitting us into his free time, and literally everything got upgraded right down to the soldered on 1/4/XLR combo plugs in the walls. IDK if soldering wall plugs is standard, but if it's anything like a standard outlet is installed, usually, then I must say that I didn't know that anyone actually went through the trouble of soldering connections in the wall socket. And those combo plugs, man... Before, we just had standard XLR jacks, but to now have that ability to use it for literally almost ANYTHING that plugs into the wall, well, it's kinda cool honestly, and also, to have not only our original 16 channels, but 32 DIGITAL channels, on our new soundboard, while still being smaller than that monstrosity in my barn I'm trying to decide if it's even worth it to try and run a can of DeOxIt through the thing to try and get it fully, or even partially working again, considering at least half, if my memory serves, or more, of the 16 channels, are either dead or going that way, well, again, that new mixer and all the stage plugins, just, WOW... For reference, I had been saying that I thought the old soundboard needed replacing, around 2 or 3 years prior to them getting it replaced, so, even if it wasn't a headache at the beginning, well... I said it needed replacing, and if towards the end of it's life/usage cycle(after I mentioned it needed replacing at some point preferably soon) I was ending Sunday morning (on nearly every Sunday) nearly having PTSD from the issues and errors I couldn't quite figure out,(not even joking) was what it took to get them to take up a collection for new audio gear, then I consider my battle scars worth it, and trust me, for me, an audio "un-engineer", it WAS a battle, one that more often than not, I felt like I was losing, and was feeling well, hopeless, to be honest... Again, no matter what all y'all industry professionals and whatnot actually do in this space, thank you all for your work, as it there's one thing in my top 3 that I could probably have used therapy or other "professional" "help" for, the experience I had with the old soundboard was the only one I can talk about, and I don't know how anyone doing this every day can take the pressure...
@@weneedtermlimits assuming you mean the "old" one in my barn, we replaced in around 2019-2020, I don't remember the brand right off, but I think last time I looked at it it had "Peavey" brand logos on it, and it's an analog (no screen, just a million pot knobs and sliders) mixer, made in around 1994, and most channels are either dead or dying in one way or another. I'm not a technician by any means, that would have the knowledge to fix it, so you'd be on your own there, but other than that, if you need one for parts, I suppose there's gotta be SOMETHING useful on it still. I know that it's a 16 channel mixer, and I know that it's got tape player "RCA" connectors for aux inputs, plus it's got separate studio/XLR jacks on each channel. I cannot stress enough the fact that it's over 20 years old and I don't have the experience to be able to diagnose whether it's fixable or even if it's worth fixing. Anyone buying or otherwise getting it would be getting it "as-is" with no warranties... Also, I'm located in Minnesota, if that information helps... I'm honestly not sure what to do with it, because I don't have the speakers, cables, etc. required to do appropriate testing with, and even if I did I'm not an audio repairman so I'd be out of my depths doing anything more than just spraying a few cans of DeOxIt all around the board...
Nice and simple instructions, good job. As a retired advanced audio instructor in southern California, it's nice to see this still being taught. DAW's have made recording easier, but the lessons of tape still apply.
Recording to tape was falling out of fashion when I started in the industry. To be honest, tape is a bear to work with but has a warmer sound in the booth compared to DAW tracks. BUT I was raised on 45’s, cassettes, CRT tvs and with corded phones. I may be a hair jaded.
@@redmcbeard4230 I agree with you Red. I started on Tape and welcomed the digital with open arms. The one thing tape did was really teach engineers how to mix and work with audio. What I have seen over the last ten years is younger DAW users do not get that knowledge using the computer to mix. I knew a Pro Tools instructor who would teach his students you don't need the faders to mix, you can just use plugins. He had no audio experience other than taking the Pro Tools instructor certification course. That's why I appreciate this channel.
Actually, gain staging with digital is totally different than with analog - with digital you're aiming for -18 dbfs, where with tape you go up to 0db or even a bit 'into the red' for a bit of natural saturation.
I've worked in radio for 35 years, and much of that time has been producing audio...ads, promos, imaging, etc. It's been a lot of trial and error. Thankfully, I've found this channel to show me what I've been doing wrong AND what I've been doing right! Thanks.
Absolutely agree.... I bring all channels down to -12 to start and adjust each channel accordingly to give me about -3 to -5 headroom. I leave master bus alone or sometimes just add a limiter to get to 0 but honestly I turn the my speakers of headphones up to hear.... but the truth is in my opinion mixing low is great for preserving your ears and if you hear issues low it will be more pronounced when you turn up the volume.
I just watched three of his videos. This guy comes straight to the point. No nonsense. That’s what budding engineers like myself like to watch. Great work bro. Subscribed to your channel
I work with analog and digital desks for live shows or services. Been doing this nearly 23 years now. If I am going in to a session without any pre work I’d do exactly what you did for my base levels on inputs and outputs. All my faders, mains (gas pedal) and bus sends are set to unity. From there I’ll line check each input by raising the preamp to a -10 on my scale and verify device volumes at 8 or 80% max volume. The first rough mix (usually my line check but someone’s band rehearsal #FML) will be getting preamp levels to an acceptable gain setting, basic PEQ/EQ of the individual channels then applying dynamics as needed. DAW recording is not my strong suit but it is just like working any physical desk. Prep before the show and use your ears, scales and RTA to even put the mix. Better dead than red on your desk/DAW. Good stuff and solid presentation. I’ll send this to a few younger folks I know that wanna learn the trade.
I operated audio for 5 years at tv news stations, yeeears ago but never learned the nitty gritty like i should have - i preferred to master the video production switcher. but now that i work in IT for a school district i find myself filling in as the sound setup person for a lot of auditorium rentals. getting this knowledge will help us set up for better rental experiences. love it.
Everyone learns different subjects and in different ways. I've found I work best teaching myself with a bit of input from an authority on a subject. This channel has made a huge difference for me since I started teaching myself audio engineering, and gain staging is something I only recently started to acknowledge. Fantastic video, thanks a ton for making this.
I want to say thanks! I've always had vocal problems w/ FB & muffled tone, EQ turned down. I've always just set the amps to 50% and brought up the mics til right b4 FB. I just tried what you said and my lord what HUGE difference. All 3 mics are LOUDER than ever, EQ's are unity, not the slightest indication of FB. My EV's went from +6dB to -8 and my RCF's went from 50% to 25%. Freaking incredible headroom. I'm just really amazed
Outside of the studio in the live event venue I think that something that a lot of people overlook is the power supply coming into each of your components. I run a mobile DJ business and use a APC G5BLK rack mounted power conditioner (not saying that this is the one to use, just what I have). I firmly believe that your audio is only as good as the incoming noise from the electrical supply lets it be. While in a studio (yours or elsewhere) the electrical supply has probably been looked at for noise issues in a mobile setting you have no control over what is coming out of the outlet you are plugging into. Whether it's over voltage or under voltage, or the central AC being on the same circuit it's a crap shoot of what you are getting IMO. I know that when I set up I've dealt with the power supply issues to the best of my abilities so as to not introduce unintended noise into the mix. Side note: This is truly a great series of videos that have been pro for every ones benefit. A person should never stop learning, and if you think that you know it all it's probably a good time to throw in the towel.
I completely agree. I have a Meyer system for outdoor live shows in Little League ballparks and for street fairs. The first thing I ask about is POWER. If I can't put each of my subs, and each pair of tops, on their own, separate 20A circuit then I won't bid the job. I also won't bid a job if the venue was built any time before 1974 (the US used aluminum wire for 10 years because the Vietnam war used all the copper for munitions). I won't do shows anymore in San Clemente, CA, for example, because the infrastructure was built in the 1920's thru 1940's and there isn't any 3-phase power in the entire city. The only way we amplify sound is with electricity. My loudspeakers love electricity and when they're well fed, they sound awesome. When they're not well fed, they sound like JBLs.
I love sound design. I love learning about it even more. Thank you for this channel. Never ever stop educating us. Bad sound makes me sick literally. You are the doc l have always needed.🙌🏾🙌🏾
Thanks Kyle I have understood your points well. I am NOW testing gain staging my guitars, Dynamic Golden Age Mic and EZ Drummer 3 drums to record 1st songs. It has taken Years (2016) to build this to do this. I have Great 1st timer Equipment by Research everything in the audio chain. So excited to start this !
Just remember: Our hearing judges loudness mainly by average levels - closer to RMS - and not so much peak(transient) levels. That's why it's good to use a VU plugin that can be calibrated so 0dB VU approximates -18FS. There are also pure loudness plugins available, or might already come on your DAW, that measure in LUFS(loudness units, full scale).
This was the best video I've seen on this topic. I hadn't given this idea proper attention, that noise can increase in tandem with harmonics when applying a lack of source volume output combined with additional gain and/or excessive gain (amplifiers, exciters, etc). I have noticed that too much wideness on instruments can reduce head room as well. I found this was a common issue starting out but now I'm finding that it helps keeping instruments down 30% to 50% volume in the beginning stages while adjusting and mastering (compression/EQ/etc) afterwards. Thank you for making these topics so comprehensive!
Good clear summary of gain staging. I usually aim around -12 give or take for my tracks and uses a limiter on output bus to bring levels to max 0 to avoid any clipping. I know if you record a bass too hot it can be difficult to tame the sound with compression etc.
I've been doing live sound for almost 34 years. In the old days, I was taught how to calibrate a whole system... From the input and output of a console, to the eq, to processing, to the crossover, and to the amp. A lot of the old systems that I cut my teeth on, you could tell that care had been taken to pair gear with proper metering through the various points of the system. A fair amount of time would be spent matching gain through the entire chain. I rarely see people do that anymore.
Its actually crazy how much of this I actually knew without being able to properly explain it. I was expecting to take some new tools away from this, but I really only learned how to explain the reasons I do what I do when I'm setting up to record. As for setting up gain, I find that the key is getting the pre set up right to begin with. Post gain adjustments are easily adjustable, where as setting up a good level for recording is only doable once. Especially when you take mic position into account. This is particularly important if there are multiple voices on one track as well, the closer I can get them on the recording the easier it is for me down the line to adjust. Not a lot of things bother me more than having two separate recordings that are miles apart in overall volume and having to try to make it all work. I generally make sure to do a few test runs, and do a rough EQ on the vocals to get an idea of where I need to be. I also tend to do a lot of spot adjustments to match the overall volume of the track itself within the DAW. Dipping the gain where things get quieter, and raising the levels when it gets louder. Usually it doesn't need much, but it definitely makes for a better overall mix in quieter sections. I find, personally, that the hardest thing is getting the EQ right. I definitely spend the most time on that.
@@PrisonTorture34 Can be, yes. First, to clarify, -10 to -12dB FS (most mixing boards use a dB VU meter instead which often tops out at +16, more or less, depending on the board). And in some cases for film, the target peak level may differ depending on what you're recording and whether you aim to make a usable mix or are just recording isolated channels. But generally, for interviews where you have 1 or 2 people, I'll often aim for peak levels at -18 on quality recorders, maybe a little higher on lower quality recorders where there's more self-noise.
You are correct lad, I have applied that Strategy since the Analogue days and my mixes are clean with plenty of headroom. If you are near South Carolina, I sure would like to have you over at my studio. Thanks
Taking the time work within those standard limits pays for itself over and over. Thank you for making this, as there are a lot of people that don't understand the concept. I've discovered, by accident, that proper gain staging makes a superior and flexible product. My mixes can be played on nearly any equipment at high levels and still not clip. The idea behind this is that the quality of the reproduction is completely dependant and limited to the quality of equipment used to play.
I love Joe Henderson too. I went back and listened to the entire video again and did not hear any Henderson used. But then, I saw it on his cell phone.. LOL.. Thanks..
Great, clearly explained video. Such an important topic as it directly relates to sonic performance. I'll share in the general conversation of system gain setup for PA systems... I don't know how many times I have seem amplifier inputs wide open for example, when often the correct setting is some fraction of that, depending on the amps internal gain specs. A good quick tool that doesn't require test equipment that I use that gets you in the ballpark for aligning system gain, is to unplug all speakers from all your amps first. Very important! Next, put on some decently recorded music that has good peak material such as drum kicks etc. Then, starting at your mixer's output, bring the signal up until you begin to get a decent clip flash on the mixer's meter then stop, and leave it there and don't move it... as this will be your reference. Then, adjust the gain on next downstream device's meter or clip light in the signal path, to replicate the same clipping flash as the mixer. (Devices that have a unity mark should remain at unity in most cases) Then, once that alignment is done, bring the main out level on the mixer back all the way down >> BEFORE
Usually I'm starting at 12 dB peak per channel on the mixing session as you said. If I mix in my arrangement session I'm turning down synths outputs to meet this requirement or I'm using gain knob in pre-mix section of the mixer in my DAW (I use Cubase). Also I adjust clip gains if it's necessary. That simple and it gives plenty of headroom on the master bus, making comfortable mixing environment and also it's giving nice headroom in mixdown for further mastering session.
I use Presonus Studio One, and we have a gain trim knob on every channel. I used to use a VU meter on the master bus and set it to -18. Then I would play each channel in solo, watch the VU and adjust. Later I started using a VU on every channel, but now I'm just looking at the FS meter, peaking at -12 dB. I also put a trim plugin in-between pluggins and/or last in the chain, because if I'm using any plugin in a more extreme way that changes the level, and the plugin(s) don't have a gain or level knob, I use the trim plugin to put the level back to -12 true peak. It's not a rule, but it helps keep everything in line, which means that I don't have to set all the faders extremely different to get approximately the same levels, e.g. one channel around zero, and another at -10.
Very good, concise video. Thanks. I usually just record in directly synths, drum machines and guitars. I used to shoot for -12, but I’ve noticed i get better results with -8 and then using gain reduction in my daw I will take it down to -12. I’ve revelry started mixing with the -6, -12, -18 rule recently as well. Again, thank you.
In the digital domain I use to set my tracks at Peak -3 dBfs (percussive Sounds) or -6 dBfs (sustaining Sounds) and then control, if they're round about between -18 to -12 dBRMS... adjusting if necessary. Sometimes I also use a VU Meter calibrated to -18 dB in the Main bus.
Really amazing video. I have been watching a lot of videos on this stuff lately and this was one of the most in depth and concise ones I have found. Plus the way you explained things just cleared so much confusion up for me, I feel like I have a much stronger understanding of things now. Fantastic work, thank you.
Mixing in logic I have everything (busses, tracks, etc) go to one mix bus, which I turn up or down as needed to avoid clipping the master buss (or I turn it up above 0 if I wanna hit the master buss compressor harder). I still think about getting individual tracks to a proper level to hit whatever processing I put on those tracks how I want it, but in terms of clipping the entire mix, this seems much quicker and easier than making sure everything is peaking at -12. It also allows me to absolutely smash some signals if I want without worrying about it.
Very good explanation! It should be the basic for all DAW users but it doesn't. It is so simple to gain stage after every (stage) plugin and it makes so many things more easy for the entire track.
I use a lot of external effects, mic pre-amps and other sundry devices for compression and signal coloring and work with an aggregate of interfaces connected to my DAW so it’s really important for me to pay attention to gain staging. Even before a session begins I like to use a 1k tone generator and pre-set the optimal gain stage level of all connected devices. I start with connecting and sending tone to the last device in the chain like a DAW interface or an input on a console feeding an amplified live room where the DAW or console faders have all been set to unity and setting optimal gain there until the tone reaches the optimal level I am aiming for and then repeating the procedure moving backward, connecting the next device in the signal chain and optimizing the input level of that device and then adjusting the output level so the tone level of the last device in the signal chain sits at the same level where I had set optimal level. I continue moving backward throughout the signal chain repeating the process until I reach the source like a mic, guitar DI, synth, etc. being careful to match impedances of the output level of each device moving up the chain and matching the impedances of output to input and switching the impedance on the output of the tone generator accordingly to match what those impedances should be and never mix-matching impedances which can cause much unwanted noise and sometimes catastrophic damage. Not saying these are the exact levels where I will be working in a session, but close enough to where a simple fader move (not gain level adjustment) is all that is needed to adjust the mix in the DAW session or on the console feeding the amplified live session or tape.
Precise and easy to understand information. I was always a little nervous moving into the digital mixing domain, so many choices and intelligent decisions to make
May I suggest that you delve into external analog signal processors getting chained up prior to a DAW? Too many get obsessive about using older analog gear when you can now get similar results with what is available in your DAW. That old analog gear and the physical connections typically add noise.
I try to have around -4db to -5db at the stereo out channel at max for my mix, which gives me enough headroom to master my song. -12 db will be a good starting point where I can bring a track down to that level and do the gain staging of other tracks around that. Really good content. Thanks mate 😊
I'm really loving the ability to put plugin UI's in tracks in reaper, and then putting the ZenoMOD VU meter in each one. This lets you adjust the gain in the track like you did with the gain plugin. It's awesome!
In Ableton I put a Utility (gain plugin) with -12 dB on any channel right after the sound source, to keep audio at suitable levels for the following plugin chain. I could make sure that my synths don't output louder volumes in the first place, but when stepping through presets, those levels get reset most of the time. By checking the inter-plugin peak meters, I also make sure that my plugins don't add a significant amount to the overall peak level of the channel. Adding up all the channels usually gives me a master peak level of around -6 dBFS, which leaves enough room for master bus compression or loudness optimization. As a bonus, I also put another Utility with +12 as the very last plugin on every channel and set the mixing fader of that channel at -12 for compensation. This gives me a little more headroom on the fader, as the +6 on Abletons faders does not leave much room to boost quiet signals.
This is not good ‘normal’ practice. You are reducing the insert level of your channel by 12db, whether it needs it or not, and you can be sure this introduces unnecessary noise when every single channel needs makeup gain. Much better to optimize your source levels whenever possible to the nominal level your DAW expects at its input. I’m guessing you’re not using preamps or any mixer. Lacking that, adjust the outputs of your sources as best you can so that your faders have the range you need. If you find it routinely necessary, make a habit of parking channel faders at -6 for setup as your new ‘zero’. Those two utilities you insert are acting as noise ‘anti-reduction’, the last thing you need.
@@artysanmobile This is true for analog audio processing, but I work completely in-the-box, so my sound sources are synth plugins or samples. I assume you are talking about quantization noise then? As DAWs (and probably most plugins) these days work with floating-point values (64bit mostly), quantization noise is negligible. As SNR for floating-point values are 192 dB (32bit) and 385 dB (64bit), just changing the volumes by a few dB has absolutely no audible impact on sound quality, regardless of where in the signal chain the volume is changed. The only source of quantization noise is when the sum is rendered to a 24bit integer wave file.
@@markuskopter I was just giving good general advice, in the box or out doesn’t matter, and I’m not referring to quantization noise but just noncoherent noise as we have in any electronic signal. By the way, you’re quoting a bunch of numbers that mean virtually nothing. Hundreds of dbs and 64 bit word lengths are just nonsense concerning audio. You do know that, right? I only mean to be constructive saying this.
@@artysanmobile I have to admit, I was wrong in stating those signal-to-noise-ratios were for floating-point values; they are for integers, actually. The SNR in floating-point processing is even much higher. I also was wrong in saying the final render were the only source of quantization noise. In fact, every calculation in a DAW has to be rounded to some resulting value, which results in quantization error, or "noise". In fixed-point systems the size of this error (the volume of the noise, if you so will) is determined by word length. Did you ever use a bit crusher plugin? It reduces the word length of the audio signal to deliberately create quantization error, which we then hear as noise. But other than that quantization noise, there is no inherent non coherent noise in the digital domain.
@@markuskopter I hope you’re just a little deep in the sauce on a thanksgiving day, enjoying life. You sure have a great story to tell. Hey, did you know that Africa is splitting into two pieces as we speak. Because that’s far more relevant to this discussion than the gibberish you just laid on me. I really do hope you’re ok, Markus.
Been using this method for years, but I've always had each track, send, master, etc. at -6dB as a starting point. Master level is never changed and no effects/sends ever go on the master either.
Very well explained basic concepts. It’s sometimes helpful to go back to basics. And you obviously understand very well what you are teaching, unlike other channels. Bravo!
There's another aspect of gain staging that isn't covered in this video that is of paramount importance to both live audio reinforcement and recording studio practice, and that is setting the maximum SPL output levels of the PA system or nearfield monitors. Bob Katz recommends his "K-System" for recording and that's a very good place to start: -20 dBFS pink noise should equate with 85 dBA SPL at the listening position, which makes 0 dBFS equal to 105 dBA SPL. When mixing live audio in small clubs, I shoot for peak SPL of no louder than 96 dBA SPL. The rule of thumb here is that if the peaks are making you wince, it's too loud. But it should be recognized that even 96 dBA SPL is technically too loud for extended exposure.
thanks, great info as usual my friend! Growing up in the 70's(yep, i'm old) I can tell you that the tendancy was always to push the limits of a hot level, and finding the sweet spot with your recording device for best results. Now, in the digital era, thats no longer necessary, and in fact its dumb to even feel the need to do that. My point is; this was the hardest habit to break for me initially, since the conditioning of 40 yars of that habit was so deeply ingrained...todays recording process has become a piece of cake to get a good quality recording on the first shot...not saying it's necessarily a better recording than analog, since both have their appeal, but it's definitely easier to get a better recording with digital... thanks again!
There is another inherent issue with having the master sliders too high or too low: the individual input sliders will not have much leeway to accentuate or attenuate the sound. It’s better to be able to control the individual input faders so they can move for a longer range and therefore able to make small and more accurate adjustments. I have seen many sound mixers keep the master faders very high and not being able to make much adjustments with the individual faders since they were placed too low or too high and thus had very little room to move up or down and that made for more difficult mixing.
Your videos are really good. I don't always agree with everything you say but you typically give an alternative explanation. Some RUclips channels have thousands of subs and I can't really understand why. Your channel is doing very well and I CAN understand why. You provide great content.
With that said, That Ashley EQ is awful. I know I sound like a gear snob but I'm simply keeping it real. Also- @8:40 There is not plenty of headroom. Your peak is around -0.5 One needs to account for FX being used on each track that will sum to the Masterbus. To be honest, in digital, even a peak of -0.5 is a little much.
I mean none of the gear he’s using is nice, a focusrite and a powered speaker aren’t nice either it’s just demo gear for the video. Those Ashlys get the job done I never had particular problems with them, it’s just I rarely use a rackmount graphic anymore
@@IWannaGoMissing For live applications I've used Ashley as well, but not cause I owned it, it's what was there and you're right, it got the job done. Last thing I want to do is sound like a gear snob, but the Ashley's had noise problems. May be more tolerable now in the digital age, being able to cut some of it out, extended noise floor.
Just came across this great info. Personally as I do mixing in a different daw than my production I prefer to setup my audio lanes to have a pretty low db FS at the start and slowly through busses and a pre master that everything is affected by get it back to the ideal threshold before mastering. But I've seen this method work to and I'm glad to see that I wasn't lying when I explained gain staging to my friends (though will be sending this to them too)
I've gotten slack about this with total denials but driving channel input gain close to or into clipping just to achieve a fatter sound on a snare for instance, slowly damages the input transistor stage. When a channel is driven hard the transistors are turned on for a longer duration and conducting current longer. This will cause a transistor to over heat and eventually fail. The same holds true for speakers and HF drivers. Having serviced a lot of consoles I could tell which channels were used for kick and snare as they were first to fail. If you want that overdriven sound use an appropriate plugin or get creative and send your drum mix into an amp, guitar amp possibly and mic it in a live room. Maybe Kyle has done a video on the benefits of even harmonics versus odd harmonics.
Thanks for bringing that to everyone's attention, Donald. I expect many will still overdrive their gear if it produces a pleasing sound, but maybe they will appreciate the experience more now.
As I watch this video, I am listening through JBL noise canceling headphones while laying on the floor and staring at the ceiling. But the headroom is more than satisfactory.
For all gain related work, I cannot recommend enough Klangfreund LUFS meter. It saves you a lifetime of eyeballing and guess work with levels and mouse clicks. It has multiple grouping options so you can level match everything according to Loudness in one set. You can gain stage/fade up mix really kick, compare dry/wet fx between tracks, and so many other things. It's very light on the CPU. Mike Senior brought me into this plugin.
Trying to set gain on multiple devices in a PA System can be a challenge! I have a Tri-Amped System. This consists of a Peavey IPR2 2000 for the HF Drivers in my Peavey SP4V2’s and an IPR2 5000 to drive the dual 15’s in each cab and an IPR2 7500 that takes care of the Peavey SP218’s. Before the amps we start with my iPad, which goes to a Yamaha MG12XU Mixer, then to a Peavey 231 Dual Channel 31 Band Equalizer that I use for the Low Cut Filter to protect my subs, then to a Peavey 35XO Electronic Crossover that feeds my amps. All devices have Gain for each channel and I’ve never been able to set each one to what I think should be the best level because I’m not sure what the level is. I’ve always left the EQ at 0dB, as well as the X-over and left my amp gains maxed and make my adjustments on the mixer. How can I get this system tuned in the proper way so I can achieve great sound and not have to worry about clipping?
I recommend keeping the meters along the signal path at a "nominal" level. For an analog meter, I will aim for peaks at +3 dB. For a digital meter, I will aim for peaks at -6 dBFS.
@@AudioUniversity Thanks for your reply. I’ll give it a try and see what happens. Too many Rock concerts in the 80’s is to blame for me wanting the music loud and clean. Like muscle cars, no matter how much power you have, eventually it just isn’t enough.
i've been using the K-14 system for a little while now & have gotten pretty good results. i like to adjust the levels going into the mixer channels so the signals are peaking usually no higher than -10dBFS, maybe up to -8dBFS for drums or percussion. then, once i've gotten a decent rough mix going, i will apply EQ & compression to my drums & bass, getting a good balance between namely the kick drum & the bassline. i follow that up with soloing the kick & taking note of its level on a VU meter. next, i'll bring the bass back in & make sure that each time the kick & bass hit together, the VU level increases anywhere from 2-3dB. once i have them balanced well, i send them to a kick & bass bus for glue compression. finally, i'll mix the other elements around the kick & bass, while keeping the kick & bass levels together throughout (ie; if one changes, they both change, etc.) using the K-14 system, i'm able to get a LUFS reading of around 10-12 integrated after pushing my limiter on the master up by an available 6-8+ dB, while still maintaining a -1dBTP.
Honestly these videos are more educational than when I went to SAE in Athens. That place is full of people who tell you half baked knowledge and don't even cover the full picture or go in depth as to how things are done.
Don't always set your Master to Unity 0db. If you're working with a massive live sound system which isn't properly attenuated at the amps you're gonna be turning Channel faders up only slightly before it gets really loud. Low faders means 2mm moves give you a 5db difference, so you're better off with a Low Master, High Channel faders for easier fine tuning. This is situational and only an example of venues I've worked in.
If you can’t adjust the amps to the proper output level for the venue, I agree. Use the master fader as a volume control. However, it’s probably best to turn the amps down when possible.
You can have any gain on your daw tracks, even you can have visible enormous clipping of the signal and it wouldnt have any involving to the result signal if you take care about master bus volume. It's almost impossible to overload internal daw bus cause all of the modern daw have 64bit mixing algorithm. The only place when you can get clipping of your mixes it's the master section and hardware output of your sound card and etc. However there are many plugins which require certain input level due to their processing algorithms(overdrive, distortion and other saturators) so you have to follow the level to get proper result. But it is quite all right if you get some clipping on any internal bus of your daw except master bus.
This was the best explanation of gain staging and its application I’ve ever watched….and I’ve watched a lot trying to figure it out. Thank you. I’m still confused as to why some of my mixes are louder (kick thumps more or snare slaps better or bass is deeper and more upfront) on some tracks more than others even though I perform gain staging in the relatively same way each time? Then when I remix the quieter tracks to match the louder tracks output, I clip! It’s so confusing! And I haven’t even tried to mix on my Toft atb32 yet!
First: Kyle...your videos are absolutely VERY good in all aspects!! Thank you for this hugely helpful and detailed information. I'd like to ask three questions here: 1: Am I right when I conclude that - given the natural limitations of the human hearing spectrum - I will never escape the fact that those bass heavy instruments NEED a significant dB advantage (higher peaks on the dB meter) to ‘stand out’?...even after some (deliberate or accidential ) saturation? 2: If so: can I solve this problem by simply lowering the gain on all the other tracks? (and get the volume back in the mastering stage?) 3: Does it make a technical difference if I choose to lower each and every single track individually or save myself some time by lowering the gain of the whole groups that they belong to?
Are you familiar with the "shoot to the right" approach in digital photography? If not: it's basically setting the exposure to shift the resulting photo's histogram as much to the right (brighter tones) as possible without clipping. (Or at least not clipping any part of the photo you care about.) That increases the signal to noise ratio in the recorded image, since the underlying noise (thermal and otherwise) of the sensor is constant. Then you pull the exposure back in post which reduces the noise along with the signal. It also makes for smoother gradients, e.g. if you take a shot of a dark scene where the recorded pixel light levels range from 10 to 100 you'll have 91 distinct levels of brightness recorded, whereas if you increase the exposure so that the range is from 20 to 200 you'll have 181 distinct levels. I imagine similar effects could be seen in a digitally recorded sound wave. Anyway, it seems like you're kinda doing similar things here with digital audio recording, and I thought I would share that. It does make me wonder if anyone has made an effort to come up with an algorithm that can take a number of tracks and their recording & mixing parameters, and output a suggested optimized mix for preservation of maximum signal precision, or even suggest better recording parameters for the next take. It seems like it should be mathematically possible, though I don't know off hand how much number-crunching it would take.
For studio, When I'm tracking and mixing, I try to keep my mixes at around -20 lufs which allows me the headroom to reach around -12 lufs in mastering. Great video. I was curious about what you'd say for the live application. I had been keeping my foh speakers at a certain level and controlling overall volume with my master fader.
In integer PCM (traditional digital) you have an additional problem with audio recorded extremely low. Not only when will noise be a problem when gaining up, but the signal will have no resolution, so you may up get digital aliasing introducing weird sound. With a good fp32 (float) PCM workflow, everything is much more forgiving and going above 0db is no issue. Maybe it is a bit of an abuse but my mixes typically always go above 0db and then is rained in by a limiter on the master bus. For spoken word / podcast that sounds more than well enough to my ear. For video work I’m hoping we’ll see everything transition to floating PCM within the next few years so audio work becomes focused on capturing the best audio (best signal / most dynamic range) instead of throwing reserving headroom. Let the ADC capture great audio without managing gain staging much :) A lot less stressful not having to worry about clipping or recording too low. When you don’t have an audio guy on set and need to manage video, audio, capturing PC output, supporting talent and what not, traditional error prone integer PCM gain staging is an additional stress factor I’d prefer not to deal with. A great sounding FP32 ADC built into every camera XLR input would be great!
My advice: Most people listen to music off their devices like phones computers etc, bounce your project and play it off a normal device to compare before considering your projects done there are many times I've done this and notice massive volume differences, but bounce it as an MP3 as the noise increases a lot more which gives an accurate perspective or just get some grade A monitors bounce is on your computer as an mp3 and play it back, be aware that any gain you change when recording is destructive though and no amount of compression will change it as the track has already clipped.
I never consider my tracks done until it has been run through and sounds decent on a portable BT speaker, a phone, basic ear buds, and good through PC speakers and car stereo as most music is listened to in the car for your average user. I have learned the lesson of hearing a mix out of my monitors and being proud, then listening through literally anything else and being embarrassed and saying, "Wow, that sounds like absolute, grade A dogshit."
It’s odd how “simple” it is conceptually yet as soon as I’m listening to other music that has a lot more volume it always ends up with trying to match it. And bye bye goes your headroom. Or working with things tracked on different days etc where you can just end up in a crazy battle trying to level things out or automating like mad, or wondering why you’ve pushed the fader quite high yet the result is quieter than another one set way lower. Having a clear aim (i.e the -12dBFS thing) basically answered one of my biggest headaches for a while - what to aim for across the board roughly before starting to actually mix. Cheers :)
Since, I'm using a mixer that has a dBFS, I'll be looking for that -12dBFS on all my channels. I think that will be a good start to eliminate feedback. Thanks @Audiuniversity 👍🏾
And while the comments are flying, I'd like to state what just came out of my mouth after watching this audio tutorial. Firstly, I appreciate your friendly but down to business projection. You are articulate and intelligent in your field. I agree with your mixing style and took note of your points. What also was appreciated was that you didn't spend the front end, middle and back end of your video trying to sell by asking to "like" and "subscribe" and putting distracting animation with loud clicking noises. This is the second vid in two days in which I ran into you and guess what? I "liked" and "subscribed"! Thank you! :)
Great video! Question: near the end, you say to set the levels of all tracks to around -12db, allowing the Master Bus to still have plenty of headroom. But yet, all the track faders are at unity. SO, if you didn't use the individual track faders to get the tracks to that -12db level, how did you get them to that desired level?? I would have thought that's what the track faders were for... to get the individual tracks to the desired levels? (-12db, in this case)
You should use the faders to mix the tracks to the desired level. However, you can set the PRE-FADER level to about -12 using clip gain, a gain or trim plugin, and the output level on your other plugins.
8:22 he set the levels through the waveform files on the playlist. Question: If this is what people always do for gain staging then howcome there isnt drum kits with all the sounds set/normalized to a certain level (-18db for example that most vu meters are calibrated to) so that you dont have to go through this tedious process everytime? Or is there any drum kits out there that are?
Don't you love when you've been running audio systems for 40 plus years and within 5 minutes a 20 year old kid has already started to teach you something.
Glad to hear that, @The Aleons! Thanks for the kind comment.
@@AudioUniversity yeah I was worried that I might have been peeking my Daw but my cakewalk sonar maxes out at zero on the meters. They turn red about -6 so I actually run a little in the red just like a VU which I like. Someone else was arguing with me about running my d a w m in the red and I'm like no man I can hear clipping especially digital clipping I can hear that s*** from across the block LOL
Haha! Digital clipping is hard to miss! In reality, digital signals in the DAW sound exactly the same regardless of level until they actually reach 0 dBFS and start to clip. That is, unless you’re using plugins specifically designed to for non-linearity. Of course, signal to noise ratio varies at different levels, but there’s A LOT of room for error in the 24-bit digital realm when it comes to noise.
💯
But what have you been doing all this time?
Teaching is a gift! No matter what you do in life, use your gift kid!
One thing that’s hard for us older guy’s to wrap our heads around is the amount of people that are into audio these days.
It used to be a secret society but not anymore.
Plus, Re- teaching the fundamentals is …. Fundamental.
Thanks, UNC! I really appreciate that!
I just completed a 2 year audio production university program. This channel is still ahead of many professional information sources.
1000 years later, people still talking about gain staging. great!
This video finally answered "stupid questions" I've had for years, such as "I want playback to be louder - do I turn up the volume on the audio interface, the computer, the guitar, the DAW faders...?" I think people are too embarrassed to admit that they don't know how to properly set up everything before they even hit record, and you even gave us suggested % levels to work with. Thank you so much!!
I'm glad it was helpful! Let me know if you've got any suggestions for future topics that would help.
I have watched literally every video on youtube regarding tutorials on mixong and mastering etc and I can say with full confidence that this chap is by far the best mixing and mastering teacher that I have ever come across - so happy to have found this channel today - thank you so much for the superb content!
You've not come across Gregory Scott and kush audio
@@UncleBenjs Will have a look at those recommends, thank you!
@@jeremyfox7599 Kush After Hours with Gregory Scott. You'll love those videos, some of the best mixing advice out there
Thank you for these info
As an industry veteran of 45 years, I feel like you glossed over the importance of setting the gain of the pre-amp correctly. The goal of a pre-amp is to take signals that are microvolts (mics and DIs) and boost them up to line level. Every source requires special attention. A vocal mic typically needs more gain than a kick drum mic. Once you've mastered pre-amp gain, everything else downstream operates at line level. For me, the pre-amp is THE most important element of gain staging.
Technically, I have a microphone that comes with line level output built-in, but considering that that's probably a pretty rare thing still, even these days, I'd still have to agree with you... I'm only a volunteer soundboard operator and have no training in the subject though, so my knowledge is almost nothing compared to all you industry veterans out there, and let me just say thank you to every one of you no matter what part of the industry you're in, because even in my volunteer status, I know a bit about how hard an A/V guy can sometimes work and STILL not be 100% sure if it's my fault or the equipment's fault, when something happens. Let's just say, "25 year old soundboard" and "channels going out", and you'll probably know exactly how bad of a time I had for the last couple of years before 2020 or so when our church got enough money to have our audio installation company we use for our technical work, actually finally get and install a new soundboard. Technically, the installer's parts source didn't have the requested soundboard available except for a wait-list or something, so the church had to purchase it from sweetwater themselves instead, to meet the installer's timeframe for fitting us into his free time, and literally everything got upgraded right down to the soldered on 1/4/XLR combo plugs in the walls. IDK if soldering wall plugs is standard, but if it's anything like a standard outlet is installed, usually, then I must say that I didn't know that anyone actually went through the trouble of soldering connections in the wall socket. And those combo plugs, man... Before, we just had standard XLR jacks, but to now have that ability to use it for literally almost ANYTHING that plugs into the wall, well, it's kinda cool honestly, and also, to have not only our original 16 channels, but 32 DIGITAL channels, on our new soundboard, while still being smaller than that monstrosity in my barn I'm trying to decide if it's even worth it to try and run a can of DeOxIt through the thing to try and get it fully, or even partially working again, considering at least half, if my memory serves, or more, of the 16 channels, are either dead or going that way, well, again, that new mixer and all the stage plugins, just, WOW...
For reference, I had been saying that I thought the old soundboard needed replacing, around 2 or 3 years prior to them getting it replaced, so, even if it wasn't a headache at the beginning, well... I said it needed replacing, and if towards the end of it's life/usage cycle(after I mentioned it needed replacing at some point preferably soon) I was ending Sunday morning (on nearly every Sunday) nearly having PTSD from the issues and errors I couldn't quite figure out,(not even joking) was what it took to get them to take up a collection for new audio gear, then I consider my battle scars worth it, and trust me, for me, an audio "un-engineer", it WAS a battle, one that more often than not, I felt like I was losing, and was feeling well, hopeless, to be honest... Again, no matter what all y'all industry professionals and whatnot actually do in this space, thank you all for your work, as it there's one thing in my top 3 that I could probably have used therapy or other "professional" "help" for, the experience I had with the old soundboard was the only one I can talk about, and I don't know how anyone doing this every day can take the pressure...
I agree. This cannot be under stated
@@northwiebesick7136 tell me about your old console, I may want it.
@@weneedtermlimits assuming you mean the "old" one in my barn, we replaced in around 2019-2020, I don't remember the brand right off, but I think last time I looked at it it had "Peavey" brand logos on it, and it's an analog (no screen, just a million pot knobs and sliders) mixer, made in around 1994, and most channels are either dead or dying in one way or another. I'm not a technician by any means, that would have the knowledge to fix it, so you'd be on your own there, but other than that, if you need one for parts, I suppose there's gotta be SOMETHING useful on it still. I know that it's a 16 channel mixer, and I know that it's got tape player "RCA" connectors for aux inputs, plus it's got separate studio/XLR jacks on each channel. I cannot stress enough the fact that it's over 20 years old and I don't have the experience to be able to diagnose whether it's fixable or even if it's worth fixing. Anyone buying or otherwise getting it would be getting it "as-is" with no warranties... Also, I'm located in Minnesota, if that information helps... I'm honestly not sure what to do with it, because I don't have the speakers, cables, etc. required to do appropriate testing with, and even if I did I'm not an audio repairman so I'd be out of my depths doing anything more than just spraying a few cans of DeOxIt all around the board...
Bro man dem pretend too much
Nice and simple instructions, good job. As a retired advanced audio instructor in southern California, it's nice to see this still being taught. DAW's have made recording easier, but the lessons of tape still apply.
Recording to tape was falling out of fashion when I started in the industry. To be honest, tape is a bear to work with but has a warmer sound in the booth compared to DAW tracks.
BUT I was raised on 45’s, cassettes, CRT tvs and with corded phones. I may be a hair jaded.
@@redmcbeard4230 I agree with you Red. I started on Tape and welcomed the digital with open arms. The one thing tape did was really teach engineers how to mix and work with audio. What I have seen over the last ten years is younger DAW users do not get that knowledge using the computer to mix. I knew a Pro Tools instructor who would teach his students you don't need the faders to mix, you can just use plugins. He had no audio experience other than taking the Pro Tools instructor certification course. That's why I appreciate this channel.
Actually, gain staging with digital is totally different than with analog - with digital you're aiming for -18 dbfs, where with tape you go up to 0db or even a bit 'into the red' for a bit of natural saturation.
A light went on in my head when you said "combined noise floor". Very helpful when considering multiple devices/instruments.
I’ve been watching videos on “gain staging” lately, and this is by far the most fundamentally sound explanation. Thanks.
I've worked in radio for 35 years, and much of that time has been producing audio...ads, promos, imaging, etc. It's been a lot of trial and error. Thankfully, I've found this channel to show me what I've been doing wrong AND what I've been doing right! Thanks.
This was useful for me. I started out in the analog world. I’ve been cranking my tracks way too high for a DAW. Thanks.
Absolutely agree.... I bring all channels down to -12 to start and adjust each channel accordingly to give me about -3 to -5 headroom. I leave master bus alone or sometimes just add a limiter to get to 0 but honestly I turn the my speakers of headphones up to hear.... but the truth is in my opinion mixing low is great for preserving your ears and if you hear issues low it will be more pronounced when you turn up the volume.
Can’t compliment you enough. Thank you for sharing your knowledge and presence to help us all with our audio processing. Thanks again
I just watched three of his videos. This guy comes straight to the point. No nonsense. That’s what budding engineers like myself like to watch. Great work bro. Subscribed to your channel
Thanks, Raman! I’m glad to read this.
I work with analog and digital desks for live shows or services. Been doing this nearly 23 years now.
If I am going in to a session without any pre work I’d do exactly what you did for my base levels on inputs and outputs.
All my faders, mains (gas pedal) and bus sends are set to unity.
From there I’ll line check each input by raising the preamp to a -10 on my scale and verify device volumes at 8 or 80% max volume.
The first rough mix (usually my line check but someone’s band rehearsal #FML) will be getting preamp levels to an acceptable gain setting, basic PEQ/EQ of the individual channels then applying dynamics as needed.
DAW recording is not my strong suit but it is just like working any physical desk. Prep before the show and use your ears, scales and RTA to even put the mix.
Better dead than red on your desk/DAW.
Good stuff and solid presentation. I’ll send this to a few younger folks I know that wanna learn the trade.
Thanks for sharing this info, Red.
device volume doesnt affect gain or volume within the daw so im not sure why you need to set device volume at 80%
Great video. I turn off all plugins and set the DI to -18 peaks. Then I turn on the plugins and adjust the faders so nothing is peaking over -12.
I operated audio for 5 years at tv news stations, yeeears ago but never learned the nitty gritty like i should have - i preferred to master the video production switcher. but now that i work in IT for a school district i find myself filling in as the sound setup person for a lot of auditorium rentals. getting this knowledge will help us set up for better rental experiences. love it.
Awesome story, Alan! Thanks for sharing. Glad I can help you along your path.
Everyone learns different subjects and in different ways. I've found I work best teaching myself with a bit of input from an authority on a subject. This channel has made a huge difference for me since I started teaching myself audio engineering, and gain staging is something I only recently started to acknowledge. Fantastic video, thanks a ton for making this.
I'm glad to hear it's helpful to you, Tarran! Thanks for watching.
Excellent video! I've been dabbling in audio engineering for about 20 years now, and everything you said is pretty much what I do.
I want to say thanks! I've always had vocal problems w/ FB & muffled tone, EQ turned down. I've always just set the amps to 50% and brought up the mics til right b4 FB. I just tried what you said and my lord what HUGE difference. All 3 mics are LOUDER than ever, EQ's are unity, not the slightest indication of FB. My EV's went from +6dB to -8 and my RCF's went from 50% to 25%. Freaking incredible headroom. I'm just really amazed
This is great to hear, @SRTPCC! Thanks for sharing! I’m glad you had a breakthrough!
Outside of the studio in the live event venue I think that something that a lot of people overlook is the power supply coming into each of your components. I run a mobile DJ business and use a APC G5BLK rack mounted power conditioner (not saying that this is the one to use, just what I have). I firmly believe that your audio is only as good as the incoming noise from the electrical supply lets it be. While in a studio (yours or elsewhere) the electrical supply has probably been looked at for noise issues in a mobile setting you have no control over what is coming out of the outlet you are plugging into. Whether it's over voltage or under voltage, or the central AC being on the same circuit it's a crap shoot of what you are getting IMO. I know that when I set up I've dealt with the power supply issues to the best of my abilities so as to not introduce unintended noise into the mix. Side note: This is truly a great series of videos that have been pro for every ones benefit. A person should never stop learning, and if you think that you know it all it's probably a good time to throw in the towel.
I completely agree. I have a Meyer system for outdoor live shows in Little League ballparks and for street fairs. The first thing I ask about is POWER. If I can't put each of my subs, and each pair of tops, on their own, separate 20A circuit then I won't bid the job. I also won't bid a job if the venue was built any time before 1974 (the US used aluminum wire for 10 years because the Vietnam war used all the copper for munitions). I won't do shows anymore in San Clemente, CA, for example, because the infrastructure was built in the 1920's thru 1940's and there isn't any 3-phase power in the entire city. The only way we amplify sound is with electricity. My loudspeakers love electricity and when they're well fed, they sound awesome. When they're not well fed, they sound like JBLs.
I love sound design. I love learning about it even more. Thank you for this channel. Never ever stop educating us. Bad sound makes me sick literally. You are the doc l have always needed.🙌🏾🙌🏾
Comment number 421 because I mess up everything. It's good to finally find a video that explains this as professionally as possible.
Thanks Kyle I have understood your points well. I am NOW testing gain staging my guitars, Dynamic Golden Age Mic and EZ Drummer 3 drums to record 1st songs. It has taken Years (2016) to build this to do this. I have Great 1st timer Equipment by Research everything in the audio chain. So excited to start this !
Just remember:
Our hearing judges loudness mainly by average levels - closer to RMS - and not so much peak(transient) levels.
That's why it's good to use a VU plugin that can be calibrated so 0dB VU approximates -18FS.
There are also pure loudness plugins available, or might already come on your DAW, that measure in LUFS(loudness units, full scale).
This was the best video I've seen on this topic. I hadn't given this idea proper attention, that noise can increase in tandem with harmonics when applying a lack of source volume output combined with additional gain and/or excessive gain (amplifiers, exciters, etc). I have noticed that too much wideness on instruments can reduce head room as well.
I found this was a common issue starting out but now I'm finding that it helps keeping instruments down 30% to 50% volume in the beginning stages while adjusting and mastering (compression/EQ/etc) afterwards. Thank you for making these topics so comprehensive!
This is the best description of gain staging I’ve ever seen! You rock!
Good clear summary of gain staging.
I usually aim around -12 give or take for my tracks and uses a limiter on output bus to bring levels to max 0 to avoid any clipping. I know if you record a bass too hot it can be difficult to tame the sound with compression etc.
Why would it be difficult?
I've been doing live sound for almost 34 years. In the old days, I was taught how to calibrate a whole system... From the input and output of a console, to the eq, to processing, to the crossover, and to the amp. A lot of the old systems that I cut my teeth on, you could tell that care had been taken to pair gear with proper metering through the various points of the system.
A fair amount of time would be spent matching gain through the entire chain.
I rarely see people do that anymore.
Clear, plain, and very practical. Well done!
Its actually crazy how much of this I actually knew without being able to properly explain it. I was expecting to take some new tools away from this, but I really only learned how to explain the reasons I do what I do when I'm setting up to record. As for setting up gain, I find that the key is getting the pre set up right to begin with. Post gain adjustments are easily adjustable, where as setting up a good level for recording is only doable once. Especially when you take mic position into account.
This is particularly important if there are multiple voices on one track as well, the closer I can get them on the recording the easier it is for me down the line to adjust. Not a lot of things bother me more than having two separate recordings that are miles apart in overall volume and having to try to make it all work. I generally make sure to do a few test runs, and do a rough EQ on the vocals to get an idea of where I need to be. I also tend to do a lot of spot adjustments to match the overall volume of the track itself within the DAW. Dipping the gain where things get quieter, and raising the levels when it gets louder. Usually it doesn't need much, but it definitely makes for a better overall mix in quieter sections.
I find, personally, that the hardest thing is getting the EQ right. I definitely spend the most time on that.
Helpful run-through. Thanks Kyle.
Thanks, Curtis!
Would gain staging with this approach (peaks around -10 to 12Db, rms around -18) be applicable to mixing for film/tv?
@@PrisonTorture34 Can be, yes. First, to clarify, -10 to -12dB FS (most mixing boards use a dB VU meter instead which often tops out at +16, more or less, depending on the board). And in some cases for film, the target peak level may differ depending on what you're recording and whether you aim to make a usable mix or are just recording isolated channels. But generally, for interviews where you have 1 or 2 people, I'll often aim for peak levels at -18 on quality recorders, maybe a little higher on lower quality recorders where there's more self-noise.
Wow! Thanks for the detailed answer, Curtis!
You are correct lad, I have applied that Strategy since the Analogue days and my mixes are clean with plenty of headroom. If you are near South Carolina, I sure would like to have you over at my studio. Thanks
Taking the time work within those standard limits pays for itself over and over. Thank you for making this, as there are a lot of people that don't understand the concept.
I've discovered, by accident, that proper gain staging makes a superior and flexible product. My mixes can be played on nearly any equipment at high levels and still not clip.
The idea behind this is that the quality of the reproduction is completely dependant and limited to the quality of equipment used to play.
😊
This is the best explanation of gain staging that I've heard so far. Now I'm confident enough to try what I've learned. Thanks for the lesson!
Dude, that is absolutely one of my favorite Joe Henderson tracks and no one really knows it. Kudos.
I love Joe Henderson too. I went back and listened to the entire video again and did not hear any Henderson used. But then, I saw it on his cell phone.. LOL.. Thanks..
Great, clearly explained video. Such an important topic as it directly relates to sonic performance. I'll share in the general conversation of system gain setup for PA systems... I don't know how many times I have seem amplifier inputs wide open for example, when often the correct setting is some fraction of that, depending on the amps internal gain specs. A good quick tool that doesn't require test equipment that I use that gets you in the ballpark for aligning system gain, is to unplug all speakers from all your amps first. Very important! Next, put on some decently recorded music that has good peak material such as drum kicks etc. Then, starting at your mixer's output, bring the signal up until you begin to get a decent clip flash on the mixer's meter then stop, and leave it there and don't move it... as this will be your reference. Then, adjust the gain on next downstream device's meter or clip light in the signal path, to replicate the same clipping flash as the mixer. (Devices that have a unity mark should remain at unity in most cases) Then, once that alignment is done, bring the main out level on the mixer back all the way down >> BEFORE
Really liking this guy ,its like no nonsense, easy to understand, he has been teaching this old guy a few things
Usually I'm starting at 12 dB peak per channel on the mixing session as you said. If I mix in my arrangement session I'm turning down synths outputs to meet this requirement or I'm using gain knob in pre-mix section of the mixer in my DAW (I use Cubase). Also I adjust clip gains if it's necessary. That simple and it gives plenty of headroom on the master bus, making comfortable mixing environment and also it's giving nice headroom in mixdown for further mastering session.
I use Presonus Studio One, and we have a gain trim knob on every channel. I used to use a VU meter on the master bus and set it to -18. Then I would play each channel in solo, watch the VU and adjust. Later I started using a VU on every channel, but now I'm just looking at the FS meter, peaking at -12 dB.
I also put a trim plugin in-between pluggins and/or last in the chain, because if I'm using any plugin in a more extreme way that changes the level, and the plugin(s) don't have a gain or level knob, I use the trim plugin to put the level back to -12 true peak. It's not a rule, but it helps keep everything in line, which means that I don't have to set all the faders extremely different to get approximately the same levels, e.g. one channel around zero, and another at -10.
Very good, concise video. Thanks. I usually just record in directly synths, drum machines and guitars. I used to shoot for -12, but I’ve noticed i get better results with -8 and then using gain reduction in my daw I will take it down to -12. I’ve revelry started mixing with the -6, -12, -18 rule recently as well. Again, thank you.
Glad you enjoyed it, John. Thanks for watching and leaving a comment.
Great video! Something I have been preaching for 30 plus years. Whether it is analog or digital, the concept is the same and 100% true! Good Job!
In the digital domain I use to set my tracks at Peak -3 dBfs (percussive Sounds) or -6 dBfs (sustaining Sounds) and then control, if they're round about between -18 to -12 dBRMS... adjusting if necessary. Sometimes I also use a VU Meter calibrated to -18 dB in the Main bus.
I am an old audio engineer, I used to use minus 3 to 6 on analogue consoles and guess what, I try to record stuff digitally between minus 3 and 6 db
Really amazing video. I have been watching a lot of videos on this stuff lately and this was one of the most in depth and concise ones I have found. Plus the way you explained things just cleared so much confusion up for me, I feel like I have a much stronger understanding of things now. Fantastic work, thank you.
I’m very glad to hear this, Ghost. Thanks for watching!
bro this is one of the most informative videos on audio engineering ive ever watched great job of simplifying the information my brother
Mixing in logic I have everything (busses, tracks, etc) go to one mix bus, which I turn up or down as needed to avoid clipping the master buss (or I turn it up above 0 if I wanna hit the master buss compressor harder). I still think about getting individual tracks to a proper level to hit whatever processing I put on those tracks how I want it, but in terms of clipping the entire mix, this seems much quicker and easier than making sure everything is peaking at -12. It also allows me to absolutely smash some signals if I want without worrying about it.
I never start from unity. Eveything is at negative infinity and I gradually take up the individual track faders as I go from track to track
@@technober Right!
Very good explanation! It should be the basic for all DAW users but it doesn't. It is so simple to gain stage after every (stage) plugin and it makes so many things more easy for the entire track.
I use a lot of external effects, mic pre-amps and other sundry devices for compression and signal coloring and work with an aggregate of interfaces connected to my DAW so it’s really important for me to pay attention to gain staging. Even before a session begins I like to use a 1k tone generator and pre-set the optimal gain stage level of all connected devices. I start with connecting and sending tone to the last device in the chain like a DAW interface or an input on a console feeding an amplified live room where the DAW or console faders have all been set to unity and setting optimal gain there until the tone reaches the optimal level I am aiming for and then repeating the procedure moving backward, connecting the next device in the signal chain and optimizing the input level of that device and then adjusting the output level so the tone level of the last device in the signal chain sits at the same level where I had set optimal level. I continue moving backward throughout the signal chain repeating the process until I reach the source like a mic, guitar DI, synth, etc. being careful to match impedances of the output level of each device moving up the chain and matching the impedances of output to input and switching the impedance on the output of the tone generator accordingly to match what those impedances should be and never mix-matching impedances which can cause much unwanted noise and sometimes catastrophic damage.
Not saying these are the exact levels where I will be working in a session, but close enough to where a simple fader move (not gain level adjustment) is all that is needed to adjust the mix in the DAW session or on the console feeding the amplified live session or tape.
Thanks for sharing Dave!
Precise and easy to understand information.
I was always a little nervous moving into the digital mixing domain, so many choices and intelligent decisions to make
May I suggest that you delve into external analog signal processors getting chained up prior to a DAW? Too many get obsessive about using older analog gear when you can now get similar results with what is available in your DAW. That old analog gear and the physical connections typically add noise.
Great idea, David! Thanks.
I try to have around -4db to -5db at the stereo out channel at max for my mix, which gives me enough headroom to master my song.
-12 db will be a good starting point where I can bring a track down to that level and do the gain staging of other tracks around that.
Really good content. Thanks mate 😊
I'm really loving the ability to put plugin UI's in tracks in reaper, and then putting the ZenoMOD VU meter in each one. This lets you adjust the gain in the track like you did with the gain plugin. It's awesome!
In Ableton I put a Utility (gain plugin) with -12 dB on any channel right after the sound source, to keep audio at suitable levels for the following plugin chain. I could make sure that my synths don't output louder volumes in the first place, but when stepping through presets, those levels get reset most of the time. By checking the inter-plugin peak meters, I also make sure that my plugins don't add a significant amount to the overall peak level of the channel. Adding up all the channels usually gives me a master peak level of around -6 dBFS, which leaves enough room for master bus compression or loudness optimization.
As a bonus, I also put another Utility with +12 as the very last plugin on every channel and set the mixing fader of that channel at -12 for compensation. This gives me a little more headroom on the fader, as the +6 on Abletons faders does not leave much room to boost quiet signals.
This is not good ‘normal’ practice. You are reducing the insert level of your channel by 12db, whether it needs it or not, and you can be sure this introduces unnecessary noise when every single channel needs makeup gain. Much better to optimize your source levels whenever possible to the nominal level your DAW expects at its input. I’m guessing you’re not using preamps or any mixer. Lacking that, adjust the outputs of your sources as best you can so that your faders have the range you need. If you find it routinely necessary, make a habit of parking channel faders at -6 for setup as your new ‘zero’. Those two utilities you insert are acting as noise ‘anti-reduction’, the last thing you need.
@@artysanmobile This is true for analog audio processing, but I work completely in-the-box, so my sound sources are synth plugins or samples. I assume you are talking about quantization noise then?
As DAWs (and probably most plugins) these days work with floating-point values (64bit mostly), quantization noise is negligible. As SNR for floating-point values are 192 dB (32bit) and 385 dB (64bit), just changing the volumes by a few dB has absolutely no audible impact on sound quality, regardless of where in the signal chain the volume is changed.
The only source of quantization noise is when the sum is rendered to a 24bit integer wave file.
@@markuskopter I was just giving good general advice, in the box or out doesn’t matter, and I’m not referring to quantization noise but just noncoherent noise as we have in any electronic signal.
By the way, you’re quoting a bunch of numbers that mean virtually nothing. Hundreds of dbs and 64 bit word lengths are just nonsense concerning audio. You do know that, right? I only mean to be constructive saying this.
@@artysanmobile I have to admit, I was wrong in stating those signal-to-noise-ratios were for floating-point values; they are for integers, actually. The SNR in floating-point processing is even much higher.
I also was wrong in saying the final render were the only source of quantization noise. In fact, every calculation in a DAW has to be rounded to some resulting value, which results in quantization error, or "noise". In fixed-point systems the size of this error (the volume of the noise, if you so will) is determined by word length. Did you ever use a bit crusher plugin? It reduces the word length of the audio signal to deliberately create quantization error, which we then hear as noise.
But other than that quantization noise, there is no inherent non coherent noise in the digital domain.
@@markuskopter I hope you’re just a little deep in the sauce on a thanksgiving day, enjoying life. You sure have a great story to tell. Hey, did you know that Africa is splitting into two pieces as we speak. Because that’s far more relevant to this discussion than the gibberish you just laid on me. I really do hope you’re ok, Markus.
thanks for the upload
i was already doing most of this
but you know reassurance is a helluva drug😅
Been using this method for years, but I've always had each track, send, master, etc. at -6dB as a starting point.
Master level is never changed and no effects/sends ever go on the master either.
Very well explained basic concepts. It’s sometimes helpful to go back to basics. And you obviously understand very well what you are teaching, unlike other channels. Bravo!
There's another aspect of gain staging that isn't covered in this video that is of paramount importance to both live audio reinforcement and recording studio practice, and that is setting the maximum SPL output levels of the PA system or nearfield monitors. Bob Katz recommends his "K-System" for recording and that's a very good place to start: -20 dBFS pink noise should equate with 85 dBA SPL at the listening position, which makes 0 dBFS equal to 105 dBA SPL. When mixing live audio in small clubs, I shoot for peak SPL of no louder than 96 dBA SPL. The rule of thumb here is that if the peaks are making you wince, it's too loud. But it should be recognized that even 96 dBA SPL is technically too loud for extended exposure.
Gain staging changed the way I think about mixing. Big tip.
I honestly think this is one of the best videos I've ever seen. Thank you.
Wow, thank you!
thanks, great info as usual my friend! Growing up in the 70's(yep, i'm old) I can tell you that the tendancy was always to push the limits of a hot level, and finding the sweet spot with your recording device for best results. Now, in the digital era, thats no longer necessary, and in fact its dumb to even feel the need to do that. My point is; this was the hardest habit to break for me initially, since the conditioning of 40 yars of that habit was so deeply ingrained...todays recording process has become a piece of cake to get a good quality recording on the first shot...not saying it's necessarily a better recording than analog, since both have their appeal, but it's definitely easier to get a better recording with digital...
thanks again!
Well said! Thanks for sharing this perspective!
This was great!! Learned a lot . Probably one of the best and most understanding tutorial I’ve come across
There is another inherent issue with having the master sliders too high or too low: the individual input sliders will not have much leeway to accentuate or attenuate the sound. It’s better to be able to control the individual input faders so they can move for a longer range and therefore able to make small and more accurate adjustments. I have seen many sound mixers keep the master faders very high and not being able to make much adjustments with the individual faders since they were placed too low or too high and thus had very little room to move up or down and that made for more difficult mixing.
Your videos are really good. I don't always agree with everything you say but you typically give an alternative explanation. Some RUclips channels have thousands of subs and I can't really understand why. Your channel is doing very well and I CAN understand why. You provide great content.
With that said, That Ashley EQ is awful. I know I sound like a gear snob but I'm simply keeping it real.
Also- @8:40 There is not plenty of headroom. Your peak is around -0.5 One needs to account for FX being used on each track that will sum to the Masterbus. To be honest, in digital, even a peak of -0.5 is a little much.
I mean none of the gear he’s using is nice, a focusrite and a powered speaker aren’t nice either it’s just demo gear for the video. Those Ashlys get the job done I never had particular problems with them, it’s just I rarely use a rackmount graphic anymore
@@IWannaGoMissing For live applications I've used Ashley as well, but not cause I owned it, it's what was there and you're right, it got the job done. Last thing I want to do is sound like a gear snob, but the Ashley's had noise problems. May be more tolerable now in the digital age, being able to cut some of it out, extended noise floor.
When I say cut some of it out, I'm talking about using something like Izotope RX which is great for restoration and handling noise issues.
Just came across this great info. Personally as I do mixing in a different daw than my production I prefer to setup my audio lanes to have a pretty low db FS at the start and slowly through busses and a pre master that everything is affected by get it back to the ideal threshold before mastering. But I've seen this method work to and I'm glad to see that I wasn't lying when I explained gain staging to my friends (though will be sending this to them too)
You deserve a like, a comment and a subscribe. Not even a second of your video had unwanted information.. great video Kyle, thanks.
Thanks, Srivatsan B!
You are an incredibly effective teacher....I am your newest fan. Keep teaching and thank you for your clarity.
I've gotten slack about this with total denials but driving channel input gain close to or into clipping just to achieve a fatter sound on a snare for instance, slowly damages the input transistor stage. When a channel is driven hard the transistors are turned on for a longer duration and conducting current longer. This will cause a transistor to over heat and eventually fail. The same holds true for speakers and HF drivers. Having serviced a lot of consoles I could tell which channels were used for kick and snare as they were first to fail. If you want that overdriven sound use an appropriate plugin or get creative and send your drum mix into an amp, guitar amp possibly and mic it in a live room. Maybe Kyle has done a video on the benefits of even harmonics versus odd harmonics.
Thanks for bringing that to everyone's attention, Donald. I expect many will still overdrive their gear if it produces a pleasing sound, but maybe they will appreciate the experience more now.
As I watch this video, I am listening through JBL noise canceling headphones while laying on the floor and staring at the ceiling. But the headroom is more than satisfactory.
Totally agree, I've been doing it the same way for years and it always works! Thanks for the video and perfect explanation. ✌
For all gain related work, I cannot recommend enough Klangfreund LUFS meter. It saves you a lifetime of eyeballing and guess work with levels and mouse clicks. It has multiple grouping options so you can level match everything according to Loudness in one set. You can gain stage/fade up mix really kick, compare dry/wet fx between tracks, and so many other things. It's very light on the CPU. Mike Senior brought me into this plugin.
Trying to set gain on multiple devices in a PA System can be a challenge!
I have a Tri-Amped System. This consists of a Peavey IPR2 2000 for the HF Drivers in my Peavey SP4V2’s and an IPR2 5000 to drive the dual 15’s in each cab and an IPR2 7500 that takes care of the Peavey SP218’s.
Before the amps we start with my iPad, which goes to a Yamaha MG12XU Mixer, then to a Peavey 231 Dual Channel 31 Band Equalizer that I use for the Low Cut Filter to protect my subs, then to a Peavey 35XO Electronic Crossover that feeds my amps.
All devices have Gain for each channel and I’ve never been able to set each one to what I think should be the best level because I’m not sure what the level is. I’ve always left the EQ at 0dB, as well as the X-over and left my amp gains maxed and make my adjustments on the mixer.
How can I get this system tuned in the proper way so I can achieve great sound and not have to worry about clipping?
I recommend keeping the meters along the signal path at a "nominal" level. For an analog meter, I will aim for peaks at +3 dB. For a digital meter, I will aim for peaks at -6 dBFS.
@@AudioUniversity Thanks for your reply. I’ll give it a try and see what happens.
Too many Rock concerts in the 80’s is to blame for me wanting the music loud and clean. Like muscle cars, no matter how much power you have, eventually it just isn’t enough.
very well summarized. this should be a compulsory watch for all bedroom producers.
i've been using the K-14 system for a little while now & have gotten pretty good results. i like to adjust the levels going into the mixer channels so the signals are peaking usually no higher than -10dBFS, maybe up to -8dBFS for drums or percussion.
then, once i've gotten a decent rough mix going, i will apply EQ & compression to my drums & bass, getting a good balance between namely the kick drum & the bassline.
i follow that up with soloing the kick & taking note of its level on a VU meter. next, i'll bring the bass back in & make sure that each time the kick & bass hit together, the VU level increases anywhere from 2-3dB. once i have them balanced well, i send them to a kick & bass bus for glue compression.
finally, i'll mix the other elements around the kick & bass, while keeping the kick & bass levels together throughout (ie; if one changes, they both change, etc.)
using the K-14 system, i'm able to get a LUFS reading of around 10-12 integrated after pushing my limiter on the master up by an available 6-8+ dB, while still maintaining a -1dBTP.
Honestly these videos are more educational than when I went to SAE in Athens. That place is full of people who tell you half baked knowledge and don't even cover the full picture or go in depth as to how things are done.
Don't always set your Master to Unity 0db. If you're working with a massive live sound system which isn't properly attenuated at the amps you're gonna be turning Channel faders up only slightly before it gets really loud. Low faders means 2mm moves give you a 5db difference, so you're better off with a Low Master, High Channel faders for easier fine tuning. This is situational and only an example of venues I've worked in.
If you can’t adjust the amps to the proper output level for the venue, I agree. Use the master fader as a volume control. However, it’s probably best to turn the amps down when possible.
@@AudioUniversity dead on! That’s why I got a nice line attenuator for our powered speakers since they didn’t have an on-board output control.
You can have any gain on your daw tracks, even you can have visible enormous clipping of the signal and it wouldnt have any involving to the result signal if you take care about master bus volume. It's almost impossible to overload internal daw bus cause all of the modern daw have 64bit mixing algorithm. The only place when you can get clipping of your mixes it's the master section and hardware output of your sound card and etc.
However there are many plugins which require certain input level due to their processing algorithms(overdrive, distortion and other saturators) so you have to follow the level to get proper result. But it is quite all right if you get some clipping on any internal bus of your daw except master bus.
This was the best explanation of gain staging and its application I’ve ever watched….and I’ve watched a lot trying to figure it out. Thank you. I’m still confused as to why some of my mixes are louder (kick thumps more or snare slaps better or bass is deeper and more upfront) on some tracks more than others even though I perform gain staging in the relatively same way each time? Then when I remix the quieter tracks to match the louder tracks output, I clip! It’s so confusing! And I haven’t even tried to mix on my Toft atb32 yet!
First: Kyle...your videos are absolutely VERY good in all aspects!! Thank you for this hugely helpful and detailed information.
I'd like to ask three questions here:
1: Am I right when I conclude that - given the natural limitations of the human hearing spectrum - I will never escape the fact that those bass heavy instruments NEED a significant dB advantage (higher peaks on the dB meter) to ‘stand out’?...even after some (deliberate or accidential ) saturation?
2: If so: can I solve this problem by simply lowering the gain on all the other tracks? (and get the volume back in the mastering stage?)
3: Does it make a technical difference if I choose to lower each and every single track individually or save myself some time by lowering the gain of the whole groups that they belong to?
Really useful, thanks. You’re absolutely right, there isn’t enough info re GS online
Best explanation I have ever heard. Now I know what gain is. Thank you.
Glad to hear that! Thanks for watching!
Are you familiar with the "shoot to the right" approach in digital photography? If not: it's basically setting the exposure to shift the resulting photo's histogram as much to the right (brighter tones) as possible without clipping. (Or at least not clipping any part of the photo you care about.) That increases the signal to noise ratio in the recorded image, since the underlying noise (thermal and otherwise) of the sensor is constant. Then you pull the exposure back in post which reduces the noise along with the signal.
It also makes for smoother gradients, e.g. if you take a shot of a dark scene where the recorded pixel light levels range from 10 to 100 you'll have 91 distinct levels of brightness recorded, whereas if you increase the exposure so that the range is from 20 to 200 you'll have 181 distinct levels. I imagine similar effects could be seen in a digitally recorded sound wave.
Anyway, it seems like you're kinda doing similar things here with digital audio recording, and I thought I would share that.
It does make me wonder if anyone has made an effort to come up with an algorithm that can take a number of tracks and their recording & mixing parameters, and output a suggested optimized mix for preservation of maximum signal precision, or even suggest better recording parameters for the next take. It seems like it should be mathematically possible, though I don't know off hand how much number-crunching it would take.
Man! thanks so much for your pro attitude and kindness, what a terrific combination! 🤘
For studio, When I'm tracking and mixing, I try to keep my mixes at around -20 lufs which allows me the headroom to reach around -12 lufs in mastering. Great video. I was curious about what you'd say for the live application. I had been keeping my foh speakers at a certain level and controlling overall volume with my master fader.
In integer PCM (traditional digital) you have an additional problem with audio recorded extremely low. Not only when will noise be a problem when gaining up, but the signal will have no resolution, so you may up get digital aliasing introducing weird sound.
With a good fp32 (float) PCM workflow, everything is much more forgiving and going above 0db is no issue. Maybe it is a bit of an abuse but my mixes typically always go above 0db and then is rained in by a limiter on the master bus. For spoken word / podcast that sounds more than well enough to my ear.
For video work I’m hoping we’ll see everything transition to floating PCM within the next few years so audio work becomes focused on capturing the best audio (best signal / most dynamic range) instead of throwing reserving headroom. Let the ADC capture great audio without managing gain staging much :) A lot less stressful not having to worry about clipping or recording too low. When you don’t have an audio guy on set and need to manage video, audio, capturing PC output, supporting talent and what not, traditional error prone integer PCM gain staging is an additional stress factor I’d prefer not to deal with. A great sounding FP32 ADC built into every camera XLR input would be great!
Learning a lot of things I thought I knew from watching these videos.
For live sound I do pretty much what you detailed. Master @ unity, channel @ unity, tone plug in channel and adjust gain to show levels @ unity.
Cool video! I especially like the specific meter range example (-12 & -18) for Max and Min levels. 👍
that range is unrealistic as fuck, dude has clearly never messed with rap vocals
Padre Kyle preaching about Gain Staging..!! 😍😍 A perfect Tutorial in its kind..!! thanks Kyle... and Waiting for more..👍👍🤟
My advice: Most people listen to music off their devices like phones computers etc, bounce your project and play it off a normal device to compare before considering your projects done there are many times I've done this and notice massive volume differences, but bounce it as an MP3 as the noise increases a lot more which gives an accurate perspective or just get some grade A monitors bounce is on your computer as an mp3 and play it back, be aware that any gain you change when recording is destructive though and no amount of compression will change it as the track has already clipped.
I never consider my tracks done until it has been run through and sounds decent on a portable BT speaker, a phone, basic ear buds, and good through PC speakers and car stereo as most music is listened to in the car for your average user. I have learned the lesson of hearing a mix out of my monitors and being proud, then listening through literally anything else and being embarrassed and saying, "Wow, that sounds like absolute, grade A dogshit."
Thank you for the information gain staging. I will keep this to add notes in my audio notebook.
It’s odd how “simple” it is conceptually yet as soon as I’m listening to other music that has a lot more volume it always ends up with trying to match it. And bye bye goes your headroom.
Or working with things tracked on different days etc where you can just end up in a crazy battle trying to level things out or automating like mad, or wondering why you’ve pushed the fader quite high yet the result is quieter than another one set way lower.
Having a clear aim (i.e the -12dBFS thing) basically answered one of my biggest headaches for a while - what to aim for across the board roughly before starting to actually mix.
Cheers :)
Thank you. Like all your excellent lessons, this is so clear, concise and helpful. Your presentations are masterful.
Kyle. Your knowledge and instruction is so valuable! Awesome!
Since, I'm using a mixer that has a dBFS, I'll be looking for that -12dBFS on all my channels. I think that will be a good start to eliminate feedback. Thanks @Audiuniversity 👍🏾
That’s a good start! Check out this video for more tips: ruclips.net/video/z2ceO8D_MUw/видео.html
@@AudioUniversity I heard the terms Larsen and feedback are two different things. Do you have a video that explains that?
And while the comments are flying, I'd like to state what just came out of my mouth after watching this audio tutorial.
Firstly, I appreciate your friendly but down to business projection. You are articulate and intelligent in your field. I agree with your mixing style and took note of your points.
What also was appreciated was that you didn't spend the front end, middle and back end of your video trying to sell by asking to "like" and "subscribe" and putting distracting animation with loud clicking noises. This is the second vid in two days in which I ran into you and guess what? I "liked" and "subscribed"! Thank you! :)
Hi Kyle, you have been a great coach. Personally thank you for the short videos with much information.
Thanks, Nirmal! I’m glad to help.
Great video! Question: near the end, you say to set the levels of all tracks to around -12db, allowing the Master Bus to still have plenty of headroom. But yet, all the track faders are at unity. SO, if you didn't use the individual track faders to get the tracks to that -12db level, how did you get them to that desired level?? I would have thought that's what the track faders were for... to get the individual tracks to the desired levels? (-12db, in this case)
You should use the faders to mix the tracks to the desired level. However, you can set the PRE-FADER level to about -12 using clip gain, a gain or trim plugin, and the output level on your other plugins.
8:22 he set the levels through the waveform files on the playlist.
Question: If this is what people always do for gain staging then howcome there isnt drum kits with all the sounds set/normalized to a certain level (-18db for example that most vu meters are calibrated to) so that you dont have to go through this tedious process everytime? Or is there any drum kits out there that are?
very very valuable information. Thank you for these lessons Kyle. They really do help.
I’m very glad to hear that! Thanks for watching.