I read that the enemy in the digital domain is not noise floor, but the farther you go down in level from reference, distortion goes thru the roof. So much so that a sine wave can actually be distorted into a square wave. What are your thoughts?
@rillloudmother Sorry, I don't remember the source. Why do you ask? It was discussing why dither and limiting is applied to commercial mixes to push the signal closer to reference to avoid distortion.
@sonicfuker that's not why dither and limiting are used on commercial mixes. dithering prevents bit truncation. limiting is used for many purposes, but in professional mixing and mastering it is not used for that purpose. limiting is often used when tracking to prevent digital overs which result in distortion if the signal exceeds 0dbfs.
Even the self-styled Master of His Tools has to be open to the corrective input from those who know more than them about other than the Software they have finally chosen to champion.
It's called OPEN-MINDEDNESS.
What a big mistake it ever is to think you yourself KNOW IT ALL about everything.....Like me! LOL : )
This is a great introduction for "Gain Staging". a starting point of -10 dBFS is nice ( and the master gain @ Unity ). but without a good explanation of dBFS Peak and dBFS Rms and without a very good vu meter... it's not possibile to make a correct gain staging.. not to mention the plugins and their gain once they are inserted on a channel to eq compress o limiting a program.
A more deep tutorial and explanation on this subject are expected :).
Hey brother, keep up the good work! I have been watching your vids and their are a lot of haters or side line producers, a trying to put down you style. Keep up the vids, I am, thankful for your vids and info. I dont use pro Tools, but good knowledge and info is interchangeable, and IT WORKS! Cheers
@IScH45I lol eactly, that's what it's for. If the level is not right hitting the master fader, fix your mix, don't pull this trick with a needless bus..I think we agree.
the part about keeping the mix under -10db is just unfounded, and a myth. As long as you are not clipping, there is absolutely no difference in sound between -0.1 and -20, except the noise floor (which does exist by the way, even in the digital era).
I thought as long as you are not exciding the 0db level that no clipping is occuring and everything is okay? Also What if i get my mix all set in the DAW and ready to sum, I instead send each track back out of the DAW and into an analog mixer to be sumed via the mixer, then use the stereo outputs of the mixer to creat a stereo track of all my tacks that have been summed in the analog domain? Should i then set the gain levels on the mixer to unity gain? Great Vidoes by the way!!
This is irrelivant if you use anything besides Pro Tools, because every other platform has advanced to floating point back in the 1990's, so no clipping occurs unless it's on the input or output. No clipping can happen in the mixer or plugins... not true of Slow Tools.
@freedom0speech Pro Tools is floating point, (not HD) always has been. Clipping is a real issue in all digital systems because it doesn't sound good. Proper gain staging is critical in the analog domain as well, so no this isn't irrelevant outside of Pro Tools.
Gain staging is important for doing A/B comparisons, but what plugins are you using that can clip? Every VST plugin I've ever used is floating point for about 15 years now. You can only clip on the converters (in or out).
But yes, it's also important to understand this for outboard gear (both digital and analog).
@recordingrevolution If your peak levels dont exeed -3 you are probably ok. But Normalize wont bring the levels down. Normalize brings level up till the peaks hit all the way up to 0Dbu.
@TheProgmagog You might mean "Gain" command;) But assuming you had "damage" it wont be fixed by bringing levels down after the fact. It will only lower the volume but the clips would still be there, all beit quieter overall, still clipped. But at -3 peak level you probably dont have clipping.
@recordingrevolution I've used Logic Pro for years now, I don't remember exactly what happens in ProTools but I don't think I have ever heard clipping in Logic Pro unless its on input or the stereo bus. Is ProTools the same way?
@freedom0speech This is a misunderstanding of where the clipping is occurring. at 24 bits, plenty of heardoom equals higher fidelity in any DAW. BTW PTHD7 uses a 48 bit mix bus archetecture. That is wider than even 32 bit floating point.
24bit and 48bit only allows for more dynamic range UNDER 0dB. It doesn't add headroom above. So no, that will not fix the clipping problem.
Modern DAW software uses a combo of 32bit and 64bit floating point. All my compressors and EQ's are 64bit floating point, and the rest are 32bit floating point. No clipping anywhere, unless I clip the A/D or D/A converters. It's been this way outside of SloTools since the year 2000 or so.
@freedom0speech Headroom above 0db? 64 bit since 2000? I doubt that, but rock on anyway. I think PTLE was a big mistake because now the HD rig gets lumped in with the rest. The TDM system is leagues beyond any native DAW. But regardless of that, its not the bycycle, its the rider. If gain staging in my PTHD rig is more critical than others, so be it. There is a reason the old school veterans have mostly moved to PTHD and ICONS. I dont think they are scratching their heads about it. Neither am I
Having said that, I think it is wreckless practice to assume gain staging is not relevant at any bit rate. What DAW are you using BTW? I am only deeply familiar with Logic Pro and Pro Tools HD.I have a passing rapport with SONAR, Cubase and Reaper. If I liked them more I certainly wouldn't have shelled out so much for an HD3 Accel. All this mathematics aside, the workflow is awesome and the ability to use my analog 1176s and Pultec and so on in real time easy as a plug-in is a whole other world.
It's not about allowing yourself to be reckless. Its about having safe dB's above 0dB to avoid clipping, because **** happens. Hell, listen to all the music coming out these days done on PT HD, these engineers are clipping at what seems like every stage. It's reckless now to not add that safety net, and I'm sick of listening to the clipping.
@Studio201recording True enough my friend. Which brings us back to the purpose of this tutorial. S#*t happening is totally avoidable and here is the way to avoid it. Having a safety net is cool, but the danger is leaving no headroom for them to work with in mastering. It is always good practice to leave I would say 6db, of headroom on your mix bus and 3-6 on your recorded tracks (to preserve transients) Leaving about 6db on the mix bus is something your mastering engineer will praise you for.
You can doubt it all you want. It's a fact. The UAD-1 processed at 64bit using a 32bit floating point engine (on all native DAW's at the time) around 2000.
As for why people use PT HD, it's simply because they copy each other like you just explained. Famous people use it, therefore it must be great... we don't know why, but it just must, they must know something. LOL! Please.
@freedom0speech I have 2 UAD 1 cards. Great sound but its not the same as a truly integrated TDM system. People like Kevin Killen, Ed Cherney and Andy Johns arent just blindly following, while guys like you got the real info. You dont need Pro Tools to do great work, but we that use it (HD) know what we are doing and why. Your numbers game and reasoning mean nothing to a guy that is confident in his work. We dont need to emphasize our gear but rather the descisions we make with it.
Listen, I agree that it's all about the talent, not the gear... however, I also can't stand bullshit. I'm sorry but the TDM system isn't even powerful enough to handle most of the UAD plugins in all their glory.
You can make great music on PT HD, but I don't understand why anyone would want the headache. But, that's up to you. However, if someone tells me 48bit fixed is better than 32bit float with 64bit float abilities, I'm going to call them on it.
Also, appeals to popularity and appeals to authority mean nothing to me. I've worked with lots of famous people, and the most important thing I learned is that they are human beings like everyone else, and they do stupid shit (like copy everyone else for illogical reasons) like the rest of us. I don't care what they use, I care what sounds good and what works good.
@freedom0speech You are very right on your point. I am not pointing out merely "famous" people but icons in the field who hesitated to go digital for a long time, and finally turned in their SSLs for HD rigs and ICONs. A big part of the turn is not only PT but Waves as well. The two in conjunction along with a good controller like even the C24 TAKES AWAY all the headache I associate with DAW recording. But I have already taken too much time on technology. Its all about the music.
BTW, I get WAAAAY more out of my TDM accel cards than my UAD cards. In fact my UADs and my TC Powercore are now packed in bubble wrap. What I like with TDM is the real time usage. Its integrated with the entire DAW so you can monitor effects while tracking if you like. Also, whatever you can load, you can always use flawlessly. The UAD cards load more than they can handle and if pushed too hard they crap out.
The reason I feel so many top industry guys have moved to PTHD are many. First with its true real time capabilities both with plug-ins and outboard hardware, along with the outstanding integration with major controllers like the ICON or even the C24 has made the work experience extremely similiar to the old hardware-analog days. That and its rock-solid stability is something they require in such high dollar-high pressure situations.
I've been running my entire Nuendo DAW at under 1ms latency for the last 8 years, and it integrates into outboard gear flawlessly. These advantages are 90's ProTools advantages. It's 2012 now.
@freedom0speech Nuendo rocks! But let me ask, cause I dont know. For example, I have my ISA430, Pultec, 1176s, Requisite PAL III and others patched into my interface ins and outs. Of course I can record through them as I did in Logic. But what I couldnt do in Logic is open said device right in the insert of an Audio Track and mix it on the fly. Logic has Aux in that could be used, but not simply and directly in the audio track path. Can you do this? DO you do this?
Correct me if this isn't what you are talking about, but Steinberg added an outboard FX insert plugin in about Nuendo 4.0 or so (we're on 5.x now), so you just drop that in your inserts--wherever you want your outboard gear patched in--and assign the inputs and outputs to your audio device.
@freedom0speech Sounds about right.The way I have it (and I assume you can do the same) is, Like I said, all the gear is permanently patched. I can access it on a channel input for recording, or I can open it on a channel insert (or bus insert) just like a plug-in. My IOs are all labeled by the device name so its real quick and easy. I can open an 1176 bomb factory or the real deal just as easily. But in order to have 1ms latentcy, dont you have to set your buffer low? I ask because...
... The UAD and Powercore require substantially high buffer settings. BTW did you know that Chris Squire can hear 1ms of delay? That is something Trevor Horn pointed out to NED when they were designing the Synclavier.
The UAD plugins do add additional latency, yes. I hit the "constrain delay compensation" button when tracking, which disabled plugins with latency above 3ms (or whatever I set it to), and then disable that when getting back to mixing.
@freedom0speech In summary, I think your choice of tools is absolutely viable and any difference between your sound and anyone elses will come down to how you use what you have. As for me, I am a musician and singer first. I am fortunate enough that my music has allowed me to afford the rig I really wanted and I am really happy as well. As many reasons I can say that I love Pro Tools HD, I can counter with reasons I hate Avids business practicees. Therefore I dont upgrade. I just use what I have
Hey, I'm glad you gain-stage using PT. Too many don't and I have to listen to the clipping as a result. I think this could be avoided, but I guess there would be something else lazy engineers would do wrong... lol.
As for hearing 1ms latency, I believe it. I've heard some people say that no human can hear below 20ms, which is nonsense. It all depends on what we are listening to. If you can hear it phase against the original signal, we can probably hear 0.1ms in some cases.
@freedom0speech That is exactly what NED was trying to say to Trevor Horn. If that were indeed the case, we wouldnt be using about 20ms pre-delay into our reverbs:)
Off-topic, but I've been using 40-50ms pre-delay on my reverb's for my last mixing project, with decays well under 1 second. It seems to create an awesome 3D sound somewhere between a delay and a reverb. I used to mix delays and reverb's, but it sounded mushy, this sounds clean. Anyhow, that's a neat little trick I thought I'd mention, however very off topic.
Keep in mind that you will get similar latency on PT HD when you patch in and out of the system. Though you may not notice it until you patch several times in a row.
Latency issues are getting better with time, but they are far from perfect on any system. They will likely never be perfect, but they should get so low that they become irrelevant soon.
I should mention that patching in outboard gear like this in Nuendo will not add latency to your mix, because their is delay compensation applied.
I don't think Pro Tools has this yet, unless I'm wrong, but I think the latency (however small) added by the TDM chips and the A/D-D/A will not be compensated for when you patch your outboard gear.
This wont be a problem in most cases, but if you do parallel compression, you'll get time phasing. Have you tried this?
The Powecore uses the same chips, so the difference should be minimal. Unless you are talking about the integrated TDM mixer to avoid additional latency. That's still the only thing PT HD has on native DAW's. Otherwise, their is no point.
@freedom0speech That would be the old Motorola chips. The Accel chips are about 3 times more powerful. But still the Powercore induces latentcy so real-time usage is not an option (granted not a nessecity). My point is that with TDM archetecture, the entire project is performed on the cards. Even the track count and signal paths. It is all calculated within the TDM domain outside the computers resources, in real time. It is rock solid dependable and an HD3 ACCEL is ample.
Again, we have both put too much emphasis on the tools and not on the work. Do you think construction guys get online and bicker about which hammer you need or do you think what they can build is what is important? I only voice up because there is a lot of hatred against Pro Tools that is ill advised.Conversly there is a lot of ill advised praise to be fair. But if you had an HD/Accel rig and a C24 or an ICON, you wouldnt look back.You could still add the UADs if you wanted, but only in the mix.
I agree we should focus on the work and not the tools. But I can't stop myself when I read something that is demonstrably wrong... I have to correct it.
As I said previously, that is the only advantage left on PT HD. As for the reliability, I've worked in many PT HD studios which have had plenty of stability problems. I think confirmation bias comes into play here. So it's clipping vs. latency. I'd rather avoid clipping, as latency is so low it doesn't matter much anymore (under 1ms, plus delay compensation to avoid time phasing). However, yes, PT HD is low latency all the way through, as the mixer runs on the cards.
As for gain staging not being a problem. Tell that to all the hundreds of thousands of PT HD recordings on iTunes with clipping all over all the channels. Yeah, works great don't it....
Well, thanks for the video, but I don't know why would I put an 1024*768 image into a center of a huge 21 megapixels big black canvas. And after that, you zoom in to that tiny picture when you normalize your mix. You have 24 bits resoulution, use it! Your compressors will need it at the mastering.
@endrodyg your analogy's a bit extreme; -10 isn't unreasonable for a master mix, particularly one with no compression or processing on it. To gel the mix with some serious compression with require the master buss to be at a reasonable level beforehand; if you run any parallel compression this is then going to require even more headroom to begin with so I don't have a problem with -10 going in. Plenty of headroom for transients too plus you could arguably end up with a more "naturalistic" sound.
@christopherwoods Not really. I actually try not to exeed -6 but -10 is actually good advice in the 24 bit world. The mix master does not have to compete with a mastered production, and that is most peoples mistake; Trying to get that finished mastered sound too soon. Leaving ample headroom is good to give the guys in mastering dynamic range to work with. Those guys use TC 6000s which handle digital overs extremely well, or analog stuff that can run hot and bring up levels with no distortion.
@endrodyg also do you understand the concept of bitdepth in the digital domain? Provided your raw audio in is decent (and your ADCs are sufficient), having a higher resolution audio means your noisefloor is effectively much lower. So, you can go quieter - and crank the gain at a later point - without having the noisefloor increase to the point of annoyance. So if anything, mixing "quieter" is more easily accomplishable - and desirable - in the digital domain. Full scale shouldn't be desirable.
@christopherwoods Of course, you can go quieter but it will cost you resolution. I think -10 is a bit wasteful, I never use parallel compression after the mixdown so around -3 is good enough for me.
Hey buddy I dont wanna call u out or anything, I really like your videos and I like what youre doing. I just have one thing to add/ a correction. When recording in the digital domain, most audio engineers will encourage you to record with your peaks hitting above -6 dB. This has to do with the sampling rate. In short, in order to get the most out of what you want to record and not have any loss of signal, it has to be above -6dB, so that there are no frequencies or parts of the sound that are...
@junknztrunk09 not sampled when recording. This is what you are referring to when you hear people say that you "want to get your peeks as high as possible," and this is referring to recording not mixing. If you record with low peaks, you WILL end up bringing up the noise floor when you have to normalize it (bring up the volume) later on, in order to get it to a proper level for mastering (and yes there is a noise floor in the digital domain). In theory, it is also, best to have your master...
@junknztrunk09 bus have its peaks above -6 dB. This is because the same rules apply during the bounce, and also because you do not want to have a mastering engineer be forced to crush your song through heavy limiting, in order to bring your song up to a competitive listening level. Just thought I'd throw in my two cents...
@junknztrunk09 my solution to this is to record with high levels that do not clip, mix with low levels so you do not have to fight clipping, and then bring the levels back up when the mix is finished and before you bounce the song. You can do this by grouping the necessary tracks and then bringing them all up.
@junknztrunk09 THIS IS 16bit only!!! 24 bit your peaks should hit definately below -6, and its advisable to record at -12!!!
24 bit is much better if you've got it, most do, for the reason you don't have to boost up past 6 pushing the preamps and risking clipping. it should be used because noise floor is lower too. yes its advantageous to use it even if it will be put on 16bit cd becuase it makes actual tracking sound better.
@jorgepeterbarton when downsampling from 24- to 16-bit (at the final render stage) it's important to use the correct type and amount of dithering to minimise distortion and loss of fidelity during conversion. Bob Katz (he of the K Scale for measuring perceptual loudness) has written a very useful article on his web site (digido.com) called "Keeping Your Digital Audio Pure from First Recording to Final Master".
Also, highly advise every audio nerd to watch this video: watch?v=BYTlN6wjcvQ
@revolverrecords i did suspect that. i know -12 or -10 is right for tracking, but on the stereo track you've got no outside source and you know whether it will clip or not. I@m tracking 24bit for the first times, but yet to give out a master mix yet.
@junknztrunk09 That is old information and was the rule for 16 bit recording. The idea was to get level hot as possible before clipping in order to keep the noise floor low. Now in the 24 bit world, you have 144db headroom and there is no reason to record so hot since the noise floor is really really low now. Leaving about 3-6db headroom absolutely gives you a higher fidelity recording in 24 bit digital. The level can be made up later in mastering.
I new to your tutorials. They are helping me a lot to get things clear. Also, I'm spoting bad habits that can ruin my mixes, like turning up the volume on the master, or individual tracks instead of just turning up the volume on my headphones. Thanks a lot :D
First, thank you for the great videos! Quick question. If you screwed up and you have tracks that are "too loud" (not clipping but around -3) - can bringing down the level via "normalize" in audio suite correct the damage?
@recordingrevolution If you have PTHD, you can actually trim the fader levels bywhere all your automation will be brought down a relative level. It is an automation function that is really fast and handy, but I dont know if it has been introduced into LE.
Put on a mastering plugin on the mastering channel, turn down the output/input channel volume inside the plugin.
Whoala, works the same!
And for people wondering, this is not the same way as turning down the master fader, then your clipping is still there. But it will go away with this trick.
@Reaper1984 I use Izotope 4 in the master fader in my mix. Some tracks like snare track are clipping, but after going through the Izotope plug in, the master fader won't clip anymore. Is this a proper way yo avoid clipping? 'Cause when I brun a cd and then play it in my regular stereo I don't hear any distortion.
Hey, this is not a good way, especially for the lowend you notice it losses a lot in quality by doing it this way.
Add a equalizer on your mastering channel, and turn the volume down on the equalizer so the snare/mix is not clipping into Izotope, then bring the mastering up to the same volume as it was before.
Then compare the too, hopefully you hear a difference in sound quality!
I mean bring up the volume using Izotope, pushing it up so its the same volume as it was before you added the eq, and keep the eq as it is. Just make sure it's not clipping into Izotope, then your fine.
@Reaper1984 Thanks for the tip man! I'll try it. Anyway I followed the advices on this video and turned down a bit my tracks and resulted in a better and cleaner general sound, then I just turned up a bit the volume in izotope and got great sounding. Greetings from Chile
Seriously, I just went back and applied this to one of my songs. Instantly better sounding! I can only describe the track as having more space now. Thanks
Graham, do you perform this gain staging before any track or bus compression? Does it make sense to compress and then mix? Wouldnt compression work against you if you pre-mix before plugins?
@thefambaxley I do this before any plugins are used. I want to make sure the raw tracks are hitting the master fader conservatively. The compression is only the icing on the cake then.
@thefambaxley Actually, in the channels, the inserts are pre-fader. So basically your compression settings are unaffected by the fader gain. A very handy feature since every slide of the channel fader wont change your compression level. But good gain staging is excellent practice either way.
This is some of the best info any starting engineer can get! Everything he is saying is very true. I have been doing this for a while now and my mixes sound better than ever. Than when i master them and use compression and limiting they are just as loud as comercial recordings in the end. TAKE THIS VIDEO VERY SERIOUSLY. lol
Hi, i'd like to know if it is possible to download the videos, please. Theu are very helpful but i don't have internet access in my home studio. Thanks.
I just found your page today and subbed your vids are very simple and very helpful. But i record a lot of hip type music and typically the people i record use instrumentals of songs they like which have already been mastered. Pretty much they come to me with a mp3 two track file which im finding very hard to get their vocals to mix n sit with the mix if u could provide any insight on some steps i can take to better my mixes with those mp3 type instrumentals that would be awesome. Thanks
@kapone2323 This is really an EQ issue. You have a challenge for sure, but you need to "carve out" some EQ space in the MP3. It will take some experimenting.
So why not just turn the master fader down you will get the same results or insert a plugin into your master chain and lower the gain there, can you please explain clearly and quickly not too complicated answer as to why it is beneficial to lower the volume of each channel? What will happen if you mixed normally as in keeping the volumes at max without clipping as opposed to keeping the volume at -10 ?
@Sweetassour It's the internal mix bus that the channels are hitting too hot. I'd rather send a conservative level to the mix bus before it hits the master fader than do it "after the fact".
Nice video! Right on the spot - headroom is not a head the size of a room, but one's head might feel like the size of a room when gain staging is wrong. Now, seriously, Paul Frindle has the best technical explanation on this issue. Just "google" his name.
Graham got the samson 8mic kit and they have tracked so unbelievably well! I am very happy with them thanks so much for the recommendation! Hey I have a question that really doesn't seem to be answered in any other post unless I missed it from you or anyone else.. I saw you had a Sub Master and a Submix Channel one being a master fader and the other being an aux what are the purposes and advantages of having that over just a master fader? Sorry for the long one but thanks in advance!
Is there a way to set the 'input gain' on recorded tracks pre fader? So that you can have a level static pre mix that is not dependent on using the faders. I'd like to have my faders close to zero with balanced levels and plenty of Master Buss headroom on each track before I start making fader moves. Since faders use a logarithmic scale, I don't want to compromise useful fader range by using faders to set my initial channel levels.
appreciate the videos you're doing! i'm used to mixing vocals over a beat but not really a beat i made myself so stuff like this will definitely help :)
@romodrummerjz Like most of his videos, I think he just uses Pro Tools as his conduit for teaching, so while some of the commands and tools may be Pro Tools specific, the concepts are generally useful for all DAWs. This is what I'm expecting. I hope that answers your question! =)
I must commend you for enlightening the viewers about gain staging because many novice mixers don't even consider it. Digital metering is rubbish! So with that said, nice info keep up the great work.
Isn't the noise floor going to essentially be raised if the overall level is lowered?
sonicfuker 7 hours ago
I read that the enemy in the digital domain is not noise floor, but the farther you go down in level from reference, distortion goes thru the roof. So much so that a sine wave can actually be distorted into a square wave. What are your thoughts?
sonicfuker 2 days ago
@sonicfuker was it written in 1995?
rillloudmother 1 day ago
@rillloudmother Sorry, I don't remember the source. Why do you ask? It was discussing why dither and limiting is applied to commercial mixes to push the signal closer to reference to avoid distortion.
sonicfuker 1 day ago
@sonicfuker that's not why dither and limiting are used on commercial mixes. dithering prevents bit truncation. limiting is used for many purposes, but in professional mixing and mastering it is not used for that purpose. limiting is often used when tracking to prevent digital overs which result in distortion if the signal exceeds 0dbfs.
rillloudmother 1 day ago
@rillloudmother So there is no truth to distortion based on lower levels?
sonicfuker 7 hours ago
Even the self-styled Master of His Tools has to be open to the corrective input from those who know more than them about other than the Software they have finally chosen to champion.
It's called OPEN-MINDEDNESS.
What a big mistake it ever is to think you yourself KNOW IT ALL about everything.....Like me! LOL : )
hattonhall 4 days ago
Should be called "5 mins to fix your bad mix and be lazy."
petefaders 4 days ago
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Hi :)...
This is a great introduction for "Gain Staging". a starting point of -10 dBFS is nice ( and the master gain @ Unity ). but without a good explanation of dBFS Peak and dBFS Rms and without a very good vu meter... it's not possibile to make a correct gain staging.. not to mention the plugins and their gain once they are inserted on a channel to eq compress o limiting a program.
A more deep tutorial and explanation on this subject are expected :).
Thanks for now. :)
MrDanilong 5 days ago
great tips man keep em coming.
swiftomatik 6 days ago
that mix sounds GOOD!
pifa4life 1 week ago
Hey brother, keep up the good work! I have been watching your vids and their are a lot of haters or side line producers, a trying to put down you style. Keep up the vids, I am, thankful for your vids and info. I dont use pro Tools, but good knowledge and info is interchangeable, and IT WORKS! Cheers
AroundTheWaymon 1 week ago
what do you use the master fader track for if all your tracks are being send to the AUX Submix?
IScH45I 1 week ago in playlist Mix I
@IScH45I lol eactly, that's what it's for. If the level is not right hitting the master fader, fix your mix, don't pull this trick with a needless bus..I think we agree.
petefaders 4 days ago
Your group-all setting also pulls down your submaster and -mixfader (masterfader). I assume this isn't intentional.
dkyx 1 week ago
@dkyx it's happened to me, i do the same thing in this video. Simply turn the group off and put your master back to unity.
DJyorkei 1 week ago
the part about keeping the mix under -10db is just unfounded, and a myth. As long as you are not clipping, there is absolutely no difference in sound between -0.1 and -20, except the noise floor (which does exist by the way, even in the digital era).
JohnTheRipper2 1 week ago
so if my stereo peaks at -12 or so i get the volume back when i master?
i have noticed that it sounds better at this level. although this could be due to my headphones interface having to boost.
jorgepeterbarton 2 weeks ago
These videos are really great. Thanks a lot for sharing your knowledge.
hdkhfkd 2 weeks ago
I thought as long as you are not exciding the 0db level that no clipping is occuring and everything is okay? Also What if i get my mix all set in the DAW and ready to sum, I instead send each track back out of the DAW and into an analog mixer to be sumed via the mixer, then use the stereo outputs of the mixer to creat a stereo track of all my tacks that have been summed in the analog domain? Should i then set the gain levels on the mixer to unity gain? Great Vidoes by the way!!
2500vic 2 weeks ago
This is irrelivant if you use anything besides Pro Tools, because every other platform has advanced to floating point back in the 1990's, so no clipping occurs unless it's on the input or output. No clipping can happen in the mixer or plugins... not true of Slow Tools.
freedom0speech 3 weeks ago 4
@freedom0speech Pro Tools is floating point, (not HD) always has been. Clipping is a real issue in all digital systems because it doesn't sound good. Proper gain staging is critical in the analog domain as well, so no this isn't irrelevant outside of Pro Tools.
recordingrevolution 3 weeks ago 12
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@recordingrevolution
Gain staging is important for doing A/B comparisons, but what plugins are you using that can clip? Every VST plugin I've ever used is floating point for about 15 years now. You can only clip on the converters (in or out).
But yes, it's also important to understand this for outboard gear (both digital and analog).
freedom0speech 3 weeks ago 3
@recordingrevolution I think this guy's on drugs @freedom0speech
Swainphil 3 weeks ago
@recordingrevolution I think this guy's on drugs. Of course it applies to all producing in all DAW's
Swainphil 3 weeks ago
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@recordingrevolution I think this guy's on drugs. Of course it applies to all producing in all DAW's
Swainphil 3 weeks ago
@recordingrevolution If your peak levels dont exeed -3 you are probably ok. But Normalize wont bring the levels down. Normalize brings level up till the peaks hit all the way up to 0Dbu.
TheProgmagog 1 week ago
@TheProgmagog You might mean "Gain" command;) But assuming you had "damage" it wont be fixed by bringing levels down after the fact. It will only lower the volume but the clips would still be there, all beit quieter overall, still clipped. But at -3 peak level you probably dont have clipping.
TheProgmagog 1 week ago
@recordingrevolution I've used Logic Pro for years now, I don't remember exactly what happens in ProTools but I don't think I have ever heard clipping in Logic Pro unless its on input or the stereo bus. Is ProTools the same way?
CountToBen 1 week ago
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freedom0speech 3 weeks ago
@freedom0speech This is a misunderstanding of where the clipping is occurring. at 24 bits, plenty of heardoom equals higher fidelity in any DAW. BTW PTHD7 uses a 48 bit mix bus archetecture. That is wider than even 32 bit floating point.
TheProgmagog 1 week ago
@TheProgmagog
24bit and 48bit only allows for more dynamic range UNDER 0dB. It doesn't add headroom above. So no, that will not fix the clipping problem.
Modern DAW software uses a combo of 32bit and 64bit floating point. All my compressors and EQ's are 64bit floating point, and the rest are 32bit floating point. No clipping anywhere, unless I clip the A/D or D/A converters. It's been this way outside of SloTools since the year 2000 or so.
freedom0speech 1 week ago 3
@freedom0speech Headroom above 0db? 64 bit since 2000? I doubt that, but rock on anyway. I think PTLE was a big mistake because now the HD rig gets lumped in with the rest. The TDM system is leagues beyond any native DAW. But regardless of that, its not the bycycle, its the rider. If gain staging in my PTHD rig is more critical than others, so be it. There is a reason the old school veterans have mostly moved to PTHD and ICONS. I dont think they are scratching their heads about it. Neither am I
TheProgmagog 1 week ago
Having said that, I think it is wreckless practice to assume gain staging is not relevant at any bit rate. What DAW are you using BTW? I am only deeply familiar with Logic Pro and Pro Tools HD.I have a passing rapport with SONAR, Cubase and Reaper. If I liked them more I certainly wouldn't have shelled out so much for an HD3 Accel. All this mathematics aside, the workflow is awesome and the ability to use my analog 1176s and Pultec and so on in real time easy as a plug-in is a whole other world.
TheProgmagog 1 week ago
@TheProgmagog
It's not about allowing yourself to be reckless. Its about having safe dB's above 0dB to avoid clipping, because **** happens. Hell, listen to all the music coming out these days done on PT HD, these engineers are clipping at what seems like every stage. It's reckless now to not add that safety net, and I'm sick of listening to the clipping.
Studio201recording 1 week ago
@Studio201recording True enough my friend. Which brings us back to the purpose of this tutorial. S#*t happening is totally avoidable and here is the way to avoid it. Having a safety net is cool, but the danger is leaving no headroom for them to work with in mastering. It is always good practice to leave I would say 6db, of headroom on your mix bus and 3-6 on your recorded tracks (to preserve transients) Leaving about 6db on the mix bus is something your mastering engineer will praise you for.
TheProgmagog 1 week ago
@TheProgmagog
You can doubt it all you want. It's a fact. The UAD-1 processed at 64bit using a 32bit floating point engine (on all native DAW's at the time) around 2000.
As for why people use PT HD, it's simply because they copy each other like you just explained. Famous people use it, therefore it must be great... we don't know why, but it just must, they must know something. LOL! Please.
freedom0speech 1 week ago 3
@freedom0speech I have 2 UAD 1 cards. Great sound but its not the same as a truly integrated TDM system. People like Kevin Killen, Ed Cherney and Andy Johns arent just blindly following, while guys like you got the real info. You dont need Pro Tools to do great work, but we that use it (HD) know what we are doing and why. Your numbers game and reasoning mean nothing to a guy that is confident in his work. We dont need to emphasize our gear but rather the descisions we make with it.
TheProgmagog 1 week ago
@TheProgmagog
Listen, I agree that it's all about the talent, not the gear... however, I also can't stand bullshit. I'm sorry but the TDM system isn't even powerful enough to handle most of the UAD plugins in all their glory.
You can make great music on PT HD, but I don't understand why anyone would want the headache. But, that's up to you. However, if someone tells me 48bit fixed is better than 32bit float with 64bit float abilities, I'm going to call them on it.
freedom0speech 1 week ago 2
@TheProgmagog
Also, appeals to popularity and appeals to authority mean nothing to me. I've worked with lots of famous people, and the most important thing I learned is that they are human beings like everyone else, and they do stupid shit (like copy everyone else for illogical reasons) like the rest of us. I don't care what they use, I care what sounds good and what works good.
freedom0speech 1 week ago 3
@freedom0speech You are very right on your point. I am not pointing out merely "famous" people but icons in the field who hesitated to go digital for a long time, and finally turned in their SSLs for HD rigs and ICONs. A big part of the turn is not only PT but Waves as well. The two in conjunction along with a good controller like even the C24 TAKES AWAY all the headache I associate with DAW recording. But I have already taken too much time on technology. Its all about the music.
TheProgmagog 1 week ago
BTW, I get WAAAAY more out of my TDM accel cards than my UAD cards. In fact my UADs and my TC Powercore are now packed in bubble wrap. What I like with TDM is the real time usage. Its integrated with the entire DAW so you can monitor effects while tracking if you like. Also, whatever you can load, you can always use flawlessly. The UAD cards load more than they can handle and if pushed too hard they crap out.
TheProgmagog 1 week ago
The reason I feel so many top industry guys have moved to PTHD are many. First with its true real time capabilities both with plug-ins and outboard hardware, along with the outstanding integration with major controllers like the ICON or even the C24 has made the work experience extremely similiar to the old hardware-analog days. That and its rock-solid stability is something they require in such high dollar-high pressure situations.
TheProgmagog 1 week ago
@TheProgmagog
I've been running my entire Nuendo DAW at under 1ms latency for the last 8 years, and it integrates into outboard gear flawlessly. These advantages are 90's ProTools advantages. It's 2012 now.
freedom0speech 1 week ago 3
@freedom0speech Nuendo rocks! But let me ask, cause I dont know. For example, I have my ISA430, Pultec, 1176s, Requisite PAL III and others patched into my interface ins and outs. Of course I can record through them as I did in Logic. But what I couldnt do in Logic is open said device right in the insert of an Audio Track and mix it on the fly. Logic has Aux in that could be used, but not simply and directly in the audio track path. Can you do this? DO you do this?
TheProgmagog 1 week ago
@TheProgmagog
Correct me if this isn't what you are talking about, but Steinberg added an outboard FX insert plugin in about Nuendo 4.0 or so (we're on 5.x now), so you just drop that in your inserts--wherever you want your outboard gear patched in--and assign the inputs and outputs to your audio device.
Is that what you mean?
freedom0speech 1 week ago
@freedom0speech Sounds about right.The way I have it (and I assume you can do the same) is, Like I said, all the gear is permanently patched. I can access it on a channel input for recording, or I can open it on a channel insert (or bus insert) just like a plug-in. My IOs are all labeled by the device name so its real quick and easy. I can open an 1176 bomb factory or the real deal just as easily. But in order to have 1ms latentcy, dont you have to set your buffer low? I ask because...
TheProgmagog 1 week ago
... The UAD and Powercore require substantially high buffer settings. BTW did you know that Chris Squire can hear 1ms of delay? That is something Trevor Horn pointed out to NED when they were designing the Synclavier.
TheProgmagog 1 week ago
@TheProgmagog
The UAD plugins do add additional latency, yes. I hit the "constrain delay compensation" button when tracking, which disabled plugins with latency above 3ms (or whatever I set it to), and then disable that when getting back to mixing.
freedom0speech 1 week ago
@freedom0speech In summary, I think your choice of tools is absolutely viable and any difference between your sound and anyone elses will come down to how you use what you have. As for me, I am a musician and singer first. I am fortunate enough that my music has allowed me to afford the rig I really wanted and I am really happy as well. As many reasons I can say that I love Pro Tools HD, I can counter with reasons I hate Avids business practicees. Therefore I dont upgrade. I just use what I have
TheProgmagog 1 week ago
@TheProgmagog
Hey, I'm glad you gain-stage using PT. Too many don't and I have to listen to the clipping as a result. I think this could be avoided, but I guess there would be something else lazy engineers would do wrong... lol.
freedom0speech 1 week ago
@freedom0speech Anywhere I can here some of your stuff?
TheProgmagog 1 week ago
@TheProgmagog
As for hearing 1ms latency, I believe it. I've heard some people say that no human can hear below 20ms, which is nonsense. It all depends on what we are listening to. If you can hear it phase against the original signal, we can probably hear 0.1ms in some cases.
freedom0speech 1 week ago
@freedom0speech That is exactly what NED was trying to say to Trevor Horn. If that were indeed the case, we wouldnt be using about 20ms pre-delay into our reverbs:)
TheProgmagog 1 week ago
@TheProgmagog
Off-topic, but I've been using 40-50ms pre-delay on my reverb's for my last mixing project, with decays well under 1 second. It seems to create an awesome 3D sound somewhere between a delay and a reverb. I used to mix delays and reverb's, but it sounded mushy, this sounds clean. Anyhow, that's a neat little trick I thought I'd mention, however very off topic.
freedom0speech 1 week ago
@TheProgmagog
Keep in mind that you will get similar latency on PT HD when you patch in and out of the system. Though you may not notice it until you patch several times in a row.
Latency issues are getting better with time, but they are far from perfect on any system. They will likely never be perfect, but they should get so low that they become irrelevant soon.
freedom0speech 1 week ago
@TheProgmagog
I should mention that patching in outboard gear like this in Nuendo will not add latency to your mix, because their is delay compensation applied.
I don't think Pro Tools has this yet, unless I'm wrong, but I think the latency (however small) added by the TDM chips and the A/D-D/A will not be compensated for when you patch your outboard gear.
This wont be a problem in most cases, but if you do parallel compression, you'll get time phasing. Have you tried this?
freedom0speech 1 week ago
@TheProgmagog
The Powecore uses the same chips, so the difference should be minimal. Unless you are talking about the integrated TDM mixer to avoid additional latency. That's still the only thing PT HD has on native DAW's. Otherwise, their is no point.
freedom0speech 1 week ago 2
@freedom0speech That would be the old Motorola chips. The Accel chips are about 3 times more powerful. But still the Powercore induces latentcy so real-time usage is not an option (granted not a nessecity). My point is that with TDM archetecture, the entire project is performed on the cards. Even the track count and signal paths. It is all calculated within the TDM domain outside the computers resources, in real time. It is rock solid dependable and an HD3 ACCEL is ample.
TheProgmagog 1 week ago
Again, we have both put too much emphasis on the tools and not on the work. Do you think construction guys get online and bicker about which hammer you need or do you think what they can build is what is important? I only voice up because there is a lot of hatred against Pro Tools that is ill advised.Conversly there is a lot of ill advised praise to be fair. But if you had an HD/Accel rig and a C24 or an ICON, you wouldnt look back.You could still add the UADs if you wanted, but only in the mix.
TheProgmagog 1 week ago
@TheProgmagog
I agree we should focus on the work and not the tools. But I can't stop myself when I read something that is demonstrably wrong... I have to correct it.
freedom0speech 1 week ago
@TheProgmagog
As I said previously, that is the only advantage left on PT HD. As for the reliability, I've worked in many PT HD studios which have had plenty of stability problems. I think confirmation bias comes into play here. So it's clipping vs. latency. I'd rather avoid clipping, as latency is so low it doesn't matter much anymore (under 1ms, plus delay compensation to avoid time phasing). However, yes, PT HD is low latency all the way through, as the mixer runs on the cards.
freedom0speech 1 week ago
@TheProgmagog
As for gain staging not being a problem. Tell that to all the hundreds of thousands of PT HD recordings on iTunes with clipping all over all the channels. Yeah, works great don't it....
No. It's time to ditch PT HD.
freedom0speech 1 week ago 3
@TheProgmagog
Yes. floating point gives you safe (and unlimited) headroom above 0dB, without clipping.
freedom0speech 1 week ago 3
Well, thanks for the video, but I don't know why would I put an 1024*768 image into a center of a huge 21 megapixels big black canvas. And after that, you zoom in to that tiny picture when you normalize your mix. You have 24 bits resoulution, use it! Your compressors will need it at the mastering.
endrodyg 3 weeks ago
@endrodyg your analogy's a bit extreme; -10 isn't unreasonable for a master mix, particularly one with no compression or processing on it. To gel the mix with some serious compression with require the master buss to be at a reasonable level beforehand; if you run any parallel compression this is then going to require even more headroom to begin with so I don't have a problem with -10 going in. Plenty of headroom for transients too plus you could arguably end up with a more "naturalistic" sound.
christopherwoods 2 weeks ago
@christopherwoods Not really. I actually try not to exeed -6 but -10 is actually good advice in the 24 bit world. The mix master does not have to compete with a mastered production, and that is most peoples mistake; Trying to get that finished mastered sound too soon. Leaving ample headroom is good to give the guys in mastering dynamic range to work with. Those guys use TC 6000s which handle digital overs extremely well, or analog stuff that can run hot and bring up levels with no distortion.
TheProgmagog 1 week ago
@christopherwoods Oh Sorry! "Isn't unreasonable" you said... Never mind;) You got it.
TheProgmagog 1 week ago
@endrodyg also do you understand the concept of bitdepth in the digital domain? Provided your raw audio in is decent (and your ADCs are sufficient), having a higher resolution audio means your noisefloor is effectively much lower. So, you can go quieter - and crank the gain at a later point - without having the noisefloor increase to the point of annoyance. So if anything, mixing "quieter" is more easily accomplishable - and desirable - in the digital domain. Full scale shouldn't be desirable.
christopherwoods 2 weeks ago
@christopherwoods Of course, you can go quieter but it will cost you resolution. I think -10 is a bit wasteful, I never use parallel compression after the mixdown so around -3 is good enough for me.
endrodyg 2 weeks ago
Thank you so much for these videos! They are pure gold!!
Yardsticky 3 weeks ago
Brilliant.
tmbasser 4 weeks ago
It's about time that people understand digital recording!! Great and simple explanation of gain staging! Bravo
bbbkink1 1 month ago
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Panosmast79 1 month ago
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Panosmast79 1 month ago
Hey buddy I dont wanna call u out or anything, I really like your videos and I like what youre doing. I just have one thing to add/ a correction. When recording in the digital domain, most audio engineers will encourage you to record with your peaks hitting above -6 dB. This has to do with the sampling rate. In short, in order to get the most out of what you want to record and not have any loss of signal, it has to be above -6dB, so that there are no frequencies or parts of the sound that are...
junknztrunk09 1 month ago 2
@junknztrunk09 not sampled when recording. This is what you are referring to when you hear people say that you "want to get your peeks as high as possible," and this is referring to recording not mixing. If you record with low peaks, you WILL end up bringing up the noise floor when you have to normalize it (bring up the volume) later on, in order to get it to a proper level for mastering (and yes there is a noise floor in the digital domain). In theory, it is also, best to have your master...
junknztrunk09 1 month ago
@junknztrunk09 bus have its peaks above -6 dB. This is because the same rules apply during the bounce, and also because you do not want to have a mastering engineer be forced to crush your song through heavy limiting, in order to bring your song up to a competitive listening level. Just thought I'd throw in my two cents...
junknztrunk09 1 month ago
@junknztrunk09 my solution to this is to record with high levels that do not clip, mix with low levels so you do not have to fight clipping, and then bring the levels back up when the mix is finished and before you bounce the song. You can do this by grouping the necessary tracks and then bringing them all up.
junknztrunk09 1 month ago
@junknztrunk09 THIS IS 16bit only!!! 24 bit your peaks should hit definately below -6, and its advisable to record at -12!!!
24 bit is much better if you've got it, most do, for the reason you don't have to boost up past 6 pushing the preamps and risking clipping. it should be used because noise floor is lower too. yes its advantageous to use it even if it will be put on 16bit cd becuase it makes actual tracking sound better.
jorgepeterbarton 2 weeks ago
@jorgepeterbarton when downsampling from 24- to 16-bit (at the final render stage) it's important to use the correct type and amount of dithering to minimise distortion and loss of fidelity during conversion. Bob Katz (he of the K Scale for measuring perceptual loudness) has written a very useful article on his web site (digido.com) called "Keeping Your Digital Audio Pure from First Recording to Final Master".
Also, highly advise every audio nerd to watch this video: watch?v=BYTlN6wjcvQ
revolverrecords 1 week ago
@revolverrecords i did suspect that. i know -12 or -10 is right for tracking, but on the stereo track you've got no outside source and you know whether it will clip or not. I@m tracking 24bit for the first times, but yet to give out a master mix yet.
jorgepeterbarton 1 week ago
@junknztrunk09 That is old information and was the rule for 16 bit recording. The idea was to get level hot as possible before clipping in order to keep the noise floor low. Now in the 24 bit world, you have 144db headroom and there is no reason to record so hot since the noise floor is really really low now. Leaving about 3-6db headroom absolutely gives you a higher fidelity recording in 24 bit digital. The level can be made up later in mastering.
TheProgmagog 1 week ago
I think i´m in love with you..
totalovrkl 1 month ago
I new to your tutorials. They are helping me a lot to get things clear. Also, I'm spoting bad habits that can ruin my mixes, like turning up the volume on the master, or individual tracks instead of just turning up the volume on my headphones. Thanks a lot :D
vicsanmusic 2 months ago
thanks a ton man!
ConnorMcAuliffeDrums 2 months ago
First, thank you for the great videos! Quick question. If you screwed up and you have tracks that are "too loud" (not clipping but around -3) - can bringing down the level via "normalize" in audio suite correct the damage?
johntguitar 4 months ago
@johntguitar I would either just turn down the faders or insert the TRIM plugin on the track to statically turn down the gain.
recordingrevolution 4 months ago
@recordingrevolution If you have PTHD, you can actually trim the fader levels bywhere all your automation will be brought down a relative level. It is an automation function that is really fast and handy, but I dont know if it has been introduced into LE.
TheProgmagog 1 week ago
Good tip, but a even better tip is this.
Put on a mastering plugin on the mastering channel, turn down the output/input channel volume inside the plugin.
Whoala, works the same!
And for people wondering, this is not the same way as turning down the master fader, then your clipping is still there. But it will go away with this trick.
Reaper1984 4 months ago
@Reaper1984 I use Izotope 4 in the master fader in my mix. Some tracks like snare track are clipping, but after going through the Izotope plug in, the master fader won't clip anymore. Is this a proper way yo avoid clipping? 'Cause when I brun a cd and then play it in my regular stereo I don't hear any distortion.
thetriumphofsatan 2 months ago
@thetriumphofsatan
Hey, this is not a good way, especially for the lowend you notice it losses a lot in quality by doing it this way.
Add a equalizer on your mastering channel, and turn the volume down on the equalizer so the snare/mix is not clipping into Izotope, then bring the mastering up to the same volume as it was before.
Then compare the too, hopefully you hear a difference in sound quality!
Reaper1984 2 months ago
@thetriumphofsatan
I mean bring up the volume using Izotope, pushing it up so its the same volume as it was before you added the eq, and keep the eq as it is. Just make sure it's not clipping into Izotope, then your fine.
Reaper1984 2 months ago
@Reaper1984 Thanks for the tip man! I'll try it. Anyway I followed the advices on this video and turned down a bit my tracks and resulted in a better and cleaner general sound, then I just turned up a bit the volume in izotope and got great sounding. Greetings from Chile
thetriumphofsatan 2 months ago
Brother, Amazing vid!! I'm going to share your vid series to my students!!
saxotron 4 months ago
Seriously, I just went back and applied this to one of my songs. Instantly better sounding! I can only describe the track as having more space now. Thanks
SeeYouInBluffington 4 months ago
Graham, do you perform this gain staging before any track or bus compression? Does it make sense to compress and then mix? Wouldnt compression work against you if you pre-mix before plugins?
thefambaxley 6 months ago
@thefambaxley I do this before any plugins are used. I want to make sure the raw tracks are hitting the master fader conservatively. The compression is only the icing on the cake then.
recordingrevolution 6 months ago
@thefambaxley Actually, in the channels, the inserts are pre-fader. So basically your compression settings are unaffected by the fader gain. A very handy feature since every slide of the channel fader wont change your compression level. But good gain staging is excellent practice either way.
TheProgmagog 1 week ago
Wow, this is a great series! I've been mixing about a year, and this is a gold mine!
stebbi01 6 months ago
This is some of the best info any starting engineer can get! Everything he is saying is very true. I have been doing this for a while now and my mixes sound better than ever. Than when i master them and use compression and limiting they are just as loud as comercial recordings in the end. TAKE THIS VIDEO VERY SERIOUSLY. lol
dmoates420 6 months ago
Hi, i'd like to know if it is possible to download the videos, please. Theu are very helpful but i don't have internet access in my home studio. Thanks.
Parrampito 7 months ago
@Parrampito Right now they are only on YouTube to keep them free.
recordingrevolution 7 months ago
@Parrampito You can download youtube videos to your computer for free at zamzar(dot)com
Musicmonk84 2 months ago
I just found your page today and subbed your vids are very simple and very helpful. But i record a lot of hip type music and typically the people i record use instrumentals of songs they like which have already been mastered. Pretty much they come to me with a mp3 two track file which im finding very hard to get their vocals to mix n sit with the mix if u could provide any insight on some steps i can take to better my mixes with those mp3 type instrumentals that would be awesome. Thanks
kapone2323 7 months ago
@kapone2323 This is really an EQ issue. You have a challenge for sure, but you need to "carve out" some EQ space in the MP3. It will take some experimenting.
recordingrevolution 7 months ago
So why not just turn the master fader down you will get the same results or insert a plugin into your master chain and lower the gain there, can you please explain clearly and quickly not too complicated answer as to why it is beneficial to lower the volume of each channel? What will happen if you mixed normally as in keeping the volumes at max without clipping as opposed to keeping the volume at -10 ?
Sweetassour 8 months ago
@Sweetassour It's the internal mix bus that the channels are hitting too hot. I'd rather send a conservative level to the mix bus before it hits the master fader than do it "after the fact".
recordingrevolution 8 months ago
hi, what song is this that you have in the mix? its really good
soundslykmusic 8 months ago
I wonder if this is as important in a 32bit float environment?
Sungodv 8 months ago
@Sungodv yes it is important no matter what bit your daw is.
dmoates420 6 months ago
This has been flagged as spam show
Nice video! Right on the spot - headroom is not a head the size of a room, but one's head might feel like the size of a room when gain staging is wrong. Now, seriously, Paul Frindle has the best technical explanation on this issue. Just "google" his name.
leosaramago00 9 months ago
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leosaramago00 9 months ago
Very good!
silencekj 9 months ago
Graham got the samson 8mic kit and they have tracked so unbelievably well! I am very happy with them thanks so much for the recommendation! Hey I have a question that really doesn't seem to be answered in any other post unless I missed it from you or anyone else.. I saw you had a Sub Master and a Submix Channel one being a master fader and the other being an aux what are the purposes and advantages of having that over just a master fader? Sorry for the long one but thanks in advance!
romero0519 9 months ago
@romero0519 I recently did a video on this. Look at my channel for the Master Faders In Pro Tools video for my explanation.
recordingrevolution 9 months ago
@recordingrevolution Thank you so much Graham, don't know how i missed that one! Will watch it now! Thanks again!
romero0519 9 months ago
Is there a way to set the 'input gain' on recorded tracks pre fader? So that you can have a level static pre mix that is not dependent on using the faders. I'd like to have my faders close to zero with balanced levels and plenty of Master Buss headroom on each track before I start making fader moves. Since faders use a logarithmic scale, I don't want to compromise useful fader range by using faders to set my initial channel levels.
OpenDoerr 9 months ago
@OpenDoerr trim plugin
telecomvideo 6 months ago
great job man!!! even though I know this... I am always checking your channel .
Your doing a great job.
Shared it on my facebook for all the musicians to see...
good luck
amirhe6 9 months ago
awesome thanks! looking forward to the rest!
futbol1122 9 months ago
Excellent channel! Keep up the great work, we ALL benefit from your beautiful explanations.
212musicman 9 months ago
appreciate the videos you're doing! i'm used to mixing vocals over a beat but not really a beat i made myself so stuff like this will definitely help :)
islowmusic 9 months ago
This has been flagged as spam show
Very Nice Tip!
videosongman 9 months ago
Basic! Love it :)
Thanks!
skonrokk 9 months ago
are these 5 minute tips crucial to just pro tools or can it be done in logic too?
romodrummerjz 9 months ago
@romodrummerjz Like most of his videos, I think he just uses Pro Tools as his conduit for teaching, so while some of the commands and tools may be Pro Tools specific, the concepts are generally useful for all DAWs. This is what I'm expecting. I hope that answers your question! =)
LukeSock 9 months ago
@LukeSock Boom! Well said.
recordingrevolution 9 months ago
@LukeSock thank you
romodrummerjz 9 months ago
@romodrummerjz LukeSock said it best. This applies to any DAW :-)
recordingrevolution 9 months ago
Very Good! Thanks! :)
djwiesys 9 months ago
Thank you, looking forward to the next video!
Lokjutus 9 months ago
hey Graham, your videos are great! and you they have helped me so much with my home recording! but i was wondering if you could help me?
I use logic and was wondering what is the difference between the Output fader and the Master fadar?
weeman308 9 months ago
I must commend you for enlightening the viewers about gain staging because many novice mixers don't even consider it. Digital metering is rubbish! So with that said, nice info keep up the great work.
anytingtv 9 months ago
Great way to start off this series! Thanks Graham!
AngelBLive 9 months ago
Yass! Can't wait for the rest of these.
ScottySniper93 9 months ago