Thankyou so very much for a precise, un-cluttered tutorial covering essential recording basics with hardware and also a wonderful introduction to tracking with compression. Was very good as well to understand why exactly we should shoot for -7 to -10 db peak when tracking. Removing signal to noise ratio and also providing good signal for plug-ins. I've often noticed hiss on some tracks when I've tracked -12 / -14 peak and did'nt understand exactly why. I followed your recommendations and my tracks have become a lot cleaner and easier to mix. Well done !
Great, man! Being a rock guitarist, I 've always had a "crank' em all" kind of approach when performing live, but recording (especially in the digital domain) is a whole different thing. Simple and clear video for everyone to understand! 👏👏👏 Cheers!
Very helpful information! I would, however, like to put in my two cents and say that instead of having the gain meter of the preamp peak at 0dB, I think it should peak rather at -3dB. That way, this will ensure that actual clipping from whatever transducer you are recording from the preamp into your DAW doesn’t occur.
THANK YOU ! FINALYY, someone that GETS IT ! There is so much misinformation, disinformation and just plain WRONG information about this VERY important subject. I have ALWAYS recorded as hot as possible WITHOUT clipping, and yes I record vocals WITH compression. I typically record with the compressor at -10 dB threshold, and 4 to 1 compression. I aim for -3 dB max. What's with these 'engineers' that record at -18 dBs ? This seems to be the latest trend. PLEASE ! Thanks again for an excellent presentation ! Bill P.
I've had trouble understanding the signal process and how everything is connected to each other, and how to set levels and which levels to set, etc., etc. This very clear instructional tutorial was invaluable to me. It's well explained which gave me a way better understanding of how to go about recording. So invaluable. I liked and subscribed because I'm sure I'll be able to learn more from you. Well done!
Man I was doing it all wrong lol. I was recording with a very low signal and my daw track turned way down then after recording i would crank it back up duplicate the vocals to another track add effects and mix them together to make them sound loud and full. Rebuilding my studio now and am relearning from the basics
It has to be said that compressing a signal BEFORE recording it should only be attempted if you really know what you are doing and what kind of sound you are after. If not, this technique will needlessly reduce the dynamic range and degrade the signal/pull up noise. In most instances, it would be best practice to get the signal into the daw and then do compression later, be it in the DAW or via outboard. That way, everything is nondestructive and therefor reversable. There can be instances with unexperienced musicians in combination with unforgiving ADC, where you will need that kind of compression, to allow for a signal that won't clip at the ADC. But as long as you got a good source, a good mic a good preamp and a good converter, leave that compressor out of there if at all possible.
@@forkmusic If you have a setup that allows this, then I certainly would advice it. But the more common workflow would be to record dry and then send channels through plugins or outboard gear afterwards.
This video was extremely helpful, thank you so much I just got an outboard compressor and preamp. This video really helped me understand them and get the best settings possible
H i, just found your channel ,thank god for someone who has the ability to explain things clearly and at a pace that works.It helps us out here that want to learn .Thanks for passing on your knowledge.
you are so cool, i really appreciate your studio making series, i was researching this topic for many many hours and i guarantee you that your videos were the best. Im making my own home studio now and i will use your experience properly :p
Excellent video! Cleared lot of questions floating in my head. I have a home studio that I've been tinkering around with for a couple years and first started off using a Zoom R8 standalone. Eventually I upgraded and got Reaper DAW, and now use the R8 as an audio interface via USB. Works pretty good, but I have noticed that my sound quality playback (particularly acoustic guitar) doesn't have the that realistic crispness and fullness that I hear in the headphones when recording. I tried a comparison with a friend that uses a preamp with his Focusrite AI. It was a noticeably better sound. At first, I thought it might be the R8, but the more I research I think I need a preamp with the proper gain staging. I may invest in a preamp and see what happens.
As other videos just show input levels adjusted around -6dB to avoid clipping, the idea of initial level gain at 0 seems a bit odd. Beyond that, there is still one basic mic level that either is or is not at an optimal level, and running one unit into the next and the next and the next just looks like a great way to introduce noise, with no apparent intrinsic benefit. The concept of clean signal in, to DAW, then apply edits and adjustments to desired output just looks a lot more sensible and far less expensive.
Was thinking the same about the input level being 0 dB doest that just mean it's gonna clip? As for the hardware gear he has, audio interfaces have these components in them already (pre amps, eqs, adda Converters), they're just all smashed into one little box at an affordable price which means they arnt very good quality, buying quality hardware components sepratly will give you a much better quality signal than any audio interface as long as the right gear is used for whatever your recording and it's used correctly
@@indyfreeman7172 The concept of assembling a series of separate, "superior quality" components is not always valid. One sub-par component brings the rest down, and a degraded signal can't be improved by the rest. Each added part may introduce noise, hum, and require balancing levels between every separate unit - a recipe for problems. Integrated systems that have solid reviews from tech experts are likely to be far easier to use, and the built-in filters, amps, equalizers, converters, etc are designed for compatibility and streamlined control. Any slight loss in one area is probably not going to be that big a deal, in the end. Unless one enjoys the yards of tangled cables, and hours spent balancing and troubleshooting, most folks will be better off practicing and playing.
@@z1522 yea that's true it can cause problems that's why i said it's only the case if you use the correct gear and know how to use it. I'm not saying you can just chuck any pre amp and and adda converter together and turn all the knobs randomly and it'll sound better than any interface. If you don't know what individual hardware components to use or how to gain stage correctly to minimise background noise then yes odviously an interface will give you a better sound but that's not what I'm saying is there's much more potential for quality sound with individual components if you know what your doing.
This is the most helpful video on this topic that I've seen- I realize what I've been getting wrong in the hardware chain BEFORE the DAW. This is great content, thank you!
Gotcha, yeah if you can monitor through the software without latency that's another way to do that. I find that most people get to much latency so that isn't an option which is why they created the blend knob. Thanks for watching!
I totally understand where your coming from. You are correct. However the interface (that omega) you are using in this video is a dinosaur. Most interfaces today have good signed ASIO drivers and don't use a mix knob. I personally use a Focusrite 1820 2cnd Gen. What I really enjoyed about your video the most was your routing method of using ext pre amp, ext gear using the line inputs and not the mic pre amps. A lot of people seem to forget this or just don't realize it (unless they are pros like you and I :-) Keep up the good work.
At 8:20 you say a compressor is reducing overall volume. I was under the impression the compressor was reducing peaks in your signal based on the threshold. Noob here so wonder if you can clarify.
Hello. I have one question to this video. You insist in connecting the compressor to the audio interface so that you use a line jack in the audio interface and not an XLR jack. Why is this so important in your opinion?
And what´s the best way to record if you only have an audio interface. No hardware preamp/compressor etc I use JST Gain Reduction Plugin to record through. Is there a better way? Thanks
Dope Studio there man! It's funny how my ability to listen seemed to drop to zero instantaneously when your console first appeared on screen! I feel like I recognize those knobs lol... what kind of console is that exactly? ...Trident sounded pretty good, also, and every audio engineer no matter how advanced should keep reviewing the essentials; champions begin with the basics! :D
With all the gear you have im shocked to see you use this compressor. I own one but I would rather use my Fredenstein artistic comp or 600a. I never tried the 266 for vocals. I usually use it for bass in the mix.
Danny Cuevas So just to be clear that is XLR to TRS or a balanced quarter inch connection not phono. The reason that you need to then run back into the quarter inch connection is because that is a line input. Going into the XLR input is another microphone preamp which would just overdrive the signal
I understand that but I am looking at the video and I see an XLR input. Why not use XLR to XLR unless you just happen to have a balance XLR to TRS cable laying around. I would use an XLR balnced to XLR balance cable or have to purchase a specialized cable like the one you are using.
Because the XLR input on the interface is a microphone preamp connection, it is not a standard line input. The signal coming out of the DBX is at line level. The only line level inputs on the interface are quarter-inch. Now on the DBX unit, the XLR and quarter inch connections are all the same. So whether you use the XLR or quarter-inch, this is the same connection for either the inputs or the outputs.
Yeah I don’t know any other way to say this. You can use either the XLR or quarter inch output on your Dbx compressor but it has to go into the line input on your interface. If your interface has XLR line inputs then you can use those. If your interface has only XLR mic inputs then you are going to overdrive the mic preamp
How do I do this with a Focusrite Solo for HipHop vocals!? The only knob I should be using is the gain knob smh somebody help me. I’m using a rode nt1a mic
I will try to state this simply lol... Use a line out from your preamp or compressor or whatever you have, into the input on the front of the Solo (same one you use for the mic). Use the phantom power from the preamp not from the Solo. Turn the gain knob on solo all the way down (Left). Make sure you have the switch on the solo set to Line. Adjust the volume using the output gain knob on whatever the last item before your Solo is. After you use the meters on the analog device. Use the last output gain knob and the meters on your DAW to set desired level (i.e. -18db to -6db). If youre not using anything with the Solo simply plug an XLR cable from your mic direct to the input on the front of the Solo. Set the switch to MIC. Use your DAW's meters and the gain knob on the Solo to set levels. The Solo does not have meters on it. You will need to have the Solo's 48v Phantom power button on. Unless your mic came with its own power supply
Great Vid Weezna! I use Sonar Platinum as well. Can you tell me your process for chopping up samples then playing them on a midi device? I appreciate it!
Bro great video keep it up superb work!! But i got a question i have a rode nt2 connected to a dbx 286 do i send it to the mic line or line in on my forte and why ? Thanks in advance
I just watched a video from another mix engineer and he said to record vocals between -18 and -21. His reasoning, analog plugins have a sweet spot. In this video, you said to record between -10 and -7. I'm kinda confused. My setup is an Apollo Twin Solo. I use the 610 B Preamp and LA2A Unison Plugin. What level should I be at for hip-hop vocals?
-6 Daw line is perfect. Just kissing yellow on Daw. If you go any less, you might as well not use a compressor. You'll have to play catch up with plugins. You can always send to aux busses to hit plugins hard if you are looking for stank. The main benefit of compressor is to have consistent dynamics. Experiment and see what you like, but -6 works for me in Pro Tools. Btw, free saturation knob from softube rocks for hip hop vocals! Crank that in a aux send. Decapitator from sound toys if you want to pay and get something slightly better 👌
@@RCutOffakavEM Are you sure? I'm somewhat new to this, but I tried amplifying the sound in Audacity with a low volume input, and it introduces noise. When I match the volume level in to a more normalized level and digitally amplify less, there is less noise.
Why the hell would you have the preamp clip at 0? I tried this on mine and it distorts the audio? Horrible advice,,, I have mine to peak at -6 and it sounds perfect but peaking right around 0 is NOT a good idea, you are overloading the input
-6 is too quiet but 0 is too loud. You want the peaks at -1,5. Then you can adjust the output from the interface, so your audio peaks at -6. Then you can play around with the levels inside the DAW. You just want ro be able to record a loud signal with minimal noise. Loud meaning peaking at -6.
What about a preamp going into a compressor, then going into the Audiobox USB? The Audiobox doesn't have Line inputs. Is there a way to make it work? Right now I have the pre in the Audiobox with gain on the interface and phantom power off, seems to work fine. Yet to add the compressor in.
Great video. I have the same console, do you think the preamp of the mixer is better than the one on my Motu Ultralite or a focusrite Saffire ? What about comparing the GL vs your Lexicon preamp ?
This is too cool. I’ve had this pre amp for about 12 years and this is the first video I’ve seen someone demo the trident. Quick question. I’m running 2 compressors before the daw. La2a clone and 1176 clone. I notice these compressors give some gain itself to the vocal chain. Would it be wise to run the out of my last compressor back into the mic pre to make sure I’m getting it close to -7db before I run into interface?
& Most artists that send you files, do they have a decent Mic-Pre-Comp combo? Although... a simple Duet used these days has GREAT preamps.. but maybe perhaps inexperience, or lack of gear (or with gear, but lack of knowledge)..?
Nicholas Paris not so much the lack of gear most likely the acoustic treatment in the room or perhaps the lack of knowledge leading into the recording!
At timestamp 9:25 on your timeline you say concerning XLR connectors, "This is NOT what we want to use". I don't understand your reasoning, Please explain? Regardless, I did exactly as you recommended! Your video was extreemly helpful to me. I use some of the same gear you do. Thanks!
@@myrio1 If you have the luxury of having an interface with separate line inputs, use them. That way you aren't going from external preamp to compressor to interface preamp. If you use your line input then you get the AD conversion without going through a second preamp.
Wow! Instant sub man! Love your videos. Q: How do connect all three hardwares properly using my mic for vocal? 1. Dbx 286s Mic Preamp/Processor 2. ART Tube Mixer 3. Audient iD14 ( Interface ) 4. DAW ( Ableton Live 9 ) Any info would be appreciated!
So you would hook your mic up to the Dbx 286 mic input, then the Dbx 286 output to the Art mixer input, then the Art mixer output to the Audient input. Hope this helps
Hi, thanks for the video, you could tell me how you managed to link the omega lexicon card with the Sonar platinum program, I've been trying for several days and I will not be able to do it until now, thank you in advance for the answer
Hi Weezna. Great video. I noticed on the video that you come out from the compressor and go to the line in of the audio interface. Is the cable balanced or un-balanced. and why do you do it like this?
Great tutorial. A big question I'm having a difficult time finding an answer to is using a DI Box in the signal flow with outbound analog gear going into the interface. Should it go before the gear or placed after in front of the interface? Should you only use a DI Box for gtrs bass etc or also for vox? Thanks for your help!
The whole purpose of a di box (In my opinion) is mainly for guitar and bass so that you can essentially re-amp them and model your tone vs having to mic up a cabinet. 1. It should be directly after your instrument 2. it should go directly to your recording device Most Di boxes have 2 outs with 1 in. Hence you still can send a clean signal to your interface, but you can also use that other output of the DI box to go to your Amp.
i have an audiobox usb 96 an i notice i see no line input stated only mic/instrument so i’m assuming i can’t run an external preamp through it correct?
You can still run a pre amp into the interface. Mic into pre amp and out from pre amp to interface. The inputs on the 96 double for line inputs and Mic inputs.
This is no good. -6dbfs is way too high for a vocal. If the vocalist moves forward or increaseses their volume even slightly you are going to clip. You should be between -18 and -12 dbfs to ensure you have enough headroom
OK, I agree with this concept of going through pre amps and compressors to "control" your recordings peaks and transients but, in the digital world this is not necessary. Nor is it affordable for the average "bed room studio". I am an "old school" 70's era recording engineer and yes we used this setup to "print to tape". But, if we got it wrong the only way to fix it was "redo it"! In the digital world, "processing and editing" is "NON DISTRUCTABLE"! So why would you add, lets say compression to our "RAW TRACKS" that cannot be fixed later, because you recorded the "compression" now it's part of your track. "If you got it wrong, the only way to fix it is to redo it". As an old school recorder, the way I approach "TRACKING" in the digital world is, straight forward, keep it simple, no compression no EQ. no special routing no outboard equipment. Just your interface's pre amps (some are better than others) but, "use what you have".Lets say I am laying down a "BASS" track. I could plug my bass guitar into the HI z input of my interface and start recording and get "OK" results. (To get better results from plugging your HI z guitars into your interface find a HI z to Low z converter {DI BOX for example} you will get much better results) Think of your DAW at this point as just a "TAPE RECORDER'. It has recorded exactly what was sent into it. Now this "bass track" is what I would call your "virtual bass player", I can EQ it, Compress it, Effect it, do what ever I want to do to it until it sounds like the bass I wanted to hear, then I can "PRINT" it to another track as my bass track without changing the original "RAW TRACK" as I called it "VIRTUAL BASE PLAYER". I guess the message is "Do not process (MIX) during recording, Start processing and mixing after all tracking is complete and at a conservative level for ALL TRACKS (Around -10 to -15 DB). JUST SAYING everyone dose their own thing! '
Yeah man I get it, it is more equipment but these days I find people are willing to spend more money to get a more professional sound right off the bat. Yes if you are just recording for fun then it doesn’t make too much sense but a lot of these young artists are trying to start a career in music. And to your point about recording with a compressor or a eq to tape (daw), a majority of the studios that I know or work with are all doing this in just about every recording situation. Everyone does things differently and whatever way you like to do it it’s fine. There’s 1 million different ways to do the same thing
You can but just light compression when the signals coming in to get a more even signal and deal with the extreme peaks on the way in, then do the more heavy compression during the mixing stage with a separate Compressor plugin/plugins, I suggest watching this blokes video on how to set a hardware compressor and apply the same principles to a plugin Compressor in your daw
I've been subscribed for a few day now and I was wonder you seem to have a nice setup so why do you use Sonar instead of the industry standard Pro Tools or the fast and up and come Studio One
Hey, so it is in my opinion that all DAW's are fairly similar. I think what matters way more than software or gear is your ability to do what you intended on doing to whatever audio file. I actually started with ProTools over 14 years ago but I had issues with it. I then started using Sonar and never looked back. At this point I am so quick and efficient with Sonar that it makes zero sense to change what has been working so well for me. Not to mention, I like pushing out tracks that end up on TV, movies, and commercials that were done with Sonar, and in the end no one truly cares what I use, just as long as it sounds as expected.
Oh and another thing, I think it's really hard to get good at anything if you are always constantly changing gear and software. For a very long time, I thought the reason I wasn't getting the results I wanted was because of my gear. In the end I realized that it has nothing to do with the gear, and everything to do with experience and know how.
Ok thats great if you use a mic but what if you are solely reliant upon VST's and working within the DAW? How do you use the audio interface to make what you produce in the DAW sound good?
TrueSoundStudios so are you saying that if i record a sound from a vst (within the daw), that i do not have to adjust volumes of each channel, because it is already set properly? Is using an audio interface only for recording analog sound or does it have a key role in how your "DIGITAL" music sounds?
TrueSoundStudios im trying to figure out why my music doesn't sound clear and punchy on cellphones and car speakers like other artists music, idk if my export settings are wrong, or if im not mixing at the right volumes, or if im not mastering it properly... i use fl studio 11 and i produce edm...
I'm going to make myself the butt of about 1000 jokes here, but I don't see the point of using a mic preamp! I've been recording for 20 years, sometimes I use a preamp, sometimes I don't. I've done songs where I started without a preamp, and then halfway through switched to a $2000 Avalon preamp, and no one ever said they could hear the difference, or even noticed! I completely understand the need for compression, low cut, EQ etc., which can all be done in post. Most effects I do in post, so I don't need them. This preamp features a low filter he doesn't use. If I'm going to spend $1000 on a preamp, and mess with 3-4 extra cables and 10 knobs, I want it to be like: 'WOW! WHAT A DIFFERENCE!" or "Wow, that's A LITTLE better!" but I've simply never had that experience! 1 mic, 1 cable, go...WTF!? Am I crazy? Am I living in crazy land?
Hey so no i have had it for a while now. I bought it to use for my mobile recording rig to capture some better audio without making things to complex. So i bought this used for $400 but never would spend the $1000 to get it new. It sounds good and it is very clean with lots of headroom (there is no tubes in this preamp). In the studio i have used it a few times on a quiet acoustic guitar or a quiet vocalist but that's about it.
So I've always been told to have your signal hitting around -18 and peaks around -12 because unlike analog digital noise to floor ratio isn't as bad.So is -18 really to low?
Hey, so the problem I have been noticing is that the signal is recorded too low and it has a lot to do with it being recorded on not wonderfully quiet preamps and inexpensive microphones. To bring up the volume to get it to my desired decibel level, I am bringing up the volume but also all of the noise that is associated with that signal. Either way I have found better results by adding more signal when I record.
Hello! When I record, I also aim for -18db as Demetrice does. I have seen videos teaching the importance of adjusting the recording level around that because plugins are designed to sound the best that way. I actually have experimented and I did notice that sounds get crisper and fuller as far as frequency balances are concerned. (to my ears, at least) Here is a page that talks about it. vintageking.com/blog/2018/03/gain-staging/ An excerpt: "Most plug-in manufacturers have calibrated their units to operate with a nominal input level of -18dBFS, which has become standard practice for many engineers. However, some plug-ins (typically tape emulation units) are calibrated at -12dBFS, so it’s always best to reference the manual before using a new device." I wonder what the ideal way to go about it is. So far, using this -18db as a guildline for adjusting volume going into my DAW seems to be working really well☆
Question: I was wondering how you got the Omega to work with your pc? I have one but after windows xp I could never get the pc to recognize it. Is there some newer drivers?
I actually have never had an issue with this interface but to be honest I really only use this interface for these types of videos. I actually run SSL converters as my full time interface in my studio. I would definitely try installing the latest drivers, and possibly connecting to a different USB port when you do. Good luck
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Thankyou so very much for a precise, un-cluttered tutorial covering essential recording basics with hardware and also a wonderful introduction to tracking with compression. Was very good as well to understand why exactly we should shoot for -7 to -10 db peak when tracking. Removing signal to noise ratio and also providing good signal for plug-ins. I've often noticed hiss on some tracks when I've tracked -12 / -14 peak and did'nt understand exactly why. I followed your recommendations and my tracks have become a lot cleaner and easier to mix. Well done !
Great, man!
Being a rock guitarist, I 've always had a "crank' em all" kind of approach when performing live, but recording (especially in the digital domain) is a whole different thing.
Simple and clear video for everyone to understand! 👏👏👏
Cheers!
It’s like baking bread. Getting the right amount of water to flour ratio and other seasonings to get it good and right.
Very helpful information! I would, however, like to put in my two cents and say that instead of having the gain meter of the preamp peak at 0dB, I think it should peak rather at -3dB. That way, this will ensure that actual clipping from whatever transducer you are recording from the preamp into your DAW doesn’t occur.
THANK YOU !
FINALYY, someone that GETS IT !
There is so much misinformation, disinformation and just plain WRONG information about this VERY important subject.
I have ALWAYS recorded as hot as possible WITHOUT clipping, and yes I record vocals WITH compression.
I typically record with the compressor at -10 dB threshold, and 4 to 1 compression.
I aim for -3 dB max.
What's with these 'engineers' that record at -18 dBs ?
This seems to be the latest trend.
PLEASE !
Thanks again for an excellent presentation !
Bill P.
Just clear excellent piece of presentation. Guy knows what he is talking about. 👍
I've had trouble understanding the signal process and how everything is connected to each other, and how to set levels and which levels to set, etc., etc. This very clear instructional tutorial was invaluable to me. It's well explained which gave me a way better understanding of how to go about recording. So invaluable. I liked and subscribed because I'm sure I'll be able to learn more from you. Well done!
This and room treatment is very overlooked!
Man I was doing it all wrong lol. I was recording with a very low signal and my daw track turned way down then after recording i would crank it back up duplicate the vocals to another track add effects and mix them together to make them sound loud and full. Rebuilding my studio now and am relearning from the basics
Gain staging is very much so overlooked! Thank you for this nice video and all the great info you provided 💯💯
It has to be said that compressing a signal BEFORE recording it should only be attempted if you really know what you are doing and what kind of sound you are after. If not, this technique will needlessly reduce the dynamic range and degrade the signal/pull up noise. In most instances, it would be best practice to get the signal into the daw and then do compression later, be it in the DAW or via outboard. That way, everything is nondestructive and therefor reversable. There can be instances with unexperienced musicians in combination with unforgiving ADC, where you will need that kind of compression, to allow for a signal that won't clip at the ADC. But as long as you got a good source, a good mic a good preamp and a good converter, leave that compressor out of there if at all possible.
That's why you run a dry line as well no?
@@forkmusic If you have a setup that allows this, then I certainly would advice it. But the more common workflow would be to record dry and then send channels through plugins or outboard gear afterwards.
This video was extremely helpful, thank you so much I just got an outboard compressor and preamp. This video really helped me understand them and get the best settings possible
Isai Saint Absolutely, glad I could help
H i, just found your channel ,thank god for someone who has the ability to explain things clearly and at a pace that works.It helps us out here that want to learn .Thanks for passing on your knowledge.
Very useful. Many thanks for doing this. My dbx 266XL seems do be obsolete now, but always appreciate guidance on getting the best from it.
Clearest explanation ever, thank you!
you are so cool, i really appreciate your studio making series, i was researching this topic for many many hours and i guarantee you that your videos were the best. Im making my own home studio now and i will use your experience properly :p
I like your videos as you keep things straightforward. The fact you use Sonar is refreshing too.
Aaron Lothian Yasss!!!!
Thank you, I really try to only talk about what I think is truly important! Thanks for watching
Aaron Lothian what about a channel strip
derrick chapman what about it. The levels in an analogue channel are different from the digital equivalent. Of at least reference levels are.
An excellent tutorial! Very helpful. Thank you sir.
Excellent video! Cleared lot of questions floating in my head. I have a home studio that I've been tinkering around with for a couple years and first started off using a Zoom R8 standalone. Eventually I upgraded and got Reaper DAW, and now use the R8 as an audio interface via USB. Works pretty good, but I have noticed that my sound quality playback (particularly acoustic guitar) doesn't have the that realistic crispness and fullness that I hear in the headphones when recording. I tried a comparison with a friend that uses a preamp with his Focusrite AI. It was a noticeably better sound. At first, I thought it might be the R8, but the more I research I think I need a preamp with the proper gain staging. I may invest in a preamp and see what happens.
Very good instructional video of this subject. Many thanks!
As other videos just show input levels adjusted around -6dB to avoid clipping, the idea of initial level gain at 0 seems a bit odd. Beyond that, there is still one basic mic level that either is or is not at an optimal level, and running one unit into the next and the next and the next just looks like a great way to introduce noise, with no apparent intrinsic benefit. The concept of clean signal in, to DAW, then apply edits and adjustments to desired output just looks a lot more sensible and far less expensive.
Was thinking the same about the input level being 0 dB doest that just mean it's gonna clip? As for the hardware gear he has, audio interfaces have these components in them already (pre amps, eqs, adda Converters), they're just all smashed into one little box at an affordable price which means they arnt very good quality, buying quality hardware components sepratly will give you a much better quality signal than any audio interface as long as the right gear is used for whatever your recording and it's used correctly
@@indyfreeman7172 The concept of assembling a series of separate, "superior quality" components is not always valid. One sub-par component brings the rest down, and a degraded signal can't be improved by the rest. Each added part may introduce noise, hum, and require balancing levels between every separate unit - a recipe for problems. Integrated systems that have solid reviews from tech experts are likely to be far easier to use, and the built-in filters, amps, equalizers, converters, etc are designed for compatibility and streamlined control. Any slight loss in one area is probably not going to be that big a deal, in the end. Unless one enjoys the yards of tangled cables, and hours spent balancing and troubleshooting, most folks will be better off practicing and playing.
@@z1522 yea that's true it can cause problems that's why i said it's only the case if you use the correct gear and know how to use it. I'm not saying you can just chuck any pre amp and and adda converter together and turn all the knobs randomly and it'll sound better than any interface. If you don't know what individual hardware components to use or how to gain stage correctly to minimise background noise then yes odviously an interface will give you a better sound but that's not what I'm saying is there's much more potential for quality sound with individual components if you know what your doing.
Excellent video. I feel like I just took a mini crash-course in studio engineering!
This is the most helpful video on this topic that I've seen- I realize what I've been getting wrong in the hardware chain BEFORE the DAW. This is great content, thank you!
Fantastic outboard tracking explanation!
Another video said -6. Also look for the loudest part of a performance and adjust from there.
Unless you have a transformerless board like the SSL series. Great video.
Nice explanation. what about Expander-Gate?
Truly... An amazing explanation... Thank you 💯
Thanks for this video man, very helpful. You're a great educator.
Thank you, i really appreciate that!
Well explained Weezna, I've currently been a fan of your channel buddy. Great stuff man I learn a lot in my voice over skills
I never use a monitor mix knob. I use the faders within Sonar to set my monitor mixes. if the vocals are too loud on the Vox, pull the fader down.
Gotcha, yeah if you can monitor through the software without latency that's another way to do that. I find that most people get to much latency so that isn't an option which is why they created the blend knob. Thanks for watching!
I totally understand where your coming from. You are correct. However the interface (that omega) you are using in this video is a dinosaur. Most interfaces today have good signed ASIO drivers and don't use a mix knob. I personally use a Focusrite 1820 2cnd Gen. What I really enjoyed about your video the most was your routing method of using ext pre amp, ext gear using the line inputs and not the mic pre amps. A lot of people seem to forget this or just don't realize it (unless they are pros like you and I :-) Keep up the good work.
Excellent video
This video is extremely helpful, thank you.
At 8:20 you say a compressor is reducing overall volume. I was under the impression the compressor was reducing peaks in your signal based on the threshold. Noob here so wonder if you can clarify.
how simple video with simlpe words can change Your life ??? especial that You are begginer??? BRILLIANT TUTORIAL 10 STARS fof Your work
This was an awesome video very helpful and very informative
First interphase was that Lexicon! Was a really nice! Can not believe the price on that!?
Fantastic video
Hello. I have one question to this video. You insist in connecting the compressor to the audio interface so that you use a line jack in the audio interface and not an XLR jack. Why is this so important in your opinion?
Great tutorial man make my day bless up
Nicely explained. Thanks!
Great Info, thank you
And what´s the best way to record if you only have an audio interface. No hardware preamp/compressor etc
I use JST Gain Reduction Plugin to record through. Is there a better way?
Thanks
My lexicon Omega sound card does not give signal on Daw, how can I fix it
Thank you brother! following and subscribing right now
Dope Studio there man! It's funny how my ability to listen seemed to drop to zero instantaneously when your console first appeared on screen! I feel like I recognize those knobs lol... what kind of console is that exactly?
...Trident sounded pretty good, also, and every audio engineer no matter how advanced should keep reviewing the essentials; champions begin with the basics! :D
Beautiful explanation!!
With all the gear you have im shocked to see you use this compressor. I own one but I would rather use my Fredenstein artistic comp or 600a. I never tried the 266 for vocals. I usually use it for bass in the mix.
Many thanks for clarifying.
Thank you very much.. that was really helpful.
Super helpful- thank you.
Thanks for the most simplest but the best video !
If I add an a/d or d/a conveter, does it come after the interface or before the interface?
Thanks !
Dont hurt your interface using the gain of the preamp and the interface at the same time?
Thanks for the tips!!!
My very first question is, why did you leave the DBX with a xlr to phono cable instead of XLR to XLR? Answer me that one please?
Danny Cuevas So just to be clear that is XLR to TRS or a balanced quarter inch connection not phono. The reason that you need to then run back into the quarter inch connection is because that is a line input. Going into the XLR input is another microphone preamp which would just overdrive the signal
I understand that but I am looking at the video and I see an XLR input. Why not use XLR to XLR unless you just happen to have a balance XLR to TRS cable laying around. I would use an XLR balnced to XLR balance cable or have to purchase a specialized cable like the one you are using.
Because the XLR input on the interface is a microphone preamp connection, it is not a standard line input. The signal coming out of the DBX is at line level. The only line level inputs on the interface are quarter-inch. Now on the DBX unit, the XLR and quarter inch connections are all the same. So whether you use the XLR or quarter-inch, this is the same connection for either the inputs or the outputs.
I have a DBX 266 =XS like yours and it has both XLR and 1/4" Inputs and Outputs. Am I missing something?
Yeah I don’t know any other way to say this. You can use either the XLR or quarter inch output on your Dbx compressor but it has to go into the line input on your interface. If your interface has XLR line inputs then you can use those. If your interface has only XLR mic inputs then you are going to overdrive the mic preamp
How do I do this with a Focusrite Solo for HipHop vocals!? The only knob I should be using is the gain knob smh somebody help me. I’m using a rode nt1a mic
I will try to state this simply lol...
Use a line out from your preamp or compressor or whatever you have, into the input on the front of the Solo (same one you use for the mic). Use the phantom power from the preamp not from the Solo. Turn the gain knob on solo all the way down (Left). Make sure you have the switch on the solo set to Line. Adjust the volume using the output gain knob on whatever the last item before your Solo is. After you use the meters on the analog device. Use the last output gain knob and the meters on your DAW to set desired level (i.e. -18db to -6db).
If youre not using anything with the Solo simply plug an XLR cable from your mic direct to the input on the front of the Solo. Set the switch to MIC. Use your DAW's meters and the gain knob on the Solo to set levels. The Solo does not have meters on it. You will need to have the Solo's 48v Phantom power button on. Unless your mic came with its own power supply
thank you, this was very helpful
Great Vid Weezna! I use Sonar Platinum as well. Can you tell me your process for chopping up samples then playing them on a midi device? I appreciate it!
Thank you, so yes i will be releasing that video in the next few days! Hopefully it's what your looking for? Thanks for watching!
Bro great video keep it up superb work!! But i got a question i have a rode nt2 connected to a dbx 286 do i send it to the mic line or line in on my forte and why ? Thanks in advance
I just watched a video from another mix engineer and he said to record vocals between -18 and -21. His reasoning, analog plugins have a sweet spot. In this video, you said to record between -10 and -7. I'm kinda confused. My setup is an Apollo Twin Solo. I use the 610 B Preamp and LA2A Unison Plugin. What level should I be at for hip-hop vocals?
In my opinion, stick to the -21 - -18dbfs. More headroom - the better! You can always turn it up later, no need to push it too hard
@@RCutOffakavEM Won't turning the digital audio up introduce noise?
@@GhostSamaritan No, noise introduction. If you turn up your preamp, then yes, but never in the mix phase, when you push the faders
-6 Daw line is perfect. Just kissing yellow on Daw. If you go any less, you might as well not use a compressor. You'll have to play catch up with plugins. You can always send to aux busses to hit plugins hard if you are looking for stank. The main benefit of compressor is to have consistent dynamics. Experiment and see what you like, but -6 works for me in Pro Tools. Btw, free saturation knob from softube rocks for hip hop vocals! Crank that in a aux send. Decapitator from sound toys if you want to pay and get something slightly better 👌
@@RCutOffakavEM Are you sure? I'm somewhat new to this, but I tried amplifying the sound in Audacity with a low volume input, and it introduces noise. When I match the volume level in to a more normalized level and digitally amplify less, there is less noise.
Why the hell would you have the preamp clip at 0? I tried this on mine and it distorts the audio? Horrible advice,,, I have mine to peak at -6 and it sounds perfect but peaking right around 0 is NOT a good idea, you are overloading the input
I think you haven't watched the same video I did
-6 is too quiet but 0 is too loud. You want the peaks at -1,5. Then you can adjust the output from the interface, so your audio peaks at -6. Then you can play around with the levels inside the DAW. You just want ro be able to record a loud signal with minimal noise. Loud meaning peaking at -6.
AFFE NOWU I don’t think you guys have watched the same video I did... 0dB *analog* equals about -18dB *digital*
What about a preamp going into a compressor, then going into the Audiobox USB? The Audiobox doesn't have Line inputs. Is there a way to make it work?
Right now I have the pre in the Audiobox with gain on the interface and phantom power off, seems to work fine. Yet to add the compressor in.
Hi, thanks for sharing. Will this unit work with a zoom multi track recorder?
Great video. I have the same console, do you think the preamp of the mixer is better than the one on my Motu Ultralite or a focusrite Saffire ? What about comparing the GL vs your Lexicon preamp ?
This is too cool. I’ve had this pre amp for about 12 years and this is the first video I’ve seen someone demo the trident. Quick question. I’m running 2 compressors before the daw. La2a clone and 1176 clone. I notice these compressors give some gain itself to the vocal chain. Would it be wise to run the out of my last compressor back into the mic pre to make sure I’m getting it close to -7db before I run into interface?
& Most artists that send you files, do they have a decent Mic-Pre-Comp combo? Although... a simple Duet used these days has GREAT preamps.. but maybe perhaps inexperience, or lack of gear (or with gear, but lack of knowledge)..?
Nicholas Paris not so much the lack of gear most likely the acoustic treatment in the room or perhaps the lack of knowledge leading into the recording!
Thanks for this
At timestamp 9:25 on your timeline you say concerning XLR connectors, "This is NOT what we want to use". I don't understand your reasoning, Please explain? Regardless, I did exactly as you recommended! Your video was extreemly helpful to me. I use some of the same gear you do. Thanks!
This is the main thing i dont understand. When you say "do not" use the mic 1 & 2 channels on you Omega?
@@myrio1 If you have the luxury of having an interface with separate line inputs, use them. That way you aren't going from external preamp to compressor to interface preamp. If you use your line input then you get the AD conversion without going through a second preamp.
so helpful thank you
Wow! Instant sub man! Love your videos.
Q: How do connect all three hardwares properly using my mic for vocal?
1. Dbx 286s Mic Preamp/Processor
2. ART Tube Mixer
3. Audient iD14 ( Interface )
4. DAW ( Ableton Live 9 )
Any info would be appreciated!
So you would hook your mic up to the Dbx 286 mic input, then the Dbx 286 output to the Art mixer input, then the Art mixer output to the Audient input. Hope this helps
Hi, thanks for the video, you could tell me how you managed to link the omega lexicon card with the Sonar platinum program, I've been trying for several days and I will not be able to do it until now, thank you in advance for the answer
Hi Weezna. Great video. I noticed on the video that you come out from the compressor and go to the line in of the audio interface. Is the cable balanced or un-balanced. and why do you do it like this?
balanced!
Hi there, good informative vidéo. One question; are the Lines input in your sound card balance or unbalance? Thanks Jeff
They should always be TRS balanced. XLR or TRS balanced connecters are same
damnnn good video my mann! keep it up
Great tutorial. A big question I'm having a difficult time finding an answer to is using a DI Box in the signal flow with outbound analog gear going into the interface. Should it go before the gear or placed after in front of the interface? Should you only use a DI Box for gtrs bass etc or also for vox? Thanks for your help!
The whole purpose of a di box (In my opinion) is mainly for guitar and bass so that you can essentially re-amp them and model your tone vs having to mic up a cabinet.
1. It should be directly after your instrument
2. it should go directly to your recording device
Most Di boxes have 2 outs with 1 in.
Hence you still can send a clean signal to your interface, but you can also use that other output of the DI box to go to your Amp.
Nice Vid bro Subscribed Fam
Can I bypass the battery on a JBL EON PRO PORTABLE? I can’t seem to find another battery.
Not sure, i'm sure it's possible but it depends on a lot of things
TrueSoundStudios Thanks for responding.
i have an audiobox usb 96 an i notice i see no line input stated only mic/instrument so i’m assuming i can’t run an external preamp through it correct?
You can still run a pre amp into the interface.
Mic into pre amp and out from pre amp to interface. The inputs on the 96 double for line inputs and Mic inputs.
excellent
This is no good. -6dbfs is way too high for a vocal. If the vocalist moves forward or increaseses their volume even slightly you are going to clip. You should be between -18 and -12 dbfs to ensure you have enough headroom
OK, I agree with this concept of going through pre amps and compressors to "control" your recordings peaks and transients but, in the digital world this is not necessary. Nor is it affordable for the average "bed room studio". I am an "old school" 70's era recording engineer and yes we used this setup to "print to tape". But, if we got it wrong the only way to fix it was "redo it"! In the digital world, "processing and editing" is "NON DISTRUCTABLE"! So why would you add, lets say compression to our "RAW TRACKS" that cannot be fixed later, because you recorded the "compression" now it's part of your track. "If you got it wrong, the only way to fix it is to redo it". As an old school recorder, the way I approach "TRACKING" in the digital world is, straight forward, keep it simple, no compression no EQ. no special routing no outboard equipment. Just your interface's pre amps (some are better than others) but, "use what you have".Lets say I am laying down a "BASS" track. I could plug my bass guitar into the HI z input of my interface and start recording and get "OK" results. (To get better results from plugging your HI z guitars into your interface find a HI z to Low z converter {DI BOX for example} you will get much better results) Think of your DAW at this point as just a "TAPE RECORDER'. It has recorded exactly what was sent into it. Now this "bass track" is what I would call your "virtual bass player", I can EQ it, Compress it, Effect it, do what ever I want to do to it until it sounds like the bass I wanted to hear, then I can "PRINT" it to another track as my bass track without changing the original "RAW TRACK" as I called it "VIRTUAL BASE PLAYER". I guess the message is "Do not process (MIX) during recording, Start processing and mixing after all tracking is complete and at a conservative level for ALL TRACKS (Around -10 to -15 DB). JUST SAYING everyone dose their own thing! '
Yeah man I get it, it is more equipment but these days I find people are willing to spend more money to get a more professional sound right off the bat. Yes if you are just recording for fun then it doesn’t make too much sense but a lot of these young artists are trying to start a career in music. And to your point about recording with a compressor or a eq to tape (daw), a majority of the studios that I know or work with are all doing this in just about every recording situation. Everyone does things differently and whatever way you like to do it it’s fine. There’s 1 million different ways to do the same thing
Do I need to shut down the phantom power on my interface if I have the phantom power turned on on the microphone preamp?
Dom Shatti yes if phantom power is already being applied by your preamp you do not need the phantom power on your interface🎹🎧🚀🙏🏻
" and im check aye " Lmao
What is the track in the background?
great video :)
I find that some vocals are ariund -20 and i know the reason why...artist are recording their vocals really quiet😅
I see you record with a physical compresor unit, If i only have an interface should i record with compresor on?
You can but just light compression when the signals coming in to get a more even signal and deal with the extreme peaks on the way in, then do the more heavy compression during the mixing stage with a separate Compressor plugin/plugins, I suggest watching this blokes video on how to set a hardware compressor and apply the same principles to a plugin Compressor in your daw
I've been subscribed for a few day now and I was wonder you seem to have a nice setup so why do you use Sonar instead of the industry standard Pro Tools or the fast and up and come Studio One
Hey, so it is in my opinion that all DAW's are fairly similar. I think what matters way more than software or gear is your ability to do what you intended on doing to whatever audio file. I actually started with ProTools over 14 years ago but I had issues with it. I then started using Sonar and never looked back. At this point I am so quick and efficient with Sonar that it makes zero sense to change what has been working so well for me. Not to mention, I like pushing out tracks that end up on TV, movies, and commercials that were done with Sonar, and in the end no one truly cares what I use, just as long as it sounds as expected.
Oh and another thing, I think it's really hard to get good at anything if you are always constantly changing gear and software. For a very long time, I thought the reason I wasn't getting the results I wanted was because of my gear. In the end I realized that it has nothing to do with the gear, and everything to do with experience and know how.
Do you have to adjust the levels within the DAW?
Excelent!
awesome man!!!
Thank you!
Barf
what if you dont have a dbx????????
Ok thats great if you use a mic but what if you are solely reliant upon VST's and working within the DAW? How do you use the audio interface to make what you produce in the DAW sound good?
This is only when recording analog signals. If it's already in the DAW then the gain staging has already been determined.
TrueSoundStudios so are you saying that if i record a sound from a vst (within the daw), that i do not have to adjust volumes of each channel, because it is already set properly? Is using an audio interface only for recording analog sound or does it have a key role in how your "DIGITAL" music sounds?
TrueSoundStudios im trying to figure out why my music doesn't sound clear and punchy on cellphones and car speakers like other artists music, idk if my export settings are wrong, or if im not mixing at the right volumes, or if im not mastering it properly... i use fl studio 11 and i produce edm...
I'm going to make myself the butt of about 1000 jokes here, but I don't see the point of using a mic preamp! I've been recording for 20 years, sometimes I use a preamp, sometimes I don't. I've done songs where I started without a preamp, and then halfway through switched to a $2000 Avalon preamp, and no one ever said they could hear the difference, or even noticed! I completely understand the need for compression, low cut, EQ etc., which can all be done in post. Most effects I do in post, so I don't need them. This preamp features a low filter he doesn't use. If I'm going to spend $1000 on a preamp, and mess with 3-4 extra cables and 10 knobs, I want it to be like: 'WOW! WHAT A DIFFERENCE!" or "Wow, that's A LITTLE better!" but I've simply never had that experience! 1 mic, 1 cable, go...WTF!? Am I crazy? Am I living in crazy land?
did you recently get that trident preamp? Ive been looking at the same one! how do you like it? what do you like it on?
Hey so no i have had it for a while now. I bought it to use for my mobile recording rig to capture some better audio without making things to complex. So i bought this used for $400 but never would spend the $1000 to get it new. It sounds good and it is very clean with lots of headroom (there is no tubes in this preamp). In the studio i have used it a few times on a quiet acoustic guitar or a quiet vocalist but that's about it.
So I've always been told to have your signal hitting around -18 and peaks around -12 because unlike analog digital noise to floor ratio isn't as bad.So is -18 really to low?
Hey, so the problem I have been noticing is that the signal is recorded too low and it has a lot to do with it being recorded on not wonderfully quiet preamps and inexpensive microphones. To bring up the volume to get it to my desired decibel level, I am bringing up the volume but also all of the noise that is associated with that signal. Either way I have found better results by adding more signal when I record.
Hello!
When I record, I also aim for -18db as Demetrice does. I have seen videos teaching the importance of adjusting the recording level around that because plugins are designed to sound the best that way.
I actually have experimented and I did notice that sounds get crisper and fuller as far as frequency balances are concerned. (to my ears, at least)
Here is a page that talks about it.
vintageking.com/blog/2018/03/gain-staging/
An excerpt:
"Most plug-in manufacturers have calibrated their units to operate with a nominal input level of -18dBFS, which has become standard practice for many engineers. However, some plug-ins (typically tape emulation units) are calibrated at -12dBFS, so it’s always best to reference the manual before using a new device."
I wonder what the ideal way to go about it is.
So far, using this -18db as a guildline for adjusting volume going into my DAW seems to be working really well☆
@@Satnamtaro i would say your right about this as well, from a professional audio engineer experience. There's a reason why dbFS was introduced.
Calibrate the recording chain
FINALLY
How can I get in tune with you for mixing services
derrick chapman All the info is in the description of this video
Hey guys whats up, Im WHEEZNUT (good vid btw)
Oshri Zohara nope, not even close
Question: I was wondering how you got the Omega to work with your pc? I have one but after windows xp I could never get the pc to recognize it. Is there some newer drivers?
I actually have never had an issue with this interface but to be honest I really only use this interface for these types of videos. I actually run SSL converters as my full time interface in my studio. I would definitely try installing the latest drivers, and possibly connecting to a different USB port when you do. Good luck
you have to update the drivers i had to do it with windows 7 now it finds omega i just need to know how to set up lexicon to record vocals
Use ASIO4ALL