[QUOTE=dave mustaine #2;18451847]Distressors are amazing. Best compressor I've used yet. The Fatso is a really good unit as well.

In terms of software compressors there are some really good ones as well. There is a really good replication of the 1176, the SSL bus comp is really good. I always use that on my master channel and drum bus. The SSL E channel strip is a must imo. It is the best EQ I've used, good compressor, good gate, and if you push it to just below it's limit you can get some great sounds.

Has anyone ever used a device called a "Finalizer"? It's a piece of outboard gear that does sort of an "auto master" job. It can get some good sounds for people that are either on a limited budget or don't have the first clue how to master.

I remember my first attempt at mastering all I did was put an L2 limiter, an SSL comp and a c4 (multi eq) comp on the master channel and called it a day lol. Wasn't the best job that's for sure.[/QUOTE]

check out steve slates mastery plugin, it's 350 and it gets it close enough for demo work...

Does it have a free trial? I usually try out plug ins before i get em. I just downloaded a free trial of drumagog and I really want it now but can't afford it haha.

[quote=dave mustaine #2;18451862]An interesting tid bit of information. Did you guys know that George Massenberg uses no analog gear whatsoever? Nothing, everything is digital. I think that's kinda strange.[/quote]


Where did you hear that? It's especially weird, seeing as how he has an entire company dedicated to building analog equipment.

I read that in the interview he did with Howard Massey in "Behind the Glass". The whole book is great I highly recommend it.

I also believe the company he has builds digital gear that tries to replicate analog sounds. But I'm not entirely sure on that so don't quote me.

[url]http://www.massenburg.com/[/url]

He does do a lot of work in the field of A/D/A conversion, but the products are pretty much all-analog all the way.

hmm. strange for him not to use it then...

My personal opinion is that for a person of his caliber and experience to completely ignore analog would be foolish.

Some of my favourite gear is analog. The great river pre, anything made by API is great. The 550 is a really great eq.

I do know some people that will only mix in the box and use no analog outboard gear.

The man of the future!

Sorry for not replying guys didnt expect such a response!

Im using Cubase 5 as my DAW, Superior Drummer 2.0 (mapping and then importing them into my project on multichannel) and a Line 6 UX2 for guitars with Pod Farm and ReCabinet.

Iv always been taught in Music Tech classes when i was at school that you want everything peaking at as close to 0 as you can without it clipping. Is this what people talk about pre fader and post fader for?

Would you guys say to get the levels for my drums with some space between the orange and red then?
And i wanted to bus them all into one track when its done so i can control the overall drum levels when i have all the guitar tracks recorded.

At the minute im pushing the levels in SD up so i can get as high an output as possible into Cubase, and then go from there.

pre fader is what you are thinking of. You should have your pre amps running somewhat hot (in the yellow not orange) to get the best sounds but your faders should be in the green.

You don't have to comp your drums to one track, all you have to do is make a drum bus. Create an aux track and bus all the drums to the aux track. A good technique to tighten up the drums is to put a compressor on the drum bus as well as any seperate drums that need it. You can also add a limiter to the bus with a limiter, but don't go to far with the limiter, no more then -5 or -6 dB of limiting, and don't go too far with your compression on the bus either, no more then 6 or 7 dB and no more then a 3:1 or 4:1 ratio (usually).

Hope this helps.

Thanks for the advice mate. I imagine this should be done in the mastering stage once all the other instruments are recorded?

John Scripp (of Massive Mastering) actually wrote an interesting blog post about "proper" recording levels:

[url]http://www.massivemastering.com/blog/index_files/Proper_Audio_Recording_Levels.php[/url]

It basically boils down to finding the best-sounding spot on your preamp, and ignoring what the digital meters are saying on the way in (provided you aren't clipping). I reckon there's something to that, since it's something that has been supported quite a bit by the stuff I've heard/read. At minimum, I know Nika Aldritch talked about it in an AES presentation I was at (he might have even touched on it in his book, but it's been long enough since I read it that I can't recall off hand).

I tried out this practice on my last big session (I normally did the "loudest before clipping" approach routinely). I don't know that it had a huge difference, but it certainly didn't [I]hurt[/I]. In the digital realm, it's essentially trivial to simply change the amplitude levels if you captured too quietly.

That article is quite interesting. Kinda goes against everything iv ever been told but he puts across a good argument.

Is the same principle applicable for guitars?
Im recording guitars using a UX2, so for example, if i record a dry signal into Cubase were ideally would the guitars be peaking on the input phase and the output phase?
As it stands i can either get really weak signals into Cubase that sound pathetic, or over-saturated signals that clip unless the faders are pulled down to rock bottom.
Reading through the manuals, Guitars should be always 'in the green' on the input phase, but i dont have any idea about output.

Compressors and limiters and the like can be applied during the mix and actually I highly recommend it. Mastering usually takes all the songs on an album and puts them through compressor (among other things) to make all tracks consistent.

It's almost become a detriment to music in a way. Dynamics are almost completely gone from music now. Take a waveform from the 70's and compare it to one from today. Anything from the past decade will look like a brick.

Im recording guitars using a UX2.
So for example, if i record a dry signal into Cubase where ideally would the guitars be peaking on the input phase and the output phase (within Pod Farm)?
As it stands i can either get really weak signals into Cubase that sound pathetic, or over-saturated signals that clip unless the faders are pulled down to rock bottom.
Reading through the manuals, Guitars should be always 'in the green' on the input phase, but i dont have any idea about output.

If you're asking about the best "input level", I think the best solution is to do some tests yourself on this one.

Record yourself playing something simple a few different times, with each take using a different input gain. Then use your faders so that each take plays back at the same volume. Then solo each one up and choose the one that sounds best to you.

As per Scripp's blog post, you can also use a Y-splitter or doubling pedal (or similar) to capture the same take on 2 different inputs (of the same type) at different input gains. This would be the closest thing to a "scientific" approach you can do.

[quote=Kuffuffled;18451405]+4dBu is simply an operating level. It doesn't have anything to do with loudness does it?[/quote]
Yeah and it's not where it clips either. It as you say is just an operating level.

Where it clips is different from hardware to hardware, RTFM.


Xomblies: I manned up and bought a Rosetta 200, if you have any finished/rough mixes I can smash it through that if you wanna do some experiments. That thing has soft limiting and [I]The[/I] [I]Aptomizer [/I](say it like the opening to an epic cheesy action movie). So i'm pretty sure I could make it look like a perfect fucking block if I tried.

I think i may be over-complicating it for myself.

Im running some EMG's into the norm on my UX2 not the Pad input. So when im getting the levels on podfarm it clips on my rec/send fader, but not the overall ouput gain stage thing, and it isnt clipping within Cubase, it peaks at around 1.19 on the channel level.

Im trying my hardest not to sound like a massive noob haha.

[quote=dave mustaine #2;18452645]pre fader is what you are thinking of. You should have your pre amps running somewhat hot (in the yellow not orange) to get the best sounds but your faders should be in the green.
[/quote]
The article kind of beat me to it, but there are standards, but yes use your ear and find where the preamp sounds best on that source in that room with that player. 9/10 something you did last session might not work this session. Best way is to acquire an ear for when analog gear is operating at it's sonically optimal level. This usually means a bit of saturation in tracking, but with 24bit digital, you're always going to clip the preamp before you clip the A/D's.

Even better get a preamp with an output trim and never have to worry about clipping your converters again when you're pushing the input stage!


Also: Don't attempt to drive shitty USB interfaces preamps, it will not sound good.

I dont want to drive it too much, i just want to get a good enough signal going in to work with.

[quote=benfan;18456137]I think i may be over-complicating it for myself.

Im running some EMG's into the norm on my UX2 not the Pad input. So when im getting the levels on podfarm it clips on my rec/send fader, but not the overall ouput gain stage thing, and it isnt clipping within Cubase, it peaks at around 1.19 on the channel level.

Im trying my hardest not to sound like a massive noob haha.[/quote]
I think you are.

lets solve this...


You should have some sort of meter or clip light on your interface? If that ain't clipping, you aren't clipping your pres and most likely not the converters.

When using an interface like this you want to not push it very hard because it'll probably sound harsh. The electronics in this caliber of equipment is very good so you don't want to push your luck with going anywhere near clipping.

So do this. Take all plug ins off the channel you're tracking to. Then set the fader to zero. Play whatever part you want to play. If it's chugs just constantly chug and adjust the gain on the UX2 till the channel is reading roughly -18dbFS. This is usually where plug ins are designed to operate at and is a good start point.

Then chuck the amp sim on. Now try get some tones going. Once you've got a half decent tone going I want you to adjust the output of the plug in match the overall peak level you had before (-18dbFS). Bypass Pod Farm in and out whilst chugging and use your ears (and eyes on the meters) to make sure there's no difference in volume change from your DI and your amp sim. Now distorted tones are going to have a higher RMS or average level so will appear louder but you should be able to do it by just the meters.

DO THIS EVERY TIME YOU RECORD A NEW TRACK.

It will allow you to have enough digital headroom so shit don't get nasty and enough headroom on the master so when you quad track with two different amps your overall mix won't be over 0dbFS before you even start mixing.

This applies to almost anything in audio. Make sure the levels in are the same when you're going out (if you can). It's just gain staging and makes sure you can make rational decisions about your sounds, because if it's louder when it comes out of the compressor, you might just think it sounds better because of that. Matching levels is key to really knowing what's going on before and after.

Quick analogy: Before and after shots on commercials where in the first picture she isn't smiling, is wearing trackpants and hasn't brushed her hair in a week, then the after photo she's had make up put on, the lighting is better and overall is produced to fool you into thinking something is better.

It's the same with this, make sure the first picture is as close to the second one or you can't make any rational judgments.

Thanks for the help. Im having a play around now.

Basically what your saying is that the output of my amp sim into my DAW should be the same as my DI going in?

If im wanting to group all my Drums/Cymbals, Room Mics/OH or Rhythm or Melody guitars together for compressing them or adding reverb and stuff. Do i have to create an FX Channel and send them to that?

Im using Cubase 5.

why are you using cubase in stead of reaper

reaper is for broke ass niggas trying to play me too



























[IMG]http://static.tumblr.com/jdoh5hu/OS2ldtvnc/trollface_small.jpg[/IMG]

Haven't used Cubase in a long time so I'm not sure.

If you are able to make an aux track (i'm assuming cubase's equivilant would be an FX track) Then you just assign the output of each channel so that it buses to the Aux/FX track.

Then whatever plugs you put on the aux/fx track will effect anything being bused to the track.

[quote=JoshIsNumber3;18462127]reaper is for broke ass niggas trying to play me too



























[IMG]http://static.tumblr.com/jdoh5hu/OS2ldtvnc/trollface_small.jpg[/IMG][/quote]
lol josh where you been son

What you want is a Group Channel track. Once you create the track, name it something like "Drums". Then change the output on the desired drum tracks to Drums instead of Stereo 1.

Anyone adept at Side chaining drum tracks. I want to mix a little into my drum group (kick, snare, cymbals). Do i have to set up a seperate send from my group track for the compression, or just use it as an insert on the group channel?

No idea how to do it in Cubase sorry.

In Bro Tools you just set up a send from the track you want to trigger. Then for example in the compressor plug in you just select the side chain feature and all is a go.





When are people going to learn...everything in Pro Tools makes life so much easier ;)

Yeah Pro Tools is really great. Especially with pro tools 9 not needing an interface to mix, it's getting even better.

I was actually recording a vocalist today who said she wanted the pro tools file, but also the seperate audio files in a different folder. She said she wanted to be able to do it in pro tools but then bring it in to garage band to make it way better. I just laughed to myself.

what do you think of my cover? D:

http://www.youtube.com/watch?v=CtWMkpCPXEI

Ya know, I think you definitely have to have a nice front end, quality mic, preamp, D/A...I mean, you are absolutely not going to sound good if your front end sucks. But monitors, I'm not so sure. The most important thing regarding monitors, IMHO, is to know them, know what they sound like in reference to your headphones, your car, home laptop, your home stereo, etc. Like, who cares if they're flat? What environment is flat, anyways? Save your money, and spend some extra time getting to know your monitors, and your room.

[quote=interluder;18486421]But monitors, I'm not so sure. The most important thing regarding monitors, IMHO, is to know them, know what they sound like in reference to your headphones, your car, home laptop, your home stereo, etc. Like, who cares if they're flat? What environment is flat, anyways? Save your money, and spend some extra time getting to know your monitors, and your room.[/quote]

You're ignoring the possibility that the monitor/room chain might do some terrible, terrible things that make it impossible to get any sort of accurate sonic picture.

For example, you might have a large node in the fundamental speech frequencies. No amount of "learning" is ever going to make up for the fact that certain important frequencies [I]just don't exist[/I] in your work environment. This is also true the extremes of frequency range as well: if they aren't getting output there's no way you can work with them in the first place.

I agree (more or less) about the "flatness" issue, but you gotta have a reasonable starting point to build from. Also keep in mind that having a dynamite front-end means diddly if you can't hear those differences anyway because of your horrible monitoring (i.e., from your perspective, you've just wasted your money on the premium stuff).

i agree with moseph, a neuman u87, through a1073 with an 1176 using the apogee a/d d/a converters isn't going to mean shit if you cant hear what it really sounds like.

I'd also like to add that a flat room is key to making good recordings/ mixes. Micing up a snare top with an sm57 is common and can make for an amazing snare sound, the main problem most new engineers run into is room sound, not only live room sound but their monitoring setup as well, if your room is reflecting all over the place and phasing out your monitors (which are also coloring the sound) you're not going to know what it actually sounds like to the mic

[quote=Xomblies;18487557]I'd also like to add that a flat room is key to making good recordings/ mixes. Micing up a snare top with an sm57 is common and can make for an amazing snare sound, the main problem most new engineers run into is room sound, not only live room sound but their monitoring setup as well, if your room is reflecting all over the place and phasing out your monitors (which are also coloring the sound) you're not going to know what it actually sounds like to the mic[/quote]

I definitely concede that "flat" is the best conditions for your monitoring environment. It's the closest approximation of a "generic" room because its neutrality means you're mostly hearing just the audio. I don't agree that it's [I]absolutely[/I] essential: I'd rather have a decent-sounding room for $1500 worth of work than a perfect room for $15,000. There are more critical things to spend the money on than a "perfect" listening space.


One thing I'm less certain of, however, is how treatments should affect sound transmission in terms of reflections/diffusion. "Live" and "dead" both have their advantages, and neither one seems like the clear winner to me.

How many people listen to music in a dead room

shitty post nm

[QUOTE=Kuffuffled;18487613]How many people listen to music in a dead room

shitty post nm[/QUOTE]

it's not about listening in a dead room it's about getting a neutral sound so that way it doesn't matter what kind of room you listen in ;)

Yea I couldn't make a sentence to fix my statement so I said f it

That works^

[QUOTE=Moseph;18487597]I definitely concede that "flat" is the best conditions for your monitoring environment. It's the closest approximation of a "generic" room because its neutrality means you're mostly hearing just the audio. I don't agree that it's [I]absolutely[/I] essential: I'd rather have a decent-sounding room for $1500 worth of work than a perfect room for $15,000. There are more critical things to spend the money on than a "perfect" listening space.


One thing I'm less certain of, however, is how treatments should affect sound transmission in terms of reflections/diffusion. "Live" and "dead" both have their advantages, and neither one seems like the clear winner to me.[/QUOTE]

moseph, when you hear my shit in its entirety you'll shut the fuck up and stop half agreeing with me :p

[quote=Xomblies;18487666]moseph, when you hear my shit in its entirety you'll shut the fuck up and stop half agreeing with me :p[/quote]


I don't know man, I've heard this spiel before: big overture, moderate show :)