[quote=The Transporter;18451000]why would you want your drums on one track[/quote]

Off the top of my head:

(01) You're working with a limited-track device (solid state recorder, multi-track cassette device, or open reel recorder). In other words, you gotta do some bouncing to get all the parts onto a single medium.

(02) You're shooting for a minimalist style of recording, so you might be using only 1-4 mics for a fairly large kit anyway.

(03) You're shooting for a retro-sound: back in the early 60s (and earlier) you often had all the drums on one track (or even one microphone).

[quote=Xomblies;18451023]analog gear (even a mackie mixer) peaks at +4db where digital peaks at -24 i could be wrong with digital but i KNOW analog is +4db.[/quote]

Most internal digital mixing won't even soft-clip, since the actual math is being done using 32-bit (or higher) numbers and you're only keeping the top 24 (or 16, but that's getting more rare). And so long as you keep your overflow mathematics within the proper ranges on the output stage, you won't ever actually see any issues.

The real limitation you're looking at is likely going to be the bit-depth of your digital-to-analog conversion, which for most gear nowadays is probably based on the +4 dBu line level standard. So if you're gonna run into headroom issues, it's pretty much always going to be at the DAC.

Also, there's lots of analog out there that's actually based on the -10 dBV standard. A lot of it is older stuff, and you'll generally see unbalanced outputs using that standard (for example, older format recorders and contemporary gear designed to be compatible with it). But you can't assume that "analog" = "+4 dBU."

The best solution would of course be to check any manuals and/or markings on the gear. Otherwise, the balanced/unbalanced convention tends to hold pretty true, unless you're using vintage gear from prior to about 1985, where I can't claim enough knowledge to say if that holds up or not.

1176, distressors, api 2500 and the ssl xrack summing is all +4db, i assumed everything was but that is a good thing to look into if you were say mastering with mid level gear. i guess yeah you could prettymuch get away with a mack attack if you know what you're looking for :)

you're saying even if i bust out to analog and sum, it won't soft clip unless i'm at some crazy bit rate?

[quote=Xomblies;18451067]1176, distressors, api 2500 and the ssl xrack summing is all +4db, i assumed everything was but that is a good thing to look into if you were say mastering with mid level gear. i guess yeah you could prettymuch get away with a mack attack if you know what you're looking for :)[/quote]

I'm pretty sure DJ-oriented stuff still uses -10 dBV standards, but I'm really talking about older stuff, but not necessarily cheaper stuff.

All of my older formats use -10 dBV, even the "professional-grade" Tascam MSR-16. Even the 90s-era DTRS (DA-78HR) and ADAT (XT-20) machines have both levels as an output.


[quote=Xomblies;18451067]you're saying even if i bust out to analog and sum, it won't soft clip unless i'm at some crazy bit rate?[/quote]

No, I'm saying that as long as you stay inside-the-box, you're probably not gonna actually clip during the processing. Clipping is really a hardware problem, so unless you clip on the ADC, or you don't drop down into the proper range before hitting the DAC, your headroom problems are actually artificial. You can probably verify this in a superficial way using any DAW software: set up a regular channel feeding a "master" channel that feeds the DAC. Use a non-clipped audio file and play it through the regular channel. Turn up the fader on the original channel until that channel clips, but turn down the master fader again so it won't. You shouldn't hear any artifacts of clipping on the output, even though the red lights are blinking on earlier in your digital chain.

If you really want to know more about what's going on there, look into Two's Complement Mathematics. It's pretty simple to understand, but not straightforward to describe using text alone (like in a forum such as this).

hmmm, that still doesn't explain the perceived loudness and why so many mastering engineers destroy their input converters by "clipping" clearly it does SOMETHING otherwise the "in the box" masters would sound as loud as what these guys are getting

[quote=Xomblies;18451076]hmmm, that still doesn't explain the perceived loudness and why so many mastering engineers destroy their input converters by "clipping" clearly it does SOMETHING otherwise the "in the box" masters would sound as loud as what these guys are getting[/quote]


Well, there's a couple things that I think got lost in the discussion.

(01) Note that I explicitly stated your experiment needs to use a clip-free audio source. If you clip on the input, then it's always gonna be there.

(02) What I'm really getting at here, is that clipping is a hardware issue. The converters (both ADC and DAC) are hardware components. As long as you have a well-formed signal feeding the inputs of both those components, you won't get any "audible" clipping. In the abstract digital realm, clipping happens, but it's largely artificial, since the math they use in computers inherently allows for accurate overflow representations mid-process. Again, you gotta drop back into the expressible range before you hit the final stage.

(03) The other big caveat I gave last post was "as long as you stay inside the box." This is basically lip service to #2.

(04) I'm pretty sure I've heard ITB masters that were just as loud as something put out by analog/hybrid engineers. Now, the question of whether or not those sounded "as good" as the analog/hybrid guys is a wholly separate matter. Keep in mind that the vast majority of audio heard nowadays is digital, so you're always working with the limits of digital full-scale anyway (regardless of how "analog" your signal chain was).

(05) The big thing I wanted to point out was just that digital isn't always -24 dB (what standard are you referring to there? I'm not familiar with it) and that analog isn't always +4 dBu. Everything else is just me dwelling on parts that aren't important in practice (as I tend to do).

EDIT: one other thing that occurs to me. My strict "in-the-box" requirement also means you can't even actually [I]listen[/I] to the audio signal without having to meet the conditions for the DAC (i.e., the values are within the range of the DAC's proper operation). Sound (that is, physical vibrations of air molecules) is inherently analog, so that should kind of illustrate how pedantic the discussion has actually gotten here. #5 is really the only important part in practice, by a long stretch.

[QUOTE=Moseph;18451092]Well, there's a couple things that I think got lost in the discussion.

(01) Note that I explicitly stated your experiment needs to use a clip-free audio source. If you clip on the input, then it's always gonna be there.

(02) What I'm really getting at here, is that clipping is a hardware issue. The converters (both ADC and DAC) are hardware components. As long as you have a well-formed signal feeding the inputs of both those components, you won't get any "audible" clipping. In the abstract digital realm, clipping happens, but it's largely artificial, since the math they use in computers inherently allows for accurate overflow representations mid-process. Again, you gotta drop back into the expressible range before you hit the final stage.

(03) The other big caveat I gave last post was "as long as you stay inside the box." This is basically lip service to #2.

(04) I'm pretty sure I've heard ITB masters that were just as loud as something put out by analog/hybrid engineers. Now, the question of whether or not those sounded "as good" as the analog/hybrid guys is a wholly separate matter. Keep in mind that the vast majority of audio heard nowadays is digital, so you're always working with the limits of digital full-scale anyway (regardless of how "analog" your signal chain was).

(05) The big thing I wanted to point out was just that digital isn't always -24 dB (what standard are you referring to there? I'm not familiar with it) and that analog isn't always +4 dBu. Everything else is just me dwelling on parts that aren't important in practice (as I tend to do).

EDIT: one other thing that occurs to me. My strict "in-the-box" requirement also means you can't even actually [I]listen[/I] to the audio signal without having to meet the conditions for the DAC (i.e., the values are within the range of the DAC's proper operation). Sound (that is, physical vibrations of air molecules) is inherently analog, so that should kind of illustrate how pedantic the discussion has actually gotten here. #5 is really the only important part in practice, by a long stretch.[/QUOTE]

i'll have to ask, but those numbers are what mastering engineers refer to, i think most of it really has to do with perceived loudness. Not sure exactly how it all works, but i know it does

+4dBu is simply an operating level. It doesn't have anything to do with loudness does it?

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.

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=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.