when it's time, it will come to you

:)

[quote=EmbraceRandom;18299378]those mackies are ok, a good starting point at least!

edit:
regarding the PT9 crossgrade, I don't need anything else do I? I.e., i don't have to buy a new iLok do I, if I've already got one?

Just making sure they haven't produced a new iLok or anything...

Also, besides the auto delay comp, what else am I gaining (as a PTLE8+MPTK2 owner) by crossgrading?[/quote]
They have got a new iLok out. But you don't need it!

I do though, it's so much sexier and holds so many more licenses.

[quote=EmbraceRandom;18300019]:(

I want to [I]want[/I] to get it, kinda seems as though I need someone to persuade me to get it[/quote]
Are you on PC?/Windows 7?

PC with Vista 32-bit Home Premium.

Come on Convec, persuade me!!

edit: man that softube FET is unreal!

[quote=EmbraceRandom;18300710]persuade me!![/quote]

Finding reasons isn't so hard. Whether or not they're persuasive is up to you:


(01) There's no guarantee that they'll offer the upgrade price forever. If you don't take advantage of it now, you might end up footing the full price down the road.

(02) They will likely drop support for version 8 as much as 12 months before they drop support for 9.

(03) You can finally ditch the cornball Digidesign hardware and get something that isn't manufactured based on the fact that the software is actually why people are buying it.

(04) Typically speaking, any outside talent you work with isn't going to be impressed unless you have the latest & greatest versions of everything, so it makes sense to stay current.

(05) You can almost double the number of simultaneous I/O with your new non-cornball hardware, meaning you've got a lot more flexibility with capturing large groups (or multi-miking sources) and also creating different headphone mixes.

(06) You can wander into any random workspace with your iLok and use whatever hardware is available at any given time, which is good if you travel between studio locations or want to do any work on the road.

(07) The fact that you [I]want[/I] to want this is probably just an indication that you actually want this but don't want to admit it.

(08) You can be the cool kid with the brand new bike and make everybody else jealous.

Haha I like how we go from the more salient points to 'you can be the cool kid' :D

Don't get me wrong, I do want it, but that's because I like spending money (that I should really save for food etc) on gear/software. I just had to warrant spending £90.

Fair points though, especially the first two. You may have persuaded me.

PT9, purchased.

You motherfuckers :p

Cool, when should we start rattling off lists of reasons you should buy Cubase?

Haha neveeeeeeerrrr!!!

[QUOTE=Moseph;18300945]Cool, when should we start rattling off lists of reasons you should buy Cubase?[/QUOTE]

eew

sorry but cubase is super gay

reaper is the sex, and there is no reason not to try it since (a) the demo goes for ever and isn't crippled (b) it's dirt cheap

After all that, the motherfuckers have cancelled my order because they refuse to ship out of the states and the UK store doesn't have the PT9 crossgrade :mad:

where did you buy from?

studica.com

i thought id ask this here.
does anyone know a good usb microphone to use when recording my guitar?
i thinking of purchasing one when my job comes through and i was thinking of going like $100 - $150.
anyone have experience recording with usb condenser microphones on their amps?

here's a really good answer:

go to a studio and spend a few hours with someone who knows their shit for 30 bucks an hour

edit: i don't just mean the closest engineer, find out what's got a good rep, you can learn from them too :)

[QUOTE=Xomblies;18301864]here's a really good answer:

go to a studio and spend a few hours with someone who knows their shit for 30 bucks an hour

edit: i don't just mean the closest engineer, find out what's got a good rep, you can learn from them too :)[/QUOTE]

well, i dont have any actual songs or anything.
i just wanna do home demos for little ideas and whatnot, but i still want it to sound somewhat decent.

i'd spend the money on some kind of usb interface for DIing your guitar then. There's some really good amp modeling plugins out there, and you can do a lot more with a DAW/ interface than you can with just a usb mic

edit:

since avid has come out with a new line of mboxen the last gens are WAY cheaper:

[url]http://pro-audio.musiciansfriend.com/product/Digidesign-Mbox-2-Mini-?sku=700498[/url]

so id just get an interface then a shure sm57 or something?
sounds like a good idea, more money spent but i guess it would be a good idea thanks

no problem, there are cheaper interfaces too, but i like digi's clocks better than the others.

just browse around and i'm sure you'll find something that suits you best. And yeah that and a 57 is just about all you'd need, and if you wanna do shit late at night with headphones just Di the guitar and REAMP later

how exactly would i DI my amp anyways
i got a fender hot rod deluxe.
im very noob at this so excuse the lame questions

DI means direct input, you record a completely dry signal as it sounds without going through an amp. You'd still be playing through your amp and would hear it as such but the recorded signal would be completely dry. The purpose of this is so you're free to change the sound you want later via reamping without having to record additional takes.

You'd be using a DI box like this one:
[URL]http://www.countryman.com/store/product.asp?id=52&catid=10[/URL]

Fender Hot Rod's are pretty nice amps, with a 57 and a half decent interface you can get some good tones.

Plus to reamp properly you'll need a reamping box which is more money again. Or a DI that's able to be used in reverse.

Im dragging this thread back from the dead.

When mixing drums, do you guys prefer to have every track peaking as close to 0db as possible?
Im getting my levels for the drums individually before i bus them all to one track and would like some insight into how other people do it.

why would you want your drums on one track

no way, i try to give everything a bit of breathing room away from 0 db. you can always make it louder during mastering, but i always try to mix low and in the green

yes

"mixing in the green" (healthy amount under 0 dB) is absolutely, 100% the way to do it

do not mix orange yellow or red ever under any circumstances

or youll end up like periphery. botting the internet forums trying desperately to get one more fan to put food on your over compressed, poorly arranged table

mix drums on individual tracks, them sum to one stereo track with a comp on it. If you REALLY want to get crazy with it, bust all the drums out of your interface into some kind of summing mixer, 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. If you've got any kind of outboard gear, using it before you sum down to the stereo mix will sound drastically different than it's modeled plugin. Reason being is when you hit hard electrical circuits you'll get a drive that the plugins can't replicate. plus processing audio and summing down in analog form does something to the presence of the drums.

yay analog!

yeah i think that's why that periphery album sounds super compressed, bulb must have mastered that shit in the box. the guy who did underoath's define the great line busted out with analog, did a left, center and right channel "printed" two audio tracks and snagged those out of the audio files folder rather than bouncing. you will get louder mixes WITH dynamics if you do it that way

EDIT: forgot to mention, if you have good converters getting a good master is easier as well... you could essentially smash your input as clipping and it'll act as a limiter... if that makes sense. Something like an apogee duet with their self proclaimed "soft limiters" might help achieve maximal loudaucity

[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.