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ChristsDickWorship: Dec 7, 2004; Annihilate your demons

nigga plz posted:

From another forum - "AA 1.5 uses the new WDM drivers while AA 2.0 uses the ASIO drivers."

So I can only record 2 tracks at a time using WDM?

What you quoted seems to imply you should look for WDM drivers that are specifically labeled as "multichannel." It seems like WDM isn't limited to 2 channels, but some drivers only allow 2 inputs at a time. I'm really not sure because I've always used ASIO instead.

# ¿ Jul 14, 2007 21:52

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# ¿ May 16, 2024 03:11

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Laws posted:

I thought the MBoxes come with built in focusright preamps. Are the condenser mics I'm considering better served by outboard preamps?

Well, they don't come with Focusrite anymore (the original MBox did, the MBox2 stuff does not because Focusrite went and made their own interfaces), but they come with decent preamps. I think you'll be fine with the onboard preamps for a while, but it's an option you can upgrade later.

I agree with Elder though about Pro Tools being a fine place to start. You can use Pro Tools hardware with any other software if you decide you absolutely can't stand it, although it will cost you more since you basically paid for a copy of Pro Tools with the price of your interface (unlike some cheaper interfaces that will give you a $100 version of Cubase rather than a $250 version). But with Digidesign hardware and software you don't have to worry about things like conflicts or making the software compensate for the delay inherent in the interface or anything like that, because Pro Tools LE only works with a handful of interfaces that are all owned by the same company. I think Pro Tools is much more plug and play, and not having any experience at all I don't think it's going to be any harder to dig into than any other software.

Also, before you get too worried about not spending enough money, think about mic stands and cables as well. Even just for 2 mics that could easily be another $150-$200.

# ¿ Jul 15, 2007 00:28

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

nigga plz posted:

And is this downloadable for free or do you have to pay?

You wouldn't have to pay. I'm sure Presonus would make WDM drivers available on their website if they exist. Like I said, I've never used them but I imagine they're all in the same package. I think there is actually a demand for them because Sonar/Cakewalk initially only used WDM and had ASIO added later, so people who run Sonar tend to want WDM as it is "more native."

I'm not really sure if you lose functionality or not with WDM drivers instead of ASIO. I want to say you don't, although there may be a slight difference in latency or something.

# ¿ Jul 15, 2007 07:59

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Dirk Muscleton posted:

Since when I used the "karaoke" option I ended up getting the complete opposite of what I wanted, it's actually rather promising, and leads me to believe that if I could get rid of the "music" (instead of the "vocals"), the result would be near-perfect.

Unfortunately that's not as possible as it seems. Cancelling out the center channel is easy because you're using things that exist on both sides equally (which is the definition of 'center' in stereo). Unfortunately it's not so easy to cancel things on the outsides because that's stuff that only exists in one channel, so there's nothing to cancel it with.

Basically there's no way to do that because although the pops and crackles are on the left, so is the stuff in the center. Any simple canceling like what a karaoke plug-in does will happen to everything on that side, not just the pops.

Maybe you'd have some luck if you can find some sort of plug-in designed to encode (or maybe they would call it 'demux'?) a stereo mix into a 5.1 mix? If you found some kind of processing that does that well, what it outputs as the center channel might be what you're looking for.

edit: errr, you could try reversing the phase of what you got out of your karaoke plug-in and summing it with the original track. v:)v

hurr, you already tried that

ChristsDickWorship fucked around with this message at 16:42 on Aug 21, 2007

# ¿ Aug 21, 2007 16:28

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Slowfuse posted:

I feel like I have my wires crossed here...does this mean: The signal goes into the I/O, becomes digital and is monitored back post software mix with 0 latency, or does it simply mean you can monitor the pre AD conversion input with 0 latency? (this doesn't seem like it would be much use, since you won't be able to hear what you are recording in the context of the other tracks in the software mix, or be able to hear any soft plug-ins you may be recording with)

It means the latter, but it is useful. The delay from the DAW to the output is generally negligible, it's going all the way in and coming all the way back out that can throw people off. For most interfaces there is software or controls on the unit to mix 0 latency incoming signal with playback from the outputs as well. MOTU's software for that is called Cue Mix I think, not sure about Alesis.

# ¿ Aug 29, 2007 05:29

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Slowfuse posted:

Oh I see, I thought that was the case...so you would simply have to mix in the raw signal with the stereo mix software playback whilst recording....

Thats a bit of a drag...

Latency with firewire interfaces isn't a huge concern, I was explaining how the 0 latency monitoring works. It usually isn't a factor with firewire, but they build it into most interfaces anyway. Latency is also not a problem with MIDI because on the input there is no conversion, it just sends control data and the only significant latency is your PC creating the sound then sending it out.

Processing a bunch of realtime plug-ins and instruments could get you into too much latency, however. Most firewire interfaces seem to have about 10ms of latency from input to output. You're going to start getting into trouble as you approach 30ms, maybe sooner maybe later depending on the musician. So if your DAW is loaded down with plug-ins and virtual instruments to the point that it is causing 15ms-20ms of latency you are starting to get into the danger zone.

The professionals do it with a mixing console routing all the inputs and outputs through it and generating headphone/monitoring mixes through that in real-time. Or by having PCI-based interfaces and DSP processing cards that total around 5ms of latency from input to output even with plug-ins (Pro Tools HD). You can monitor through firewire interfaces fine as well, but you probably want to go easy on the plug-ins during tracking.

# ¿ Aug 29, 2007 23:50

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

StivBators posted:

Is there a standalone digital recorder that can do the overdrive of an analog recorder?

I that natural sound when you drive an analog recorder past peak, and it still maintains the complete signal. I guess with digital, all the bytes or bites get all filled up, and it just clips.

But, if I could find a digital recorder in which the A/D conversion happens post-mixer, that would be great, as I could still get that nice analog overdrive tone.

Any help would be most appreciated. Thanks!

If you want to do this you'll probably need a separate analog mixer feeding your A/D convertors. Then you can overdrive the preamp and send that channel to a bus or aux, route the bus/aux output to the convertor and turn the output down until you have ideal levels feeding the recorder.

A lot of high-end digital mixers for concerts or recording provide this feature, where you can overdrive the preamp then attenuate the signal before it goes to the ADC so you clip the preamp but not the convertor, but I don't know of any recording interfaces that sport this feature.

It's not going to sound like tape distortion exactly, if that's what you're looking for, but you can at least make the preamp distort. If you're looking for tape distortion in a DAW your best bet is to find a plug-in that does it. Won't be perfect but I know many people who are happier with that than with nothing at all.

# ¿ May 15, 2008 07:07

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

DominionOfSatyr posted:

I have a quick question:

I want to trigger my Bass Drum directly to my Firepod to Cubase for recording. Can it go straight from the trigger to Cubase, without the need of a module/controller? Can't I just set up a MIDI track and record ? ..or am I just crazy?

What comes directly out of your trigger is actually an audio signal. You could record the audio from your trigger and then sound-replace the clicks you get from it with Drumagog or something similar, but the module/controller is what translates those clicks from the triggers into MIDI data, so without one you cannot record it as MIDI.

# ¿ Jun 11, 2008 16:31

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

RivensBitch posted:

I've never heard of this...?

If you move your 002R a lot (like in a mobile rack, being shipped, etc), the power harness inside the case likes to come unseated. Most people who use them outside an installed studio setting and depend on them to work all the time end up opening them and gluing or soldering it in place. I believe that's what Digidesign unofficially recommends if you call them up and complain to them about it. In a kind of "if there were a problem this is what we would recommend, also we can mail you instructions on how to do it properly" way.

# ¿ Jun 15, 2008 11:47

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

iamlark posted:

Has anyone here used the SSL Duende yet? I'm looking at your Rivensbitch. I'm thinking about getting one for some better EQs.

I haven't used the Duende so if you're asking about the hardware I don't know.

I have used the Waves SSL bundle, and honestly I was not impressed. I would end up just ditching all of their channel strips for Digidesign EQ 3s and another compressor because they ate up a shitload of processor and I didn't think they were particularly clean or predictable like their analog counterparts. When you used a lot of them it sorta seemed like maybe you were getting some "analog" sound, but I remember deciding it was just an increase in noise floor.

But hey, a lot of people swear by them and I did like the bus compressor plug-in (not sure if that comes with the Duende).

# ¿ Jun 27, 2008 17:45

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

marrone posted:

I have a pretty decent a/v receiver. Is it possible I could use it as a preamp with a microphone, even though the receiver only has rca inputs?

I was thinking of using an xlr-rca cable to put the mic in an input, and connect my computer to the tape monitor input to record. Could this work?

Or would it be better for me to just take the mic and plug it into my line in with an xlr-trs cable?

There is no amplification happening from an input to the tape out.

# ¿ Jun 30, 2008 18:39

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

WanderingKid posted:

Edit: speaking of reverb, number 2 on my lust list is an Eventide Orville which is quite simply the most spectacular effects unit I have ever heard.

Have you ever used their Eclipse? I am seriously starting to think about dropping way too much money on a personal rack of dynamics and effects. What amounts to a 1U H3000 would be nice, but I'm kinda worried about it being tough to navigate.

And since we're talking about spending our life savings, the rack I want will ideally be from top to bottom:

TC D-Two (I play with delays on the fly a lot and this thing is just too easy to use... some other delays sound better but the complicated series of delays you can set up with this thing in 15s is unbelievable)
Eventide H3000 (or Eclipse if I ever get a chance to play with one)
Lexicon PCM90 (just for drum verb)
SSL XLogic G Series Bus Compressor (pure magic on the stereo bus)
2 Empirical Labs Distressors (or 4 ideally)
KT Square One Dynamics (8 utility compressors or gates)
2 Drawmer DS201s (just for toms)

In these times we live in that investment will be better than the stock market, right? :shobon:

# ¿ Jun 30, 2008 22:34

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

In a studio I might do something like that, but I would be buying my rack mostly for mixing live, and it would suck royally to have to navigate through an H8000 to get to all the different FX I might want to tweak on one little screen. Although believe me, it is tempting...

I am pretty familiar with the H3000 and pretty much all the Eventide stuff has the same navigation and programming. I'm just not sure about the Eclipse because its half the height. The studio I used to work at had an H3000 and many of the FX racks the audio company I work for now sends out to concerts have one in there. There are usually 3 or 4 other units that are easier to use in the rack and I'm always pressed for time at concerts, thus I rarely mess with the H3000 (just go to the TC M5000, SPX1000s and Lexicon PCMs that I know a lot better) so I'm a little out of practice. Most people stay away from them for straight up verbs because there are easier units to use for that and stick to the Eventide for the lead vocal. A lot of engineers straight up insert the H3000 (or whatever model) on the lead vocal channel instead of running it on an aux and just use the dry/wet mix in the unit to regulate the level on it.

The 7600 is the newer Orville isn't it? I don't think the Orville is in production anymore. I'm sure the newer stuff has more capability, but from what I've heard for the most part, Eventide is Eventide. The H3000 sounds incredible as well and the new stuff probably has better presets but from the samples you linked I didn't hear anything way more amazing than what I've heard out of an H3000 after someone took the time to tweak it out. I don't need anything more fancy than that probably, and that SSL bus compressor would get jealous if it wasn't the most expensive thing in the rack.

# ¿ Jul 1, 2008 01:35

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

marrone posted:

Even if I have a dynamic microphone?

Yup, they still need at least 20-30dB of gain to get to line level with someone yelling into them. Try and use them to record an acoustic guitar or something and they'll likely need more.

quote:

If I connected it to a mic preamp first, would that work?

Yes, a microphone preamp should do the trick to get between then microphone and the line-in. But probably not an AV preamp.

# ¿ Jul 1, 2008 05:44

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

RivensBitch posted:

Get a simple, cheap DAW host and route your keystation midi to the built in synth in your onboard sound card. Whats the free DAW everyone uses anyways, audition?

I think you mean Audacity.

# ¿ Jul 2, 2008 00:00

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Gibbon_WBA posted:

Thanks, I did this. No luck

So the Ozone does actually show up in your devices? That means it at least thinks the drivers are installed right.

I don't use Cubase, but the next logical step is to make sure the input of the track you are trying to record to is set to the Ozone's 1/4" instrument input to record your guitar. I'm not sure how you see a possible list of input sources in Cubase, but you should make sure it is recognizing the inputs and outputs of your Ozone and that the track you are trying to record to is looking at the correct input.

It might just default to the built-in mic.

Don't worry about the MIDI stuff it has nothing to do with recording audio.

# ¿ Jul 4, 2008 03:33

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Make Ready posted:

I've heard it's missing some features such as MacDrive and DV toolkit, so I'm taking the comments here with a grain of salt.

MacDrive is a 3rd party program that lets PCs read Mac formatted drives. It does not come with Pro Tools. DV Toolkit does not come with Pro Tools LE, it's an add-on. Pretty sure you can buy the expansion for M-Powered too.

I'm taking your previous sources of information with a grain of salt.

edit: You're right actually, DV Toolkit is not compatible with M-Powered. So yea, you can't spend that $1200 if you buy M-Powered. Also doesn't support Digitranslator, so that's another $500 you can't spend. If you don't have the money to jump head first into Digidesign hardware, you don't really have the money to actually miss any of what M-Powered is lacking.

EDIT: READ THE POST BELOW THAT IS WHY THE THREAD IS NAMED AFTER HIM

ChristsDickWorship fucked around with this message at 19:03 on Jul 14, 2008

# ¿ Jul 14, 2008 17:07

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

hmmxkrazee posted:

THe volume in the software is turned up and the mic knob is at about 6/10. If I make it higher it gets static-y and the "shhhhh" sound gets alot louder.

Yea, that's called noise floor. Poor preamps have high noise floors. It could also be room tone, just the air moving/fans/etc in your room being relatively loud in comaprison to what you're recording.

What are you trying to record? If you're mic'ing a guitar cabinet you shouldn't be having a noise floor problem I would think. If you're mic'ing a finger-picked acoustic (without finger picks) or fairly quiet vocals it doesn't surprise me that you'd run into a noise floor issue.

Compression will actually raise the noise floor more than the gain knob probably will (no promises though). Compression will lower the program material with respect to the noise floor, so while the whole signal gets louder the ratio of signal to noise gets lower. Adding gain to the input will raise the level of the hiss, but it should also equally raise the level of the signal so it should be your most ideal solution.

# ¿ Aug 14, 2008 17:22

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Rashomon posted:

What software sequencers does everyone here use? I have been doing Pro Tools that came with my MBox but I'm probably going to upgrade soon and possibly get a new computer just for recording. Logic seems awesome but I don't really know how I feel about getting a Mac. Cubase is supposed to be great but for some reason looking at screenshots I am underwhelmed -- I just don't feel like it LOOKS that good, not that that is necessarily a problem. I'd like to learn to do more MIDI stuff (I know nothing so far).

If you know nothing right now you might want to go ahead and get your feet wet with MIDI in Pro Tools. As far as production goes you can pretty much achieve the same results with any sequencer, so it's totally a matter of how you work and what you like. Unless you've tried and feel like Pro Tools' interface is making it harder for you to grasp MIDI, I think learning how to work with it in the environment you're most familiar with would be the best place to learn, and when you know enough to know precisely how Pro Tools is ruining your workflow you'll be able to make a more informed decision about where to go from there.

# ¿ Aug 18, 2008 20:22

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

I know very little about Logic, but when I complained to someone about how the meters made no god drat sense he told me that the default meter settings make no sense, but there is a way to change them. Unfortunately I am not offering an actual solution, but I think there is a solution to make them actually read dBV, which they do not seem to be metering by default.

# ¿ Oct 8, 2008 13:39

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

RivensBitch posted:

(this should be the title of the next megathread btw)

Only if we can arrange every first page reply to be "NO"

# ¿ Oct 9, 2008 03:43

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

John-311 posted:

Obviously the lighting guy isn't creating that live each time they play. What program do lighting designers use? Is there a cheap/freeware version?

You would be surprised. Lighting consoles are often designed, much moreso than sound consoles, to step from one "look" to the next. How many changes are made from look to look are entirely up to the lighting person. Many of them also provide a tap tempo button to tap out changes between however many looks you want. On bigger shows there are multiple lighting operators taking cues from one lighting director/designer, one person cueing all of the moving light programs and one person working on the par cans (in addition to the people running the spotlights).

In short, yes many lighting designers spend a lot of time before the show customizing programs and looks that will sync to the song. However, I've never seen a tour that just basically hits play and the lights chase it all. The LD is a pretty important part of every concert's crew.

One Martin Mac 500, which is what you see doing a lot of the moving and color changing in most shows is about $6000. Just for reference.

# ¿ Oct 12, 2008 03:28

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

I wasn't trying to be completely discouraging by throwing out huge numbers, just trying to point out that what you see at major shows is on a whole other level from what you can do on your own, in both price and in technically pulling it off.

There are certainly more affordable (although still not cheap) DMX-capable moving lights and LED lights that can be programmed and used creatively with things like American DJ's MyDMX interface/software and controlled with a laptop, although I have no idea how comprehensive something like MyDMX is in its support for other company's lights.

Just be aware it's a pretty involved process and their market isn't driven by home-users or amateurs at all. You're probably not going to find a product designed for people who might not want to be experts or power users but want to be able to do some cool custom stuff with lighting, whereas in the audio world there are lots of DAWs designed to make complex things easy to use.

# ¿ Oct 13, 2008 16:55

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Vaporware posted:

According to the tech guys it's the Gain Potentiometer, not the Pre-Amp.

Either A) they need a beefier preamp to get through that potentiometer, or B) they need to stop claiming their preamp has 55dB of gain when the potentiometer will only let you turn it up to 45dB or whatever. Either way, I tend to consider the potentiometer part of the preamp for all intents and purposes.

I have recently lost a shitload of respect for M-Audio. Their driver development has been laughable recently with changing over to Leopard and Vista and on top of that I bought a MobilePre in July that is already dead. If they are outfitting their gear with potentiometers that cripple their preamps that is just too much. You can redlight a Beta 57 with a Mackie VLZ or a Presonus Firebox, at normal speech levels, I promise. Whether their preamps have more gain or whether the taper of their pots is different I have no idea, but comparing that problem to those pieces of gear is disingenuous.

To answer your previous question, yes you want to record with peaks as close to 0dB as possible. You cannot really add gain later, certainly not as effectively as with a preamp on the input.

# ¿ Oct 15, 2008 21:33

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Coca Koala posted:

Am I trying to diagnose a problem with the guitar, or is there some crucial piece that I'm missing which goes in between my guitar and the preamp, and there's no way to connect an A/E guitar directly to the preamp for recording purposes?

What you're doing should be working if the guitar works fine with an amp. I assume you're using the 1/4" input on the front of the MobilePre?

If the buzz changes volume when you turn the knob, Audacity would seem to be looking at the right input. Have you tried changing to input 2 (on the back) and reassigning Audacity to record that one? Could be that your channel 1 1/4" input is borked.

# ¿ Oct 25, 2008 01:21

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

ReDiNmYhEaD posted:

mike,

I'm trimming drum takes to composite a track together. There are 9 audio tracks for the drum takes. I have trimmed the audio to an ideal clean hit. I have the audio pieces butted up to each other. Since the audio isn't overlapping I just want the volume to be at zero when the clips transition.

If the audio isn't overlapping how do I get rid of the clip I hear when the audio transitions between the two clips?

The clip you hear is the lack of audio and then it coming back in with no fade. You want to untrim them, overlap them and fade them together with a quick crossfade. You may find that you want to make new tracks some of them (maybe overheads or room mics) so that you can more slowly fade them out if possible and not make the transition as obvious. You may find yourself dropping the fade in different places on different tracks, or with differing lengths of crossfade. You'll want to do the editing track by track most likely.

Editing drum takes like that can be really difficult. Second only to editing acoustic grand piano recordings in my experience...

ChristsDickWorship fucked around with this message at 22:38 on Nov 5, 2008

# ¿ Nov 5, 2008 22:35

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

mclast posted:

I'm using a Macbook, 2.16Ghz Core 2 Duo, 2 gigs of RAM. Is this powerful enough for low-latency, decent recording? Should I invest in a better computer before recording gear?

That should do fine for a while. My Macbook Pro with the same processor and half the RAM is stable recording 6-8 tracks at a time with a Digi002 or mixing down sessions of 12-16 tracks with some plug-ins.

quote:

I'm looking at Pro-Tools compatible interfaces - as the industry standard, it seems like the best use of my time to start learning it. Are there significant differences between Pro Tools M-Powered and LE?

AFAIK one only works with M-Audio devices and one works with Digidesign devices. Functionally they are the same. I seem to remember M-Powered not being able to use some of the upgrades, like the DV Toolkit for working with video, but that probably won't effect you.

quote:

Is there a significant advantage to using MIDI in this situation? Also, are there significant advantages to FireWire over USB?

MIDI's significant advantage is that it is not audio, it is just signals telling your drum brain what samples to trigger. If you were to record the MIDI, you could change the drum samples inside Pro Tools later if you decided you wanted a different kick or snare sound. Or you could use a program like DKFH or BFD to generate all of your drum samples in the first place and just control it with your electronic kit.

Firewire interfaces tend to have lower latencies, although USB's latency is plenty usable.

quote:

I'm looking at the Mbox, the Mbox 2, or the Mbox 2 Pro, depending on the importance of MIDI and FireWire. Obviously, there are some pretty dramatic price differences between these devices. What should I be focusing on?

I would buy an MBox 2 before a used MBox, as I know a lot of MBoxes that don't work anymore (they haven't been manufactured in a couple of years and they weren't built incredibly well to start with IMO). You would buy the MBox 2 Pro if you wanted the extra audio inputs, not just to upgrade to firewire.

ChristsDickWorship fucked around with this message at 18:57 on Nov 24, 2008

# ¿ Nov 24, 2008 18:55

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Not Memorable posted:

live recording

Live recording is extremely difficult (and expensive) to multitrack. The 828 you linked has 10 "usable" inputs, and only 2 of them have gain control. The other 18 inputs are digital ins and would require you to have 2 or more 8-channel units with ADAT output to use, and if you aren't going to buy Behringer that will be another $400 each for something like a Presonus Digimax. Most places in 3000+ venues will have 48 channel consoles and will not have any qualms about using all of those channels. How will you choose which 26 to use in a situation with 30 channels?

The other issue is stability. You can't stop your DAW during a show in between songs, so you have to know that it isn't going to crash. A crash in the last song after it's been running for an hour could lose you the entire session. Crashes start getting a lot more common when you're recording 20 tracks than when you're recording 3 or 4.

My suggestion? Spend the $800 you would spend on an 828 and get a pair of decent large diaphragm condensers. Like a pair of Shure KSM27s or something similar. Get a 4-input interface instead of a 10 to 28-input interface and record the boardfeed and put the microphones on a tall stand in a good part of the room, or in front of the soundguy since the mix is usually best near where he's hearing it. This way you have the sound in the room (and the audience is usually a pretty cool thing to have in the audio of a live video shoot) and you have the boardfeed to make things clearer. If you're not much of an audio guy you really aren't going to have fun mixing down multitrack sessions that came off a stage anyway.

Also, I'm pretty sure you just didn't set your levels right in that blip link you posted. That mix sounds really good for a boardfeed, although it is vocal heavy (boardfeeds always are because guitar players can't set their amps at reasonable volumes so there is never much guitar in the mix). It probably wasn't clipping in the room, it is probably just overloading the input on your camera. You need to pay close attention to that. Would you put up a camera, focus it once, white balance it at soundcheck when the house lights were on and then never pay attention to it again? Probably not, and you can't just set and forget audio either.

stun runner posted:

Is the MOTU Ultralite MKIII generally recommended? It does what I want and then some, and it looks like it has pretty good reviews overall but I thought I'd check with you guys first. I'd be using it with either a Macbook or a G5.

I've only heard good things about them generally speaking. I've had bad experiences with the MOTU Traveler, but never actually used the Ultralite.

# ¿ Dec 1, 2008 00:38

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

That's what the room mics are for. If the mix sounds good in the room, I would think of them as your primary recording and then when it turns out the vocals are a little lost or muddy like they can be at shows, you have this vocal-heavy boardfeed to blend in and fix it. Think of it as not having enough guitar and drums rather than having too much vocal because that's what's happening: the soundguy is reinforcing the stage volume so it sounds right in the room. Unless the soundguy just totally sucks, the things that aren't prominent in the boardfeed will be very prominent in the room.

Maybe you can find a bootlegger/taper kinda guy who will help you out for some beers or something? There's no one way to describe the best place to put your mics or the best way to process them afterwards, but a local taper will probably know the best places in most of the local venues. You could always experiment at shows you weren't videoing if the bands/clubs are cool with it.

If you are serious about multitracking, the MOTU 8Pre is cheaper than the 828 and would give you 8 adjustable gain inputs to have some control over what the soundguy is sending you, and you could add 8 more with a Digimax or whatever later for a few hundred more. 16 inputs would let you multitrack a rock band pretty comfortably most of the time, even if they're a headliner since you could choose the channels you need and leave off extraneous ones. I've never used an 8Pre and don't personally know anyone who has one, but maybe somebody here does?

# ¿ Dec 1, 2008 03:12

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

It will usually refer to it as a 4x2 (4 in, 2 out) or something in that format. Usually when they're talking about ins and outs in the same phrase like that they mean simultaneous and when they're describing what all the jacks are on the unit you shouldn't assume they're all simultaneous. You do have to read the fine print, they don't make it easy.

I'm pretty sure you're right about the 2 interfaces you're talking about though: the Saffire can record 6 inputs at a time (including 2 digital) and the Lambda only has 2. But there are some interfaces that will label inputs Mic 1, Mic 2, Line 1, Line 2 and you can actually use all of them, so just looking for numbers may not always work.

# ¿ Dec 1, 2008 06:23

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

RivensBitch posted:

uk10rguy - I think you're overcomplicating things here.

...

As for removing plosives, try a compressor with a sidechain input. Create an aux track and bus the output of your vocal channel to it, then put an EQ on the aux channel. Now using hipass and lowpass filters, isolate the frequencies of the sound you want to de-emphasize (low frequencies for plosives, high frequencies for S sounds). Now set the output of your aux to another bus, put a compressor on your main vocal channel, and set the key/sidechain input as the bus your aux channel is outputting to. Now you can compress JUST the plosives or S sounds, because those sounds are what is triggering the threshold of your compressor.

I think you're overcomplicating things a bit as well. That process will not JUST compress the plosives, it will compress the whole signal only when there are plosives. The thing about plosives though is they are already almost always the hottest parts of the signal in the first place, so just compressing the track without a filtered sidechain will usually do the same. Isolating the plosives to trigger a compressor could easily result in a vocal track that pumps oddly. Think about what a Justice track would sound like if the kick drum was muted...

It makes a little more sense for sibilance/de-essing because those sounds are generally just annoying distortions, not noticeably hotter than the rest of the track, so a filtered sidechain engages the compressor when the original track might not.

The ideal tool for plosives or sibilance is "dynamic EQ," which is a processor that has threshold-based EQ filters. Basically think about a compressor, only when a particular frequency range hits the threshold it engages an EQ filter rather than gain reduction. I'm not sure how common these plug-ins are or if they are always referred to as "dynamic EQ" but that's what BSS and TC call their units that do it. Those are the 2 I've used and they work great (assuming TC's EQ Station has the same algorithms as the Powercore), but I'm sure there are others that will work well too.

ChristsDickWorship fucked around with this message at 03:56 on Dec 21, 2008

# ¿ Dec 21, 2008 03:53

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

RivensBitch posted:

I've usually been able to get a sidechain compressor to work on plosives if I'm concentrating the filter on 100hz and below, but your point is well taken. I have dynamic EQ on my powercore and I'll definately try that out the next time I'm working with a track like that. But of course, I usually don't run into this problem as I a) have a pop filter, and b) lately have only been working with vocalists who have enough mic technique to know not to blow their consonants that hard.

True, the best way to remove them is in the tracking room. I have a lot of experience fighting with them because I do a decent amount of live sound with lavaliers, which is a nightmare of plosives and proximity effect depending on how the person has their head turned, even if they do know how to speak well. It's rare that my high pass filter leaves much of anything under 100Hz on vocals in most of my mixes.

There are actually compressors, like the Distressor, that have optional HPFs on the sensing circuit so even if you don't EQ plosives out before the compression, plosives won't trigger gain reduction and cause that annoying pumping.

# ¿ Dec 21, 2008 20:21

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

mezzir posted:

Looks like TDM doesn't have formant, pitch, and throat modelling knobs. And also as far as I can tell, either TDM or Native are btoh available in RTAS (protools format) or VST.

TDM is not VST or RTAS, it's a format designed to run on dedicated DSP chips. The name means "Time Division Multiplexing" and comes from the old Pro Tools "TDM" systems which are now basically called "Pro Tools HD" systems in their present form. Pretty sure many DSP processing cards support that format now, so if you have a standalone DSP card to run plug-ins, TDM is what you want.

Also, it's only those "automatic" features that don't work with the TDM version, probably because they aren't really meant to be run in real-time. You get full Native licenses with your TDM version though, so you could use those features on your host CPU.

# ¿ Dec 31, 2008 18:08

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

RivensBitch posted:

TDM is ONLY for protools HD, I am not aware of any other DSP cards that can run it.

That was my initial thinking, but when I looked it up I saw that Digidesign had opened the standard so it's not proprietary anymore and I figured more cards were supporting it. Upon closer inspection it looks like other companies are only using it for routing on soundcards, not for running plug-ins.

# ¿ Dec 31, 2008 18:52

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

My MBox does not work anymore, personally. I only know one person who still has a working MBox. I hear their most common point of failure is their USB port.

Odd that it might start acting up right when you buy Autotune, but if the iLok isn't plugged in and Pro Tools is failing, I'm not sure what else to tell you...

# ¿ Jan 14, 2009 17:42

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

RivensBitch posted:

My MBOX is in excellent condition and I've probably had it for 4 years now. I don't understand how anyone can have a broken Mbox lying around and not bother to crack the case open to see what might be the problem. Broken USB port? Solder the fucker back in place. What are you going to do, void the 30 day digidesign warranty?

The one I know that works was soldered, as was that guy's 002R when the power harness kept coming loose inside of it. Mine isn't as obvious a fix because mine has some sort of crosstalk on the outputs, although it still records clean. Yea, sure, I might be able to fix it if I opened it up but there are other things that annoy me about the MBox when using it as anything but a Pro Tools dongle (like why can't I plug in a line level signal on an XLR cable?).

I probably know more people with working ones, but as far as people who move them around a lot and use them all the time...

Personally I'm pretty rough on gear. My MBox lasted 3.5 years or so, the MobilePre I bought to replace it for carrying around to concerts only lasted like 6 weeks in my backpack before some kind of horrible phantom power loop started happening, making it worthless for use with my RTA mic. I think I'm going to buy one of these as soon as my next paycheck clears. Those things are built like tanks for road warriors and location video stuff.

# ¿ Jan 15, 2009 21:22

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Three Red Lights posted:

This may be a stupid question but would a faster PC reduce my latency? Or am I stuck with the same latency as long as I'm using the same audio interface?

A faster computer isn't going to increase the speed that the audio goes into and back out of the PC as that relies on the A/D, the implementation of the protocol, your divers, etc, but it could shorten the time it spends processing while it's inside.

One thing that may minimize latency is running at the highest sample rate you can, assuming that doesn't make your hard drive or CPU choke. There probably won't be a noticeable difference between 48KHz and 44.1KHz, but at 96KHz the time it takes to convert the audio is halved compared to 48KHz because of the way A/D converters work. For instance, a console like this goes from 5-6ms of latency at 48KHz to 2-3ms at 96KHz.

I'm honestly not sure how significantly this would affect a DAW's latency because I've never tried it, but theoretically it should. With a console like what I linked, all the processing is designed to run on DSP in real-time so you see almost every ms that gets saved by A/D, but I'm not sure if it will be so obvious with so many other variables in the signal path of your DAW.

# ¿ Jan 17, 2009 19:57

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Chipyy posted:

As far as I can make out, the AKG 240S has been discontinued and replaced with the 240 MKII, which is supposedly not built to quite the same standards (although still very good), and the response is not quite as flat.

If you're interested in the 240S I'm pretty sure you can still get them. That zzounds page is almost certainly not referring to the mkIIs because the picture and price are wrong if it is (mkIIs are all black and $200).

# ¿ Jan 30, 2009 16:05

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

Apathy=Awesome posted:

Can anyone tell me if this is a good deal?

http://www.gigasonic.com/m-audio-audiophile-2496-powered-7.html

Thanks guys

It's a good idea if you were planning to buy Pro Tools M-Powered anyway since you're basically getting the interface for $10 for buying the software.

If you don't have another interface though, that one isn't going to be adequate for recording microphones by itself with no preamps and RCA connectors instead of XLR. What are your plans for the gear?

# ¿ Feb 12, 2009 00:34

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# ¿ May 16, 2024 03:11

ChristsDickWorship: Dec 7, 2004; Annihilate your demons

The difference between using a send and a bus is that a bus will follow the channel's master fader to decide how much level to send to the reverb. With a send you get to decide, so for instance you could send background vocals hotter to the reverb than the lead and the lead will sound more up front while the backgrounds sound farther away, regardless of where their faders were set for the mix.

Pretty much everyone uses sends instead of buses because of that.

If you arent sending your channel to the master bus it isn't in the mix. If you're still hearing it it's because your reverb plugin isn't set to 100% wet. You really want to route everything to the master, unless you are using buses to compress groups of things and then routing that bus to the master.

edit: Realizing this post might be unclear here's an example of how buses will screw you:

OK, you have 5 vocals. Say you bus them all to a reverb and use the dry/wet setting on the reverb to control how much reverb there is. If you later decide to turn up one of the vocals for a lead, you will also be turning up the reverb on that vocal. You usually don't want to put more reverb on something you want to cut through and be up front. If you then turn the wet/dry knob to fix that, you are turning down the reverb on every other vocal. You may find yourself having to rework this entire gain structure every time you touch the vocal faders.

If you use an aux send you basically have a separate fader for the amount of reverb and the amount of the channel in the mix, which gives you much more refined control. Of course you will want your reverb set to 100% wet probably.

ChristsDickWorship fucked around with this message at 17:57 on Feb 17, 2009

# ¿ Feb 17, 2009 17:40

The Something Awful Forums > The Finer Arts > No More Drake > Musician's Lounge > HOME RECORDING MEGATHREAD: Rivensbitch Teaches Recording

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