Jean-christophe Boulay
Forum Replies Created
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What I love is that those super-advanced computers all go “blip-blip” every time they touch a button. That must be so annoying! That’s one of my sound design pet peves along with the concealed bomb that emits a 90 dB SPL “blip-blop”.
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com -
Even though 400 and 600 Hz may be easier on the ears, they are not reference tones per se. The main level tone for any reference level is and always will be 1kHz. Other frequencies should be used to assure alignment along the whole frequency range between two system. Since the end of analog tape, the 20, 150, 400, 600Hz and 10 and 20 kHz tones see almost no use. They are still available on certain machines, though.
400 and 600 will be pretty damn close to 1kHz on the VUs anyhow, and this won’t create any problems in practice, but any sound engineer getting your stuff down the line will not be super-impressed. Standards exist for a reason, and the 1k tone is one of them. Even if this all works in practice, don’t be surprised if you hear back from a studio engineer wondering what the hell you were doing. I’d scratch my head for sure.
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com -
I work on 737s every day and recommend them whole-heartedly. They don’t have a transparent sound by any means, but if you’re going to record VOs, you’ll probably be processing your sound to get somewhere in the 737 vicinity anyhow. The compressor is really nice and can go choppy or smooth without a hitch. The EQ works for broad strokes, which is what you want from a preamp anyways. You can sculpt later on, in the box.
I’ve tested many competing preamps, including the Millenia range, when opening new rooms and I always end up going for the Avalon in the end. They’ve got a lot of gain, too, which is useful if you have a broad range of mics. I think you can tell I like my Avalons 😛
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com -
You don’t have to separate them 10 feet apart. That’ll screw up any complicity, especially with a child. You can have them face each other, each with their own cardioid mic (the cardioid part is very important here) with the mic stands placed as close as possible. Even if they overlap, you can cross-edit, as the other talent’s voice is in the null of the cardioid pattern. What little bleed will be left at the top and bottom of a sentence can usually be hidden with well-placed fades. Don’t stop them from overlapping, as that is the best way to kill a vibe. Any session with children can be hard, and will work your psychology skills as well as your technical chops, and you want things as smooth as possible, with no technical concerns for the child. Otherwise, you’ve got a looooong day ahead of you.
Have a good time on this. When they work well, sessions with kids can be super-fun.
And please do ditch the Beta SP from the signal chain. You’ll be making yourself a huge favor in the long run.
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com -
Match your audio timecode format to the timecode format that will be on the picture by the time the project gets in the sound engineer’s hands. Mismatching makes EDLs undecypherable and future edits and re-conforming hell and will not make you any friends on the audio side, trust me.
As for bit-depth and sample rates, 24/48 should be your minimal goal. If you want HD audio to go with your HD picture, 24/96 is your minimum. Even if the final delivery format ends up being 16/48 (and that’s not HD in any way), give your sound guy a hand and record at a 24-bit depth. The increased dynamic range at the recording stage will give a lot more wiggle room to get decent level without clipping or falling into the noise floor. 16-bit can be great, but you need a very good sound recordist for that.
Good luck on the project.
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com -
That unit should do everything you want it to. As for the Behringer “reputation”, it shouldn’t affect you, as none of your audio will pass through the unit. It’ll have no impact on the final product. 200$ is pretty darn cheap for 8 motorized faders, so they are surely not that great. They should work fine nonetheless, just don’t fight them when they try moving. They probably won’t have a great feel, either. But that is to be expected from any company at that price point. Looks like a decent choice to me.
good luck in your projects,
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com -
That must’ve been lotsa fun, Ty.
At the studio here, we use Soundminer software to archive, database, audition and transfer sounds over the network. The system we use is total overkill for you, especially if this is a lone contract, but they do make other versions. I’m not sure if anything fits your bill but it may be worth checking them out at http://www.soundminer.com. Also don’t hesitate to drop them a line, they’ve always been nice and helpful with us.
good luck!
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com -
What you’re missing is not an amp, it’s the power supply. Since there’s a tube in the mic, it requires current that is different from what the Mackie, or any preamp, will feed it through phantom power. The power supply plugs into your wall socket and there’s a special cable between the power supply and mic. You plug your XLR going to the Mackie on the power supply. So without the power supply, you just can’t get to the Mackie because your mic has no power. You need both the Mackie and the power supply.
The other problem is that you can’t really be sure the 4060 works unless you have the power supply. Kind of a catch-22 there. You may be able to convince an Audio Technica dealer to let you test your mic with one of their power supplies, but that is assuming you have a nice neighbourhood audio shop, which is not given to everyone. Also, I don’t know where the cost of the power supply fits into your budget, but if you spent 800$ on a new mic, it would still probably be inferior to the AT4060.
good hunting!
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com -
Oh dear! There is no question in my mind you want to use that AT. The 4060 is a very nice cardioid condenser, exactly what you want for voice-overs. Just to help convince you, it’s a tube mic. That is some serious hardware you have lying around. In fact, I don’t know an enginner who wouldn’t reach for that before the MD421 which is considered a tom drum mic. You want to use that thing. Getting your hands on an AT8560 power supply should be your first priority.
Replacing the MD421 for a voice-over mic would get you into buying something similar yet inferior to the AT4060, so it’s kind of a moot point.
USB mics are dead-simple to use but none of the existing models are really really good mics. They tend to be entry-level mics. Most of them are way under the 300$ barrier, and in microphones, you tend to get what you pay for. If you didn’t have the Mackie desk, the Delta 44 and a 4060, it could be a way to go. It’d kind of like leaving the 65 Mustang with a bad radiator in the garage to buy a Ford Echo, in your wituation. The 4060 is worth almost 1300$ with power supply, to put things in perspective.
So find a power supply for the AT, hands down. That mic could be worth a better preamp than a Mackie desk some day, but if budget is of the essence, leave that for later.
If you can’t find a power supply or it’s too expensive, you could offload the 4060 and get extra dough for another mic. That would be too bad, though.
good luck on your trilemna. (is that even a word?)
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com -
The good old MO…
So, I’m assuming they’ll be doing this on a Dolby DMU CPU machine, which they surely will. The most basic thing they will need is the 5.1 audio, either as one 5.1 audio file or as 6 split tracks, named correctly with L,R,C,Ls,Rs,S tags in the file names. Once that’s loaded, it will be outputed through AES from either the desk or the ProTools rig. The DMU only has facilities for AES/EBU input, not for SDI as it isn’t an audio format per se.
There may be some mastering to do on the audio, and there should be one complete playback to measure the LEQ level. LEQ is Dolby’s way of measuring volume over time and if your product is to be viewed in a movie theater, there must be strict regulations on that, with a fixed maximum LEQ reading. You don’t want to have your mix refused because it’s too loud. If you have a few hours to burn and want to know every little detail about LEQ, go here: https://www.dolby.com/assets/pdf/tech_library/AES5900.pdf
Once that’s done, the actual encoding to MO is done in real-time. So that’s another playthrough. After that, the engineer may want to listen back to what he encoded on the MO. That can be done through spot-checking.
Basically, in 3 playthroughs it should be done. I do this at least once a week and it’s a pretty straightforward process, as long as the mix doesn’t need to be touched up. I’ve never seen anything go horribly wrong in this type of session either, so you’re pretty much in calm waters.
Have a good end of project pint when this is done!
JC Boulay
Audio Z
Montreal, Canada
http://www.audioz.com