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Old 11-12-2011, 07:11 PM
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Default Recording info on my latest CD

I've had a couple of requests for some details about my recording process on the CD I just released. I'm not real sure what would be useful, but I'll try to give a brain dump here and feel free to ask questions if I left something out you care about.

Basically, all guitars were recorded in my home studio, which is detailed here. The gear lineup was:

a pair of Brauner VM1s, spaced pairs, about 14 inches apart -> Great River MP2H -> Cranesong HEDD A/D converter -> RME AES-32 card, to Logic on my Mac

A stereo ribbon mic, AEA R88 -> AEA RPQ preamp -> MyTek Stereo 96 A/D -> RME AES-32 into Logic.

Monitoring chain was Logic-> AES-32-> Dangerous D monitor controller and D/A box, to Barefoot MM-27s with Dynaudio BM6As as secondary smaller monitors as a double check.

On a few tunes, I used Schoeps CMC6/MK41s instead of the Brauners, and there may be a tune or 2 where I used the Brauners+Schoeps, but in general, I settled in fairly early on the ribbon + Brauners as working well together and giving me a a range of tones to work with. The R88 was placed dead center in front of the guitar, raised a bit to avoid the soundhole, and set to M/S mode. So in mixing, I had a choice between either set of stereo pairs, both stereo pairs blended to any degree, or the R88 in mono as a center fill for the stereo Brauner pair.

The 1st few tunes would have been recorded with mics in the center of my room, but after a while I (and my wife) got tired of having to work around the pile of mics stands in the middle, and on her urging, I moved the mics to the corner of the room, so that I was facing the corner. I originally feared this would sound bad, but in fact the sound was, if anything, better. The downside was that my back was to the computer. Sometimes that was good - no distractions watching the meters as I played. Other times I totally missed that I hadn't armed a track or something - those were naturally all perfect takes that I failed to capture :-) But all in all, this is a nice setup. Combined with a FaderPort remote control, I can just sit down, hit record and play. Stop/rewind/do another take, all without leaving my chair. Mics are always up, so I can be recording in seconds. I did find the iPad AirDisplay to be a nice way to beam a window from my Mac to a music stand in front of me while I was doing any level or balance checks.

The actual recording took about 6 months of evenings and weekends, which could have been shorter if I'd not gotten side tracked or lazy a few times, and if I'd nailed down the tunes to include earlier. As it is, I recorded about 30 tunes, which I had to pair down to the 14 I released. I typically tried to record one tune in a session, and at least do a rough mix/edit, which might spill over into subsequent days. When you're doing everything yourself, it's easy to lose perspective after a while, so it helps to let a tune rest and come back to it.

For recording, I more or less followed a pattern I learned from recording Steve Baughman. Steve was quite rigorous - he'd record a half dozen takes all the way thru, then listen back with pen and paper in hand, and draw +'s and -'s on a timeline to indicate particularly good or bad spots in each take. Then we'd take the one that had the most +'s, and/or the one that had the best feel and use that as a base, and edit any bad spots. I don't have Steve's patience, but I generally did the same thing, multiple takes all the way thru, then address any small issues in the best take. Generally, I'd know after recording, which take "worked". Logic has a marvelous facility for editing multiple takes (most modern DAW's have some form of this), where each take is a new subtrack of the original. Assuming the takes line up, you simply select the the good region, and Logic compiles all the good parts into a single track. If you use a click track, this is amazingly easy. Without a click track, it's a little trickier because things don't line up, but can still be done. My goal, of course was always "no edits", if only because editing is a lot of work. I hit that on a few tunes, but most tunes had a couple of small edits, like maybe I preferred the ending off an otherwise lousy take over the ending on the best take.

Most tunes I recorded without a click track, but especially the ones I knew I was going to try to get someone to overdub, I played to a click - usually I used a drum track, it's just more fun and less stiff feeling.

Once I had a good take and had made whatever edits were needed, I would bounce the results to a new track, so I always have the original stuff there. With Logic's "Quick-Swipe Take Editing" you can even have multiple edits, all non-destructive. On the new bounce track, I usually spent a bit of time in Izotope RX doing a little cleanup. Very cool tool that lets me kill anything from string squeaks to the dog barking. The R88 tends to be a little noisy, so on some tracks I used broadband noise reduction to reduce that a bit.

Final mix then just consisted of choosing between my mic options, from all one or all the other, to anything in between. I think most tracks are mostly the Brauners with 50-75% R88 blended in. I rarely needed any EQ, tho I usually put a UAD Cambridge EQ plugin either on the tracks or the final bus with a sharp high pass filter, cutting out anything below around 30Hz. Just gets rid of that rumble from traffic, fridge and other noise makers. I also often use the tiniest touch of compression using the UAD LA2 plugin. If I can see the meters move, it's too much, but even that seems to just add a little beef to the sound. I generally used 2 reverbs, a Lexicon Native plugin Room reverb and one of the longer lexicon reverbs, Hall or Plate. For most tracks, I'd set these so I could just barely hear them, then back off. On a few tracks I ended up using a TC Electronic VSS3 reverb instead, or the UAD EMT 250 plugin. Just a matter of what felt good at the moment.

Other useful tools were Voxengo's MS-decoder (free VST plugin), a Waves S1, Izotope Ozone, just sort of an all in one EQ+compression+exciter+stereo imaging dodad, and BlueCat's visualization tools, especially the Stereoscope. It's a reasonable, if not quite perfect replacement for Adobe Audition's visualization tools that I lost when I moved from PC to Mac. It just helps verify the stereo balance.

Some other instruments were recorded elsewhere, partly due to proximity to the musicians, and also to take advantage of other engineer's expertise at recording percussion and violin, which I have little practice at. Laurel Thomsen's violin and viola were recorded with a Neumann U89, Brice Rice's percussion tracks were done with a Neumann KM184 pair. working with another studio was simply a matter of loading my tracks on a thumb drive and showing up at the other studios, coming back with their tracks on my thumb drive to load back into Logic. One studio used Pro Tools, the other Digital Performer - no problem, it's all just wav or aif files. I also had a few people come to my place for overdubs, and 2 tracks were done by mail, just send the guide track as an mp3, get the overdubbed part back by You Send It, and load into Logic. As long as you're careful to know where the track starts, it's easy to line everything up.

Everything was recorded at 96/24 and mixed the same way. I sent the files, electronically, on to Cass Anawaty at Sunbreak Music for mastering. He sent me a first draft, which I mostly liked, but suggested a little less compression. He also bounced one track back to me with some suggestions for small mix changes, and after hearing the master, I heard a few small issues that I hadn't noticed before in my mixes, and updated a few more tracks. He made a second pass, and then uploaded the results to discmakers for pressing. Cass mastered at 96/24, so I also ended up with the full HD tracks, which I released via bandcamp, which shows up as a "store" tab on my website. Cass's mastered tracks sound subtly different, tho it's hard to put a finger on what the difference is. The most obvious thing is that the levels are hotter, and the transitions between tunes are natural. Things also seem a tad "rounder", with a just generally smoother feel. My original mixes were done to the K-14 standard, using the UAD Precision Limiter as meters (limiter disengaged, just watching the meters). I also calibrated my monitors (See here and here for some details), which made it pretty easy to leave headroom, as full volume with these settings is *really* loud.

Artwork was done by discmakers,and was kind of last minute. I enlisted my wife to take a couple of photos sitting on a living room chair he night before I sent stuff into Discmakers, I grabbed a couple of nice guitar shots that Charles Webster had taken of my Claxton a while back, and sent them into discmakers and told them to have at it. I took some bits and pieces of what discmakers did for the CD package and created an online booklet for itunes, so there would be liner notes.

And that's it, other than backing up all my Logic projects to DVD in case I ever need to revisit them.

So, hmm, this was meant to be "brief" :-) Hope there's some useful info in there somewhere for recordists, feel free to ask about anything I left out.
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Last edited by Doug Young; 11-12-2011 at 08:19 PM.
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Old 11-12-2011, 08:35 PM
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Lots of info Doug. When recording I usually play a tune through a few times. The first time played is rarely the best performance as I relax into the music as I go.

So you got the Lexicon Native Reverb. I like that and I also like the Waves Renaissance Reverb using it with impulse responses from the EMT 252 and non linear sets.
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Old 11-12-2011, 08:55 PM
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Quote:
Originally Posted by rick-slo View Post
Lots of info Doug. When recording I usually play a tune through a few times. The first time played is rarely the best performance as I relax into the music as I go.

So you got the Lexicon Native Reverb. I like that and I also like the Waves Renaissance Reverb using it with impulse responses from the EMT 252 and non linear sets.
Yeah, I usually find that once I get a take that I think was good, it takes the pressure off and then I can do better. Doesn't matter if the take really was good - just that I believe it was :-) Sometimes it works the other way, tho. I think a few of my tunes were actually 1st takes. After that you start obsessing with remembering every little detail you want to put in, and it all just implodes on you.

I really like the Lexicon Native stuff. Really transparent, you can layer on tons of reverb and not sound like it. But the UAD EMT 250 is a cool reverb as well, not transparent at all, just rich and gooey, and the VSS-3 has some great sounds. For a while I was playing with the Bricasti impulses,and those are nice. It'd be great to actually have a Bricasti, but after I got the Lexicon set, I stopped dreaming about that one, at least for now. For a long time I was using my TC M4000 hardware device, but I put that in my live rack, so it's too much hassle to reconnect it for recording.
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Old 11-12-2011, 09:05 PM
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Originally Posted by Doug Young View Post
.

But the UAD EMT 250 is a cool reverb as well, not transparent at all, just rich and gooey
Very rich IRs. With these I use less than 1% wet to dry, not the 10% to 15% I use with the Lexicon Native.
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Last edited by rick-slo; 11-13-2011 at 09:17 AM.
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Old 11-12-2011, 10:01 PM
corbetta corbetta is offline
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Way cool, Doug.

I am a big fan of bigger/longer takes. For the kind of stuff I do, edits are unavoidable, but bigger takes yield a much better musical feel in my experience.

As I mentioned, I will be spinning some tracks off your new record Monday night on Guitar Journeys.
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Old 11-13-2011, 01:34 PM
alohachris alohachris is offline
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Default PM'ed Ya Doug

PM'ed Ya Doug,

alohachris
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Old 11-13-2011, 07:26 PM
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I was asked offline about the faderport, which is a useful gizmo when you're recording yourself. It's just a little box that connects to the computer via USB and can control various functions of your recorder, so you don't have to get up. Ultimate in laziness :-) Actually, it's nice to not have to move once you get set up between takes, keeps things more consistent. There's no audio going thru the faderport, it just sends control messages to the DAW, and it works with many (most?) systems. It has basic transport controls as well as a slider for volume, pan control, and so on. All I ever use is the record/stop/rewind buttons. I attach it to my mic stand with an OnStage boom adaptor:



I don't recall what the mount on this was, I think it was supposed to just sit on a table, which is pretty useless for most needs, so I made a little holder for mine. Just used a mic clip extender and attached it to an electrical outlet cover from Home Depo, which is glued (or bolted? I don't recall) to a piece of wood. Then velcro on the wood and the faderport holds it in place.



There are other gizmos like this, the Apogee GIO is one I've seen but not tried, and there are are several ipad/iphone apps that will act as remote controllers, but I don't want to be having to chase my wife down to re-commandeer my ipad when I want to record.
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Old 11-15-2011, 09:13 AM
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Interesting read. Do you find a significant difference between 96k and 48k? I've kind of dismissed 96k as overkill but maybe I just don't have a good enough recording/monitoring chain to hear the difference.
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Old 11-15-2011, 09:55 AM
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Interesting read. Do you find a significant difference between 96k and 48k? I've kind of dismissed 96k as overkill but maybe I just don't have a good enough recording/monitoring chain to hear the difference.
I've never tried a blind test. I rather doubt I'd reliably be able to tell. I use 96 because there's no reason not, and some people feel they can hear the difference.
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Old 11-15-2011, 10:29 AM
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I usually have recorded at 88,200 mainly because I felt it would allow better post recording processing, such as compression and reverb. Now I feel that the quality of the filter algorithms for upsampling and downsampling in the software I am using is so high that recording in 88,200 is optional. I have been recording more with 44,100 lately and can not hear a difference.
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Old 11-15-2011, 11:01 AM
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Re: "overkill" This kind of gets into a philosophical area. I try hard not to believe in fairy dust. With an engineering background, I tend to be somewhat practical. It's easy to demonstrate that there are minimal (often inaudible) differences in mics, preamps, guitars, sample rates, etc, etc. But I still prefer to go for overkill to the extent I can. I don't want to come away thinking "I played great, my gear just didn't capture it well". Sort of an excuse way of thinking.

I'd rather use the best stuff and best settings I have access to, and basically know if it doesn't sound good, there's only one possible place to point the finger. Either I did a bad job of engineering or I did a bad job of playing. Either way, the problem is me, and I know what I have to do. I won't sound better once I buy that new mic, or once I raise the sample rate, or... I'll sound better after I practice :-)
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Old 11-22-2011, 11:54 AM
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Thanks for the write up Doug. Very informative to see your work flow. I think it was all worth it, your cd is a big step from your first album, and sounds great.


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Old 11-22-2011, 03:24 PM
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Interesting read..thanks.

With regard to listening levels and calibration, it amazes me how many people that actually work in the industry ( I work in Film and TV...http://www.imdb.com/name/nm1427982/ ) have no idea of the importance of proper system calibration.

Thanks again for sharing your info.


btw..

In one of your links it refers to Film as having a 83db SPL reference standard and I will quote this line.....

"To calibrate a monitor to the film-standard, play a standardized pink noise calibration signal whose amplitude is -20 dB FS RMS, on one channel (loudspeaker) at a time. Adjust the monitor gain to yield 83 dB SPL using a meter with C-weighted, slow response. Call this gain 0 dB, the reference, and you will find the pop-music "standard" monitor gain at 6 dB below this reference."

This is not true for 5.1 calibration .....


Dolbys' current film standard is 85db SPL .

When calibrating our systems ( 1khz @ -20db = 0VU ) I run Dolby pink noise @ -20 and calibrate each front speaker ( Left Center and Right ) to 85db rms/c-weighted.
The Ls / Rs ( surround speakers in a 5.1 setup ) to 83db and the LFE to 89db

This has been true for every mix I have been involved with, and all Printmasters for these mixes were done by myself and a Dolby Labs rep as we record the film sound ( 5.1 and LtRt ) to Dolbys' MO recorder. From there the MO creates both the Dolby Digital and Stereo Optical ( LtRt ) tracks that exist on the final film release.
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Old 11-23-2011, 08:21 AM
nik_c nik_c is offline
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You have a avery nice set-up and some very nice mics at hand! I wish I had some of simialr stuff
The resason why you found the sound very nice when you moved the mics into the corner might have been because that way you were catching some nice early reflections?

nik C
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Old 11-23-2011, 12:26 PM
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Quote:
Originally Posted by nik_c View Post
You have a avery nice set-up and some very nice mics at hand! I wish I had some of simialr stuff
The resason why you found the sound very nice when you moved the mics into the corner might have been because that way you were catching some nice early reflections?

nik C
Possible, tho that corner is pretty much filled with fiberglass, so it's quite dead. Corners can be problematic, they're often a place where bass builds up, and then there's the reflections that can happen. But you just never know what will sound good, so trying different locations when your experimenting with recording is a good idea.
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