Me and My Routing, FWIW

TascMan

Soundaholic
Joined
Oct 17, 2012
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From
Lancaster, PA USA
Gear owned
DM3200 UH-7000 3030R2R
Hey there everyone.
I have been on this board since its inception and on our old one for years before that. I am back here almost everyday, but can't always post. I am a 43yo corporate jet pilot for my main job, but I record at home as a hobby, as well as do live sound for friends. I've been doing both since high school. Everything I know about recording and live sound is from years of reading, working with others, and trial and error. That probably trashes my credibility, but so be it. I also play drums (since I was 12), and have been involved with several bands over the years, as well as music programs in high school and college. I was a music Ed. major with a percussion minor for 3 years at U of MD, before changing to English and graduating in Dec of '93, then going to pilot school. I am gone often, so I don't constantly have bands coming in and out like many of you. I have a few regulars, mostly inexperienced, as well as tape/vinyl transfers and I just go with that. Since I have 3 kids, my time is also often tied up with them. Here is an album I just finished for a Christian artist named Lana Witmer;- http://www.cdbaby.com/cd/lanawitmer -. I am currently working with a punk sound based artist. A whole different thing from Lana's stuff!

I have learned so much from these forums, and, as it has been said so many times before, this is definitely the best forum I have been on. The users/posters are both cordial and intelligent which I came to really appreciate over the years. We get very few inquiries like "....I just bought a DM....how do I record my band and get signed?" or " ....I just bought a 5000 dollar mic. So, what does "MIC" stand for again?" Which are pretty common questions on other pro audio boards. So, Cheers to TascamForums, its members and a special thanks to the moderators. I just wanted to say that. I really enjoy being part of it.

I have been asked how I have my rig setup and routed, so I thought I would share. It's not very impressive, but it may help others somehow. I bought my DM3200 back in Feb 2008, used, from Palace Studios, over EBay. Come to think of it, almost everything I have is from EBay. Or Parts express. I almost bought a large analog board (I had been using my Mackie Onyx 1640 previously), but I wanted to keep things compact, and still have all of the flexibility of digital. You probably bought your DM for the same reasons.

My "home Studio" is set up like this....I have a full size desk from Dash Studio Furniture that houses my 3200, with effect racks on both sides. For an OLD picture, see here: http://www.dashrecordingstudiofurniture.com/DC.html Mine is the 3200 obviously. My setup has changed quite a bit since then, but the desk hasn't. All that stuff is in what would be our dinning room, on the first floor of my house. Below that, in our basement, I built a soundproof, and pretty dead, 12ft X 16ft tracking room, where I house my drums, guitar amps, and a ton of "impulse buy" mics. I have a 24 channel snake in there and I run all mics through that, up to my board in the dining room. I have two patch bays, some outboard pres and effects, and a wonderful digital patch bay that I love, the midiman DigiPatch 12x6.

For the most part, I mix in a hybrid way. I typically track one or two tracks at a time and layer instruments and voices to create a raw mix. I may have the drummer start, playing my drum kit to a metronome sent out from Reaper, or a scratch track. I record my drums directly into the MIC inputs in the DM, 1=kick, 2=snare top, 3=snare bottom, 4=HiHats, 5-8 are my four toms, 9-10 are my overheads, set left and right, 11 is usually a mono room mic, I might add another room mic on input 12. For no particular reason, input 13 is usually a TB mic for the player, 14 and 15 are usually for instruments, like guitar or bass and I usually record a DI track from the guitars, so I can add in a different guitar sound later, if necessary, via software. Ch 16 is my own TalkBack mic, a 58, which I turn on or off at my board via the footswitch input. Sorry Tascam, I can't understand why the built in talkback mic is at the top of the board. Since my snake is 24 channels, I use snake CHs 16-24 for various separate preamps for vocals, or acoustic. I am not a very big MIDI user as you can see. Although, I keep a Roland XV-5050 in my rack and have some VSTis if I need them. (addictive drums, and some synth stuff) I also keep a novation ReMote 49SL compact keyboard controller up here just in case I need it.

I have been using Reaper for some time now, which I still like, and AA3 for wave editing and creating CDs. Let me stop here and say, YES, I know the more expensive software is better, with all the included effects, but I have this and enjoy using it. I've always been very close to picking up Cubase, I just haven't yet, and reaper is very easy to use with flexible routing, and is 64bit point to point at any sample rate. OK, that said, as for my routing, I usually send each input directly to a corresponding FW channel, using Input Bypass in the routing screen, to a corresponding track in Reaper, and I also assign each input to a channel in the DM for monitoring out through the stereo buss. This allows me to record each track raw and unaffected into reaper, while simultaneously mixing for a headphone/monitor cue . I do not record out from the Buss channels, mainly because I mix most of the stuff ITB in Reaper. Once I have the waveforms worked out in reaper, with initial comps and EQ, and maybe some reverb as well, I can buss those tracks within Reaper, and send those bussed tracks back into the DM as a few stereo tracks. Similar to Stem tracks. For example, I like to mix my acoustic drum tracks to a 2 track stem and send that to paired CHS 1-2 (using "INVERT" and RTN) in the DM. Then I do the same thing for guitars, vocals, etc., adding more comp/limit and EQ and maybe some effects from the DM. I then mix those OTB in the DM and send them out the STEREO buss, back into a stereo track in Reaper. When I am happy with it, I put it up in AA3 and make any last revisions, like final volume leveling, and put it on a CD list where it is converted to 44.1/16bit and burned, or saved to a file as a .wav file, and/or MP3.

I have my DM layers as follows...1-16 are set to 1-16 mic/line inputs, normally for tracking. For OTB mixing, I "invert" 1-16 and have the returns from the FW set for my RTN inputs for each channel. I haven't needed more than these 16, since I am using stems form Reaper, but I do the same for channels 17-32 on that layer, just because I can. Normally for tracking, I have 17-24 coming in from my upper patchbay. These may include preamp outputs (2 CHs FMR-RNP, 4 CHs Sytek MPX-4A, 1 CH ART PRO Channel, and 2 CHs ART PRO MPAII), the send from my Tascam 3030 R2R, the send from a Motorola Bluetooth receiver or other stuff, ie a SONY cassette deck, AKAI R2R, Audio-TechnicaTurntable.... CH layer 33-46 is reserved for effect returns, DMs Digital returns, 8 channels of ADAT or 8 channels of TDIF from an old but still good MOTU 2408. Also, I REALLY like the setup where you can send any channel's outs to the onboard effects via two stereo auxes (using the encoder knobs) and hav them return to two channels on the board. That little thing is VERY helpful, especially for a headphone mix with some reverb. Its in the forum sticky stuff if you are interested.

I record everything at 88.2. I don't have a very good reason other than it sounds fine and the file size is not too big and the conversion back to 44.1 doesn't mess with the wave file because I am just halving the sample rate (is my belief). I used to clock off of a Lucid GenX 192, which is still in my rig, but is used more for distributing the Dm's clock. Short explanation was that I bought into the idea of needing to clock everything externally. But when I really listened to my stuff clocked externally or internally, I found no difference. I have since bought into the idea that the piece doing the recording should be the master clock for initial conversion, ie the DM, which has a fine clock circuit. I distribute that into Berhinger ultramatch PRO, (don't hurt me) to cut the sample rate down to 44.1, (since you can't do this in the DMs outs) which then goes to a TC Helicon Voice works, a lexicon MX200, a TC Electronic C300 digital comp (which I have yet to use), and the MOTU 2408. Those outputs are then routed into the midiman patch bay, and back to the DM's digital inputs where they are resampled back to 88.2.

That's my setup in a nutshell. Now my fingers hurt from typing and lunch is on the table. I hope this might help someone else. Post any questions and I will try to reply. Take care,
Steve
 
Steve you're a maniac my brother! lol
Great info!
 
Thanks Charlie, I guess there's such a thing as too much coffee...
 
Wow, now there's a full report! A good read nonetheless, thanks for sharing.
 
Thanks Arjan. I've enjoyed your posts over the years.
 
TascMan said:
Thanks Arjan. I've enjoyed your posts over the years.

And I enjoy both of yours. ;)

Great description of your rig and workflow, Steve. You've got the entire scheme nailed and well thought out. Will listen to your album when I get down to the studio to do it justice.

I use AA for 2 mix editing too. I don't find it in any way 'limiting,' though I prefer to use Ozone5 for mastering tasks.

PS: Interestingly, you're not the only pilot on this forum.

CaptDan
 
Thanks Dan, your great contributions are too numerous to list. I am interested in who the other recording pilots are out there. PM me....
 
Hey Steve...

How cool that you would post this info.
It is much appreciated.

I'm very interested in your choice of 88.2kHz.

I keep thinking I should record at a higher rate.
I'm still doing stuff at 44.1.

A question for you and others on the forum...
Do you hear an improvement?
Does 88.2 slow down the DM?
Does it slow down your DAW?
 
Glad to write it and glad you appreciated it. Thanks

Yes, I can hear a difference between 44.1 and 88.2, especially in very transient and high pitched signals, like the picking on an acoustic guitar and how my cymbals sound. It's not completely obvious but I hear it nonetheless. I picked up a pair of new monitors, Yamaha HS80Ms, and can really hear the difference on them. Huge plug for the yamies BTW.

As for the choice of 88.2, I guess I subscribe to the theory that doing all recording and mixing in a higher resolution somehow results in a better final, lower resolution product. I used to record to tape, and everything involved with that was based around keeping your noise floor as low as possible while printing as hot as possible. The equivalent in digital is more about keeping everything at 24bits or higher, rather than recording at 16bits. Having said that, I know that I cannot hear anywhere near the top frequencies of even 44.1, much less 88.2, but I know that the A/D conversion of the analog waveform of an 88.2 recording is going to be more accurate than that of a 44.1, by twice as much. And the down-sampling back to 44.1 is less destructive than coming down from 96K, or so I have read, because you are only cutting out every other sample. That's the theory I have developed over time so I do it more for those reasons. If CDs were at 48Khz, I would probably toy with 96K for the same reasons. It absolutely does NOT slow down the DM. The mixer could care less that it is converting at 88.2.

So, I did some trials and have found that there is no strain that I can see on my computer, be it processing or hard drive throughput. There is nothing slowing down my DAW, even with over 30 tracks or more. One big reason for that is because Reaper is only a little over 6MB. A tiny fraction of most DAWs. Now, I have seen the occasional slowdown when I put several hungry VSTs on all the tracks. The SoundToys stuff is a good example. So, I either find a different way of doing the same thing or I can "Freeze" those tracks (sort of like temporarily and destructively applying the effects on that track in place) or I can render them to a stem.

If it helps, I have a windows 7 machine, 3 500GB WD hard drives, all SATAII, on an ASUS P5Bdeluxe MB, 4 GB of DDR2 Ram, and an intel Q6600 processor. That was the shiz-digity in 2008, when I built it but it's old stuff in todays computer hardware world, and I will be upgrading soon to a Haswell processor, but it all still works for me.

Something that really worked for me was not only upgrading my hard drives to SATAII, but also separating important files. I.E., one hard drive has my OS and programs, another hard drive has all my audio files, projects, music etc., and the third hard drive has all my VSTs and anything extra. When recording, Reaper is in ram, windows data is swapped to and from the one OS drive while audio is being pulled from another at the same time. I think this coordinates the data flow much better than having only one drive, no matter how big or fast it is.
 
TascMan said:
Thanks Dan, your great contributions are too numerous to list. I am interested in who the other recording pilots are out there. PM me....

PM sent.

Thanks for the kind words; I'm always learning something new with my setup, and a large part of that is due to the above average intelligence and creativity these mixers seem to attract to this forum. (I mean, ya gotta have something going on in the brain cavity to work these beasts, eh? :) )

Re: 88.2/96kz: I'm careful to parse my words about higher S/Rs because I've been brow beaten by 'experts' who've insisted it's virtually impossible to discern anything positive recording higher than the 'intended release format' - unless one is a canine. :eek: This argument is based on the unasalable viewpoint that an adult human is fortunate to hear frequencies above 13kz (or, perhaps - 10kz if subjected to massive amounts of Marshall Stack assault.) Because 44.1kz is <>44.1 kz capable, and the 'best mics don't deliver much above 17kz,' it's a slam dunk. Case closed.

Or is it? I can certainly hear increased fidelity on original tracks and un-effected mixes - both 88.2 and 96kz. Bass frequencies are more delineated, transients are cleaner; stereo imaging - more accurate. Certain efx - verbs/delays - seem to perform better in a spatial sense as well. The question is whether these aspects translate when downsampled to 44.1kz Redbook or 128kbs MP3. I'll stick my neck out and allege that they do.

When downsampling/mastering DSP is applied to an 88.2 or 96kz mix, - ie: brickwall limiting, etc - the 'artifact zone' which resides atop the audible range is shifted upwards into inaudible territory. This, alone, can't do anything but help the final result. The degree to which this applies, perhaps, is program material dependent; in otherwords, a denser mix is more likely to be affected by artifacts than - say - a vocal and nylon guitar. Still, I believe what's good for the audio 'goose,' is good for the gander.

That said, I've found with my PC (2009 vintage Intel 6300/8gig ram), Protools 10 - I can't track at 96kz while applying certain processing hungry DSP - in particular - PT's time-morphing algorithm 'Elastic Audio.' After 3 overdub passes with EA resident, I experience dropouts, glitches, and the sonic equivilent of regurgitation.
The solution: don't use it, or - if it's required - save it for the final mix and use a larger Firewire buffer.

Final caveat: a well tracked/mixed 44.1kz example will always sound better than a crappy 88.2/96kz recording. :)

YMMV.

CaptDan
 
"Final caveat: a well tracked/mixed 44.1kz example will always sound better than a crappy 88.2/96kz recording" Absolutely True!!

And you make a good point about not being able to hear above13K. (jet engines at 4-6K have effected my hearing!) Something a bit geeky to relate though is that sampling at 44.1K actually results in a highest frequency of 22.5K. If you think of one cycle of a 22Khz sign wave, a 44.1Khz sample has to reproduce the pos side of the sine wave and the negative side of the sine wave to be able to completely reproduce that sine wave. So, the highest waveform frequency any sample rate can reproduce is half of that sample rate. 88.2 is capable of a top frequency of a 44.1KHz waveform for the same reason. Useful information on a chalkboard...not important in the real world. But again, I use 88.2 for the more accurate A/D waveform reproduction. The point about down sampling and moving artifacts to an inaudible zone is understood. If I was recording a very delicate, soft and highly transient nylon guitar project, I wood probably do it in 96Khz. Until then, I'm still happy at 88.2 for everything else.
 
Nice recordings Steve. What mic did you use on her voice and the acoustic?
 
Thanks Charlie! You might like the stuff I am working on now a little better. I'll have to get some raw tracks out to you to listen too. In the vein of punk metal with an artsy twist.

Voice was through a CAD M9 (tube -Tung Sol 12ax7- condenser) into an ART Pro Channel. The rest of her chain was comprised of various VSTs. I.e.; desser, EQ, comp, etc.
Acoustic was miced with a pair of CAD M77s, (I use these all the time for various things) into an ART PRO MPAII. I used a setup on the first few recordings we did, (last few on the album) which had the mics set as a spaced pair, roughly at the bridge and 12th fret. I later decided to go with one mic at the 12th fret and the other over her right shoulder. I like that setup better, but it changed the acoustic's sound consistency on the album. Oh, well.
 
Steve are all the drums real?
 
Yes, but I also mixed in some addictive drums, midi triggered from the actual drum's spike above a gate. It's something available in reaper's gate vst.
 
Really great info and thread everyone! It always helps to hear how others do their routing, etc. I have been on hiatus getting things ready to build a stand alone studio so I will probably incorporate some of his new info into the new control room setup when it's ready. Thanks again!
 

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