Topic: TASCAM DR-680 portable 8-track #6

[EarlyMorningRain]

FWIW, I've mentioned this before, but if you are recording something with mics that is quiet, it's probably better to set the mic gain to HIGH and turn the trim down than to set the gain to LOW and turn the trim up. That is, if you have the trim set to 20 or 25 with the gain set to LOW, change the gain to HIGH and set the trim at -5 or 0. There will be a lot less noise that way. I'm not sure why the instruction manual says the opposite.

Hadn't thought of this at all, but awesome info, thanks!!

[Gutbucket]

Can I link those channels in advance, before the show, and have them still be linked when I go to make adjustments after the show starts?  Or are they only linked until I exit the Rec Trim menu?  Do they stay linked through a power off/on sequence and have to be manually unlinked?

You can have it remember which channels are automatically record armed when you press the record button by setting that in the menu.  For example, that way you can have it automatically arm channels 1 trough 4 and not 5-6 and the stereo channel when you push record.  It does not remember channel ganging, but they'll stay ganged until you either ungang them or the machine is powered down.  Ganged channels must be contiguous (you can't gang chs 1,2 and 4 but not ch 3), but you can make more than one ganged set if you want (say , ch 1&2, and seperately ch 3 through 5 for two seperately ganged sets).

[mattmiller]

This isn't specifically a DR-680 question, but a 4+ track recorder question in general.  Obviously, not having to deal with the clocks of two recorders saves a bunch of time in post, but do any of you apply any sort of delay to align the SBD and MIC sources better?  In a small club, I assume this would be more of a theoretical calculation and application rather than an audible arrival-time difference between the signals.  Worry about it and correct it, or no?

[fsulloway]

This isn't specifically a DR-680 question, but a 4+ track recorder question in general.  Obviously, not having to deal with the clocks of two recorders saves a bunch of time in post, but do any of you apply any sort of delay to align the SBD and MIC sources better?  In a small club, I assume this would be more of a theoretical calculation and application rather than an audible arrival-time difference between the signals.  Worry about it and correct it, or no?

What do you consider a small club? It's about distance really. When I run onstage and SBD, I don't add any delay to the SBD. Once I'm moving the mics farther away from the stage some delay may be needed, again, depends on the distance. Once you load both sources into your software it's easy to move the SBD around so that you can listen to it and judge for yourself. I usually zoom in and set it by eye then close my eyes and listen for the sweet spot as I bump the sbd by a couple ms either way.

[mattmiller]

What do you consider a small club?

I'll be set up about 25-30 feet from the speakers.  My ears aren't good enough to hear an offset of 1-2 ms, but was curious if others could hear it or otherwise adjusted it based on a calculation.

[fsulloway]

you may not be able to say you hear the delay but you might think that bumping the sbd a couple ms makes it sound a little better. I really do close my eyes so as not to be held by any certain formulas of ms/ft. play around with it next time and see if you think it makes any difference.

[lukpac]

you may not be able to say you hear the delay but you might think that bumping the sbd a couple ms makes it sound a little better. I really do close my eyes so as not to be held by any certain formulas of ms/ft. play around with it next time and see if you think it makes any difference.

This.

I wouldn't rely on any sort of formula, and just see what sounds good. Too much of a delay is (usually) going to sound bad, but having the two right on top of each other often leads to bad phase artifacts.

Play around with the relative levels of the mics and soundboard as well. I'll favor one or the other depending on the sound and balance of each.

[Gutbucket]

lucpac and fsulloway are spot on.  Align visually if necessary to get things close, then like most everything else, tweak it by ear until it sounds best (& without looking often helps to stay free of visual bias).  Listening to stage banter between songs is often the best way to hear it clearly and adjust the delay accurately.  You'll probably be able to hear a several msec delay easily with solo stage talk.  Listening for isolated sharp transients like drum hits can also work, but they can get more confusing, and once the song is in wall-of-sound mode it's much more difficult to hear fine delay adjustment changes until everything snaps in.

I don't use or would suggest using a delay formula unless you had to mix it something on the fly without monitoring, but it's good to have a general idea of the timing involved to know when distance delay might be a factor, how significant it might be, and what the delay will sound like.  A good approximate rule of thumb is 1' of distance between sources = 1 msec of time offset between them.  If you play around with a digital delay for even a short while, you'll get a good feeling for what various delay times sound like and what to expect from a several msec delay vs a ten msec one vs 20 msec one or something longer.

From short <1msec to long >30 msec delays I hear a progression something like this as the delay time is increased:

slight panning or 'thickening'
chorus like or pseudo stereo effects
rapid slapback type delay
discernable echo

[Gutbucket]

One other thing I’ll add in case you want some technical stuff to flesh out your blind listening time shifting- People talk about ‘phase issues’ and ‘phasyness’ in a negative sense, but the only time there is no phase interaction between multiple channels which contain somewhat common signals is if all the microphones were coincidently located.  All near or far spaced microphone techniques which are not coincident X/Y setups (M/S is X/Y) introduce phase interaction between channels, which is intentional and by design.

The most significant question is always, “does it sound good or not”.  If it doesn’t sound good people tend to refer to “phasing” as a problem, but if it sounds good they usually don’t say “that phasing sounds really great”.

The phase interaction between non-coincident mic channels varies by frequency in direct relationship to distance as determined by the wavelength of the frequency in question and it’s relation to the delay time given the speed of sound.  1 foot of distance approximately = 1 msec of time at the speed of sound, yet 1 ms only = approximately one wavelength at 1kHz.  It will = many wavelengths at higher frequencies, and only a very small fraction of a single wavelength at the lowest frequencies.  So very short delays will cause audible phase interaction in the higher frequencies, while lower frequencies still have wavelengths far too long to hear a minor difference in their phase.  Still short but slightly longer delays shift those significant interactions lower across the frequency spectrum.  As some point the phase shift at higher frequencies becomes so large as to be essentially random and not significant in terms of hearing the phase interaction (multiple 100's of degrees of phase shift), while canceling / reinforcing is happening at lower frequencies with longer wavelengths.  That is all happening at shorter end of the delay time spectrum, where the “thickening, chorusing, and pseudo-stereo stuff is heard.  Once you get into the echo effects region, you are well outside the phase-interaction range for even the lowest frequencies.

[Gutbucket]

Play around with the relative levels of the mics and soundboard as well. I'll favor one or the other depending on the sound and balance of each.

Heh, a personal peeve I can't resist.. certainly not directed at anyone here specifically.
If you need to rely too heavily on the SBD, your AUD sucks.  Work on getting a better AUD. 
A SBD recording is a very welcome tool for us, but it is essentially a crutch for a bum leg. 
The world doesn't need more over-SBD heavy tapes, it needs more great AUDs 

The comment above reflects the opinion of the poster alone, and not those of this site, the band, or the unapreciative tin-eared hoards who delcare your AUD tape to be "as good as a soundboard"

[lukpac]

Heh, a personal peeve I can't resist.. certainly not directed at anyone here specifically.
If you need to rely too heavily on the SBD, your AUD sucks.  Work on getting a better AUD. 
A SBD recording is a very welcome tool for us, but it is essentially a crutch for a bum leg. 
The world doesn't need more over-SBD heavy tapes, it needs more great AUDs 

The comment above reflects the opinion of the poster alone, and not those of this site, the band, or the unapreciative tin-eared hoards who delcare your AUD tape to be "as good as a soundboard"

IMO it depends on the sound in the venue. If the house sound sucks but the SBD feed is great, I'm not going to rely on the audience recording "just because". And even with a nice sounding room sound, vocals in particular almost always sound better with some direct feed from the mics/board. Any way around it, a vocal into a mic sounds better than a vocal into a mic into a speaker into a mic.

[Gutbucket]

Whatever sounds best.

Which is yet another matter of subjective opinion of course, but is at least a broad statement most people will agree with, which makes it somewhat more objective, even if we don't all agree on what 'best' is.

vocals in particular almost always sound better with some direct feed

Good point that vocals benefit most and most often from mixing in a SBD recording. There are exceptions though.  Clear vocals are usually the primary reason I record the SBD when I can.  I still wholeheartedly believe in that intentionally overly opinionated statement in my last post. Yet much depends on things outside of our control and a SBD recording is always a good hedge bet.  It's ultimately just another tool.  Use whatever you need to make it sound best. 

I'm just throwing out a call to resist the easy cop-out of over-heavy SBD matrices.

Thanks for putting the almost in there. 

[mattmiller]

Thanks for the tips.  Everything went great last night.  I had some nerves about powering the unit with my Tekkeon 3300, but locking the voltage to 12v proved to do no harm.  Hopefully my luck continues, since operating the deck is much more straightforward than I thought it would be, and I'll probably use it more in the future.

Regarding using caution when mixing in SBD -- I'm aware of pros and cons and use it only as much as I think is needed.  I've been doing matrix recordings for a while, but with two decks.  This was just the first time I've used the 680 for it.  The delay was much more than I was expecting.  I was thinking it was something like 1 ms per 10 feet, which was why I was thinking it might be negligible (to my ears).  But it ended up being about 28 ms, which agrees nicely with the 1 ms per foot rule, since I've always estimated the distance to my recording position at the venue as being about 30 feet.

[Gutbucket]

Good to hear!

I've been doing matrix recordings for a while, but with two decks.  This was just the first time I've used the 680 for it.

You're going to love having all your tracks recorded on a single machine in sync rather than using two seperate recorders.

[Phil Zone]

On the 680 I can't figure out how to lock all the buttons not the face of the unit, being the value knob and the ones by the screen. Can it be locked somehow?