Taperssection.com
Gear / Technical Help => Recording Gear => Topic started by: kirk97132 on October 19, 2013, 10:21:14 AM
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Just cranking up a new thread as the old one was long in the tooth......
part 1: http://taperssection.com/index.php?topic=131128.0
part 2: http://taperssection.com/index.php?topic=137231.360
part 3: http://taperssection.com/index.php?topic=141730.0
part 4: http://taperssection.com/index.php?topic=147677.0
part 5: http://taperssection.com/index.php?topic=157462.0
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And to bump it to the top...
Gut I'll post what ever I find out, hopefully later this afternoon
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Gutbucket, and all........well my ganging function is a real :facepalm: :facepalm: :facepalm:
Through sheer coincidence i have been try two opposite things on my two 680's. When I couldn't get the ganging feature to work at a show, the next time I set my deck to stereo and the ganging worked.......
Well of course it did, deck-1 only has V1.11 firmware, and deck-2 has V1.20. It seems I kept trying to get the old firmware to do ganging one deck -1 which it wont. And kept getting the ganging to work on deck-2 which of course is part of the firmware upgrade. I don't know how I kept managing to try the same thing out on the same deck each time. that solves the big mystery and it has nothing to do with limiters or anything else except operator error :zombie03:
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Good hear you both figured out the issue and that no problem with the machine.
Gang on.
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So, I was wondering if anyone else experienced this situation? Just before recorded a matrix last Sunday, Channel 1 and 2 disappeared from my screen. In the past, I would go to the card, hit information, and it seemd to reboot the channels. This time, it did not. So I changed the inputs to channels 3-6 xlr ports and got a signal thorughout the set. When I tried opening the files, it said the files were corrupted and the spaced used read as "zero."
Glad I got a warranty. Any idea what happened here?
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Channels disappeared?
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So, I was wondering if anyone else experienced this situation? Just before recorded a matrix last Sunday, Channel 1 and 2 disappeared from my screen. In the past, I would go to the card, hit information, and it seemd to reboot the channels. This time, it did not. So I changed the inputs to channels 3-6 xlr ports and got a signal thorughout the set. When I tried opening the files, it said the files were corrupted and the spaced used read as "zero."
Glad I got a warranty. Any idea what happened here?
Has this happened with more than one SD card? It sounds like it could be a card issue, but that is just a supposition. I have never had any channels disappear from my screen, so that in itself sounds like a red flag, but still possibly a card reading issue?
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Tried a different card and still the same. Here's a pic
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What happens when you try to arm the channels? Nothing?
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yes, nothing :facepalm:
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But you can arm and disarm the other channels without problems?
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I can arm and disarm channels 3-6, but it did not record the other night after the channels disappeared. The files say corrupted files.
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Hmm. Do channels 1,2 work if you put the recorder in stereo mode (selected with the button on top most people never use)?
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Have you tried initializing the unit? [page 46 of the manual]
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wow i gotta get one of these puppies........... ::) ::)
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gutbucket, I was in stereo mode and switched to 6 channel. The channels returned ;D Very wierd. I'm gonna initialize the unit as a precaution to prevent reoccurence.
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Good idea to also check and confirm the firmware is up to date.
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I checked and I have version 1.2. Glad I still have a warranty though. :o
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just reportin' in! Thought I had in previous threads however I wasn't getting any alerts
???
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just reportin' in! Thought I had in previous threads however I wasn't getting any alerts
???
FYI, you don't need to post to get alerts. Just click the "Notify" button.
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hmm, well it shows as being notified as that button says unnotify now
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I've owned my DR-680 for over a year and am only now playing with it for the first time and considering using it tomorrow night at a show. I've been playing around with it for an hour or so, connected my mics to channels 1/2 and connected the outputs of my HD-P2 to channels 3/4 to play the role of my SBD feed. I think I've got most of it figured out, but there are a couple things I can't figure out, and I'm short on time for navigating all of the threads. Apologies in advance if my questions are discussed elsewhere.
1. I'm using a 16 GB card. During my tests, all the tracks were armed for recording (by default), and I hit the "display" button to check the available recording time. It reported 2:04:09. That's not enough time, so I assumed I just needed to tell it not to record channels 5/6 and the mix. I unarmed those tracks, started recording again, hit "display", and, again, it reported 2:04:09. My calculations suggest I should be getting much more time than that, so what's going on? The card is formatted.
2. To adjust the record levels, it makes me adjust each channel of each stereo pair individually. I seem to recall discussion of these being able to be linked somehow, but a quick search turned up nothing, presumably because I'm not recalling the name of the function and am not searching correctly.
These are the only problems I can find right now. Thanks in advance for any guidance.
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1. I'm using a 16 GB card. During my tests, all the tracks were armed for recording (by default), and I hit the "display" button to check the available recording time. It reported 2:04:09. That's not enough time, so I assumed I just needed to tell it not to record channels 5/6 and the mix. I unarmed those tracks, started recording again, hit "display", and, again, it reported 2:04:09. My calculations suggest I should be getting much more time than that, so what's going on? The card is formatted.
2. To adjust the record levels, it makes me adjust each channel of each stereo pair individually. I seem to recall discussion of these being able to be linked somehow, but a quick search turned up nothing, presumably because I'm not recalling the name of the function and am not searching correctly.
These are the only problems I can find right now. Thanks in advance for any guidance.
1) Remaining time is for that file, not the card.
2) Press and hold the button for one channel, then press the button for the next channel. In trim mode.
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1. I'm using a 16 GB card. During my tests, all the tracks were armed for recording (by default), and I hit the "display" button to check the available recording time. It reported 2:04:09. That's not enough time, so I assumed I just needed to tell it not to record channels 5/6 and the mix. I unarmed those tracks, started recording again, hit "display", and, again, it reported 2:04:09. My calculations suggest I should be getting much more time than that, so what's going on? The card is formatted.
2. To adjust the record levels, it makes me adjust each channel of each stereo pair individually. I seem to recall discussion of these being able to be linked somehow, but a quick search turned up nothing, presumably because I'm not recalling the name of the function and am not searching correctly.
These are the only problems I can find right now. Thanks in advance for any guidance.
1) Remaining time is for that file, not the card.
2) Press and hold the button for one channel, then press the button for the next channel. In trim mode.
Thanks!
Follow-up questions, since I've had to break everything down and get it packed up for tomorrow (to take to work with me).
1. So is there no visible measure of the remaining time on the card, anywhere?
2. 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?
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1) Not that I'm aware of.
2) IIRC the linking stays through power cycles. I think.
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1) Not that I'm aware of.
2) IIRC the linking stays through power cycles. I think.
Okay -- I'll just have faith that there will be enough record time. I'm sure the show will be less than 4 hours, and maybe less than 3 hours. So 16 GB should be enough to not even have to worry about it.
Thanks for the quick answers.
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Track linking doesn't remain through power cycles on mine. Also, you can gang/ungang track linking in trim, pan or mix level screens.
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1. I'm using a 16 GB card. During my tests, all the tracks were armed for recording (by default), and I hit the "display" button to check the available recording time. It reported 2:04:09. That's not enough time, so I assumed I just needed to tell it not to record channels 5/6 and the mix. I unarmed those tracks, started recording again, hit "display", and, again, it reported 2:04:09. My calculations suggest I should be getting much more time than that, so what's going on? The card is formatted.
2. To adjust the record levels, it makes me adjust each channel of each stereo pair individually. I seem to recall discussion of these being able to be linked somehow, but a quick search turned up nothing, presumably because I'm not recalling the name of the function and am not searching correctly.
These are the only problems I can find right now. Thanks in advance for any guidance.
1) Remaining time is for that file, not the card.
2) Press and hold the button for one channel, then press the button for the next channel. In trim mode.
Thanks!
Follow-up questions, since I've had to break everything down and get it packed up for tomorrow (to take to work with me).
1. So is there no visible measure of the remaining time on the card, anywhere?
2. 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?
If you go into the menu and look at the card it will tell you how much space you have. It doesn't count down on the display window while recording however.
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I see the levels go from -32 to +32. I'm used to pulling SBD feeds on an M10, and I know if I have to set the M10 below 3 (or so) then the signal is probably too hot. What's the equivalent on this deck?
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I see the levels go from -32 to +32. I'm used to pulling SBD feeds on an M10, and I know if I have to set the M10 below 3 (or so) then the signal is probably too hot. What's the equivalent on this deck?
I want to say I generally run from somewhere between 0 to +5 from soundboards. As long as the signal itself isn't overloaded, I would think the 680 would be able to handle hotter levels fine (with the trim in the negative range), although I haven't run into a situation like that yet.
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.
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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!!
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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).
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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?
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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.
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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.
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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.
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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.
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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
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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.
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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 :P
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"
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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 :P
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.
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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. ;)
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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.
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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.
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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?
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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?
Menu > System > Lock Mode > "T+F" (Top + Front) or "Front." You can lock all the buttons on the front or top panel, but not the top panel switches.
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so i just got my second littlebox to use with my dr680. i plugged it in and it was ch 3/4 and when i would use them channels 1/2 would be picking up levels though nothing was plugged in... anyone else have that happen? also with internal AA batteries does anyone have an issue with them lasting for a while when the unit is just sitting?
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so i just got my second littlebox to use with my dr680. i plugged it in and it was ch 3/4 and when i would use them channels 1/2 would be picking up levels though nothing was plugged in... anyone else have that happen? also with internal AA batteries does anyone have an issue with them lasting for a while when the unit is just sitting?
Cam was the rec function turned off for channels 1 & 2? The black circle under the tracks should be gone (i.e. rec turned off). Press each channel button in rec standby to turn on or off. I have not used double a's in quite some time so not sure about the life of those in certain situations.
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Going to buy a 2nd 680 to cascade for my matrixs... Amazon has it for 439.
Question 1 : Has anyone seen a better price?
Question 2 : Does anyone have a dead 680 ? (I need a button)
Question 3: Does anyone have a case CS-680 they want to sell?
Still love this unit... never a bit of trouble i didn't cause myself... pulling gold with it. I don't cruise here much anymore... i never see other tapers... I miss you all.
I'd send a sample of what i've been doing but soundcloud seems to be down... TIA
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Hey Brian,
That's a good price.
Button or the rotary knob? (I don't have anything for you, but its the rotary knob I've nearly lost a couple times)
Come hang out at Springfest this year.
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#6 button would love to
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Is there or can you check to see how much memory is left? I've looked and there is no way that I've seen to tell how much time you have left.
Thanks
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Menu> Card> Information> Executive> Card Info shows total/remaining GB. Not accessible while recording of course.
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..and you are probably aware that the display does not display the total time remaing on the card, but does display the time remaining until the next file-split.
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Not sure what happened with a recording. Recorded a whole set put the card in my computer reader and there are no wav's there. I put the card back in the DR680 it shows two file names but I get an error "File Not Found In Take"
Is my data recoverable? Do I dare use this card again? It was brand new.
Sorry if this is covered but searching taperssection has become hard.
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Not sure what happened with a recording. Recorded a whole set put the card in my computer reader and there are no wav's there. I put the card back in the DR680 it shows two file names but I get an error "File Not Found In Take"
Is my data recoverable? Do I dare use this card again? It was brand new.
I'd say there's a good chance you could recover the recordings, but it's hard to say for certain. Lock the card and don't use it for anything else until you're able to recover files.
As for using the card again, once you've recovered the files (or given up), maybe try reformatting and trying some test recordings. The first time I recorded a show with a DR-08, something crashed before the end of the show, and I had to manually recover the (incomplete) recording. But after I reformatted the card it always worked fine after that.
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Anyone else have this issue?
When I have internals batteries in my 680 they discharge and die in about a week when the unit is off and sitting, is that normal? They are new rechargeables.
Thanks
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Anyone else have this issue?
When I have internals batteries in my 680 they discharge and die in about a week when the unit is off and sitting, is that normal? They are new rechargeables.
Thanks
I don't own a 680 but I would never leave replaceable batteries in a device like the 680 when not in use - good thing you haven't left alkaline batteries in it or they may have leaked
The EAA PSP-2 will drain batteries when turned off and the sony D7/D8 would keep the clock running off internal batteries if left in but they pulled a very small amount of power and was by design
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Anyone else have this issue?
When I have internals batteries in my 680 they discharge and die in about a week when the unit is off and sitting, is that normal? They are new rechargeables.
Thanks
I agree about the leaking if left in. How does the charged battery life compare to those same batteries just sitting on a shelf?
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Anyone else have this issue?
When I have internals batteries in my 680 they discharge and die in about a week when the unit is off and sitting, is that normal? They are new rechargeables.
Thanks
I agree about the leaking if left in. How does the charged battery life compare to those same batteries just sitting on a shelf?
the rechargeables sitting on a shelf dont discharge for a couple years, and ive never tried putting in alkalines.
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Anyone else have this issue?
When I have internals batteries in my 680 they discharge and die in about a week when the unit is off and sitting, is that normal? They are new rechargeables.
I've noticed that on occasion, but not sure if it happens consitantly or not. I do leave NiMH low self-discharge AA batteries in my recorders until I replace them before the next use, mostly to keep the clocks set. I've never had alkaline cell type leakage problems using NiMH cells in any gear, anyone know if that a legitimate concern with this cell type?
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Anyone else have this issue?
When I have internals batteries in my 680 they discharge and die in about a week when the unit is off and sitting, is that normal? They are new rechargeables.
I've noticed that on occasion, but not sure if it happens consitantly or not. I do leave NiMH low self-discharge AA batteries in my recorders until I replace them before the next use, mostly to keep the clocks set. I've never had alkaline cell type leakage problems using NiMH cells in any gear, anyone know if that a legitimate concern with this cell type?
Ok so you swap them for every show? I guess I should buy an other set of batteries. What do you use for a charger? I have a 4 battery alpha something charger that's very nice and expensive, but I'd like to know what you use to charge, and it's nice if I could do 8 at a time.
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I always run the 680 and R44 off DVD batteries, so the AAs are simply for back-up and clock retention. the DVD batteries have been so reliable that for one-night gigs I've gotten complacent and manytimes don't bother replacing the AA's even if they've become discharged. I always replace them before a multi-day fest when I know I may be swapping DVD batteries.
I use Maha Imedion AAs and the Maha MH-C9000 charger, which may be a bit over-compex but works well and has modes for break-in, cell matching and analysis and runs off 12VDC so its campable. It only does 4 cells at a time though. Pretty sure Maha makes 8 cell chargers as well.
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Anyone else have this issue?
When I have internals batteries in my 680 they discharge and die in about a week when the unit is off and sitting, is that normal? They are new rechargeables.
I've noticed that on occasion, but not sure if it happens consitantly or not. I do leave NiMH low self-discharge AA batteries in my recorders until I replace them before the next use, mostly to keep the clocks set. I've never had alkaline cell type leakage problems using NiMH cells in any gear, anyone know if that a legitimate concern with this cell type?
Ok so you swap them for every show? I guess I should buy an other set of batteries. What do you use for a charger? I have a 4 battery alpha something charger that's very nice and expensive, but I'd like to know what you use to charge, and it's nice if I could do 8 at a time.
Hey philzone,
Have a 680 and leaving in batteries does drain them. I remember some post about this way long ago at TS. Not sure where though. I put in Akaline double A's and in 3-4 weeks or month plus or so they would drain. So I never run them anymore. Even though they are a back up if my external goes bad. Don't like the weight they add to the machine and they did leak once though I just caught it and it cleaned out easily. Did not like taking them out after each show and sometimes I would forget if I ran them the last time and how much did they drain from just sitting in the closet between show and 8 batteries are expensive way to go. (Found Home depot had cheapest double A batteries) The double A would run out at about 2.5 hours running 6 channels at 24/96 with phantom on for one set of channels. Most shows I attend are 3 hours so the batteries really are only a back up and to have 8 of them was too much.
So I just go with the most reliable external battery - tekkeon 3450 and it runs the 680 with even 8 channels at 24/96 for 4+ hours. Unless I am at a very long show. All I need is another tekkeon backup and I am good for 4 more hours. Many do not like the tekkeon though...
Just find what works for you and go with it.
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It's been a while since I've tested this, but I'm 99% certain the batteries won't drain if you correctly power down the unit and only drain when left in stand-by mode. Make sure you turn the unit completely off and then use the "hold" switch on the top of the unit.
From a best practice perspective:
- Not using internal AAs is incredibly risky
- They shouldn't be left in the unit for extended periods of time, but if it means you're too lazy/busy to replace them... leave them in.
You need to be using AAs. If you don't and lose power, you are basically fucked and your files will not be recoverable. Just because an external battery has adequate capacity (perhaps many times over) does not safeguard you from losing power. Bags get bumped an power cables can get yanked. Imagine a wook spills a beer on your bag and you are frantically trying to clean up the mess? Think you can do this without accidentally yanking the power cord? How about if the DC Input jack becomes loose? Mine is and if I wiggle it just right, it will cut over to internal AA power. Power failure can occur in many ways and it's suicide not to use the internals when the unit won't finish writing to the SD Card.
As far as leaving in batteries between shows, yes... you should remove them when applicable. Not going to a show for a week or longer? Probably best to remove them. Have a show in a few days? Don't even bother... leave everything in your gear bag and just make sure to properly power down the unit w/ the hold switch.
At the end of the day, this is a $400 recorder. The batteries probably aren't going to leak in a week and if they do, it's easily replaced technology.
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Just to clear it up I use external power. My battery is the naztech pb15000 and its awesome.
Those are all very good chunks of info, I guess I just have to take out the batteries when not in use.
Thanks
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Yep. One should always have charged AAs in there when recording with an external power source. Just because I sometimes get away without swapping them out them once discharged certainly isn't a suggestion to do that! It's a Berenstain Bear lesson waiting to happen.
I am curious about whether NiMH rechargeable AAs actually pose a realistic potential leak hazard or not, regardless of removing them as best-practice.
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Possible very stupid question warning, but could you just take out one battery to prevent drain?
I used to keep AA's in there, but got tired of switching out the drained ones, and almost never do anymore.
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Possible very stupid question warning, but could you just take out one battery to prevent drain?
I used to keep AA's in there, but got tired of switching out the drained ones, and almost never do anymore.
I was thinking that as well, ill try to do that at some point as long as it won't damage the unit.
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Possible very stupid question warning, but could you just take out one battery to prevent drain?
I used to keep AA's in there, but got tired of switching out the drained ones, and almost never do anymore.
I was thinking that as well, ill try to do that at some point as long as it won't damage the unit.
FWIW: I always use alkaline batteries in my DR 680, and experience no drain at all. In contrast, I have a Tascam HD-P2 which will apply a constant drain to the batteries inside. I may go quite some time between using either deck as I have a few other decks I use too, but have never had an alkaline leak. I am sure they could, but it would likely have to be in the deck a pretty long time.
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Maiden voyage Thursday for a tracking session. Local HS girl needs an audition recording (tenor player). Picked up two K&M stands (210/2 and 210/9): the build quality on these is incredible!
Plan (six track):
Avantone CK-40 in Blumlein (grand piano)
Avantone CK-1 (Busman), Recorderman configuration (drums)
AT4050 (bass)
AT4050 (tenor)
Recording in a HS music room. Can't imagine it will sound good (so I'm close micing), but I'm making a separate safety room recording with an AT825>PMD661.
I've read all the posts. I've played around with it in my basement.
Wish me luck....
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good luck..
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Hey all, forgive me if I've posted this in the wrong spot.
I was doing some SFX recording today with a loaned DR-680 when this happened:
https://soundcloud.com/james-ashton/tascam-dr680-overheating (https://soundcloud.com/james-ashton/tascam-dr680-overheating)
I had a quick search on the internet and found a few people mention that they can overheat, and I suspect that I may have fallen victim. It was around 35 degrees (95 Fahrenheit) and I was recording to all 6 channels, with +48v phantom engaged on each. The AC power indicator stayed on, even when I removed the AC plug whilst troubleshooting. After 5-10 minutes (and a software reset) it seemed to function correctly again, only to start doing it again 15 minutes later.
Does anybody recognise this fault to be due to overheating? And I guess more importantly - is it simply a heat issue that will disappear once the unit has cooled? Or does this mean the unit needs to go back to Tascam for repair? I've racked up quite a few hours with this recorder now and was quite impressed up to this point, but this definitely ain't cool.
Any advice would be greatly appreciated
James
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Based on your description, I would say that overheating is certainly a possibility. In my experience, the dropouts would gradually increase from about 1-2 seconds apart to relatively close to the pace in your recording. I assume this occurred on all six channels?
I can't tell you if the unit needs repair. In my situation, the problem doesn't seem to have caused any long-term issues. The unit functions exactly as expected once its cooled down and my ears don't detect any issues. That said, I have always been curious to take a look inside and will probably do an inspection at some point. If everything looks ok on the inside, I may consider modding the case to improve ventilation.
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Thanks for your reply - well I've had a trouble free day of recording today (it was 10 degrees cooler!), so I'm guessing that I'm free of long term issues too. Fingers crossed. Might take a cooler with some ice packs with me on the next hot one!
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My experience was similar to hi and lo's- only happened once a couple years ago, with the recording bag in the sun at an outdoor festival. At the time I was clock synced to an R44 and thought the sync may have been the issue, but that has never been a problem since. Once it cooled down it reverted to normal operation and has workd fine ever since.
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Four years late to the party. ;D
The jazz quartet recording went well, given the room I had to work with. The deck performed flawlessly, although I did use AC since it was available. Ganged function is wonderful. Headphone output quality is definitely so-so, but loud enough (and louder than my PMD661). I agree that Tascam allocated their money where it mattered most: preamps. They were definitely good enough, and quiet enough, for this gig. Unavoidable bleed between mics, but I was able to isolate instruments fairly well. I like the Recorderman technique for drums a lot; simple, and it works quite well if you only have two mics.
Appreciate all the beta testers out there and the amount of information available to me ahead of time. Helped immeasurably.
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Generally, how noisy is the 680? Are the 1/4" inputs noisier? I've recently recorded 1/4" mic-in, gain on high and had LOTS of self noise. Just a constant humming sort of hiss in the background. One time a relatively quiet, but still fully amplified show with as much gain as the machine had and the recording is basically unlistenable. Just last night ran again, mic-in, gain on high, but trim only at ~11. Still VERY noticeable noise during the quiet parts. Is this normal?
When running XLR in it has been much quieter. Maybe just simply since less gain was needed? I can't remember what my settings were for XLR in.
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Generally, how noisy is the 680? Are the 1/4" inputs noisier? I've recently recorded 1/4" mic-in, gain on high and had LOTS of self noise. Just a constant humming sort of hiss in the background. One time a relatively quiet, but still fully amplified show with as much gain as the machine had and the recording is basically unlistenable. Just last night ran again, mic-in, gain on high, but trim only at ~11. Still VERY noticeable noise during the quiet parts. Is this normal?
When running XLR in it has been much quieter. Maybe just simply since less gain was needed? I can't remember what my settings were for XLR in.
The 1/4" TRS inputs are the same as the XLR/TRS inputs, they just don't have XLR. With the gain set correctly noise shouldn't be an issue. This was recorded with 5 mics:
http://www.youtube.com/watch?v=61DgL-uij10
The guitar and vocal were recorded with dynamic mics (2 SM57s and 1 545) with the trim at +5, while the room mics were 2 AT853s, trim at -11. Gain switch set to "HIGH" for all channels. I would argue that any hiss was negligible.
What microphones are you using? When running 1/4" are you running them balanced (TRS rather than TS)?
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The other day I ran AKG 63s with Naiant Actives and PFA, last night a brand new pair of 61s. I was running TS.
I lost my XLR extension cable, so have been running my board feed cable that has RCA (both times with XLR adapters) on the PFA side and 1/4" TS on the recorder side.
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I would experiment and compare, but my guess is it's an issue with running unbalanced.
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Okay, thanks. Been putting off buying another XLR cable, but looks like that will need to happen.
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Fuuuuuuuck. Recorded tonight, started up, everything looked fine, went up front for the majority of the set, came back and there was no levels. Stopped the recording, it was kind of show to respond and when I pressed pause to see if I was getting levels the date on the take showed all zeros. Turned it off and on again and it went back to normal. Not sure how much I missed.
I formatted my card (32 g transend ultra) on Thursday. Had recorded two shows on it last weekend.
Wtf.
Just did it again like 20 minutes after restarting it. :(
Update: Formatted the card and have been recording for 2 hours with no hiccups. Going to keep it going for another few hours to see what happens.
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⬆ That sucks Allyn! If you need to you can borrow my DR-2d.
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Uuugh, got the 'battery empty' message. Not like I'm recording anything, but first time I've seen that one. Running of an external battery, no power was cut from it, but the machine stops the recording to display the message about the internal AAs. >:( Note to self, make sure that doesn't happen in the field.
I feel like these are localized problems that I could avoid with fresh batteries and a formatted card, so not hating on the 680, but its kind of frustrating.
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Possible intermittant power cable that forced it to switch to the interenal AAs?
Good that it stops recording before the internal AA's die. It will not save the files gracefully if it dies while recording.
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That's a good point. Don't think there was any movement on the power cable but I suppose it's possible.
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If the power connection is intermittant, it should automatically switch back to the external source once the connection is good. However, if the AA's don't have enough charge to keep things going before that can happen it might pop up that warning and stop recording.
Double check to make sure both your external battery and internal AA's are charged and the battery cable and connections are in good shape.
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aloha
ok so my 680, when i put in record pause with a formatted memory card (32bg) says -1:02:06 time left. when i record past this it just starts over. what am i missing here?
thanks
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Pretty sure you need to change the file size limit from 1GB to 2GB.
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ok so my 680, when i put in record pause with a formatted memory card (32bg) says -1:02:06 time left. when i record past this it just starts over. what am i missing here?
The time is how much is left for the current file(s), not how much time is left on the card.
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aloha
ok so my 680, when i put in record pause with a formatted memory card (32bg) says -1:02:06 time left. when i record past this it just starts over. what am i missing here?
thanks
You are recording at 24/96, right? With a 2gb limit, you get an hour at 24/96 - 2 hours at 24/48.
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yah thanks everyone.
what i figured but wanted to confirm. so just need to accept that every hour it starts over the running time when doing 24/96 i guess.
thanks!!
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so regardless of file size limit, if I'm running 8 tracks 24/96 the counter will reset every 20min?? there isn't a way to have the deck just keep a running time for whole project?
that is pretty weak if so and defiantly adds to the stomach acid content when doing important stuff.
anyways
thanks!!
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so regardless of file size limit, if I'm running 8 tracks 24/96 the counter will reset every 20min?? there isn't a way to have the deck just keep a running time for whole project?
that is pretty weak if so and defiantly adds to the stomach acid content when doing important stuff.
anyways
thanks!!
it's 2 g per file. If you record in stereo pairs you'd have 4 files each containing 1 hr of material and 2 g in size. You could run in mono.
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so regardless of file size limit, if I'm running 8 tracks 24/96 the counter will reset every 20min?? there isn't a way to have the deck just keep a running time for whole project?
that is pretty weak if so and defiantly adds to the stomach acid content when doing important stuff.
anyways
thanks!!
Using the DR-680 for "important stuff" is a terrible idea. If you need a feature rich recorder with reliable operation, there far better options.
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Using the DR-680 for "important stuff" is a terrible idea.
Says...you?
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Using the DR-680 for "important stuff" is a terrible idea.
Says...you?
Yes. I was one of the first adopters of the 680 on this forum and have experienced virtually every known issue regarding the 680, including two DOA units. Acidjack once said to me "I feel like if I even look at that thing the wrong way it will shatter to pieces."
It's a decent deck for what we do, but I would never recommend someone use it for important (i.e. professional) work.
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Yes. I was one of the first adopters of the 680 on this forum and have experienced virtually every known issue regarding the 680, including two DOA units. Acidjack once said to me "I feel like if I even look at that thing the wrong way it will shatter to pieces."
It's a decent deck for what we do, but I would never recommend someone use it for important (i.e. professional) work.
But yet plenty of other people haven't had issues. It's unfortunate that you've had problems, but saying that they shouldn't be used for important work is a bit over the top.
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You should really read through these threads. Most people have had one issue or another.
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You should really read through these threads. Most people have had one issue or another.
I've read through the threads and have been using my own unit for several months. I stand by my statement.
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Famous last words!
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thanks for the suggestion.
all i want is to be able to run it for a few hours running 6-8 tracks and see that it has been running a few hours not starting time over every 20 min.
good thing this was a "here can you use this thing" deal for me. oh well thanks again for the suggestions and comments.
so regardless of file size limit, if I'm running 8 tracks 24/96 the counter will reset every 20min?? there isn't a way to have the deck just keep a running time for whole project?
that is pretty weak if so and defiantly adds to the stomach acid content when doing important stuff.
anyways
thanks!!
it's 2 g per file. If you record in stereo pairs you'd have 4 files each containing 1 hr of material and 2 g in size. You could run in mono.
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Once started recording, the machine will continue to record until the memory card is full. If recording at 24/96 it will start new files approximatley every hour or every two hours depending on how it is set. Starting new files more frequently is one hedge against file loss in the case power is lost or some other problem occurs. In that situation only the current file(s) being writen would be likely to have problems.
How long the machine will record to a file before starting a new one is determined by three factors: the bit-depth/sample-rate at which you are recording, how many channels are encoded in the file, and the file size limitation.
Here's each of those factors-
1) Recording at 24bit/96Khz produces 16.5 MB/min per channel. That's just under 1GB of data per hour.
2) The DR-680 can be set to write either single channel mono files (1 channel per file) or 2-channel interleaved-stereo files (2 channels per file) for the six analog input channels. 2-channel stereo interleaved files contain twice as much data as single channel mono files. (Note that the 'extra' digital-input or monitor-mix stereo channel is always a 2-channel stereo-interleaved file)
3) The file size limitation is 2GB. Once a file reaches this size, a new one is started by the recorder. This happens "seamlessly" meaning that there is no gap in recording, or in playback once the files are rejoined or played sequentially.
So if you are recording at 24bit/96Khz, and have the machine set to record single channel mono files, the machine will start a new file for each channel approximately every two hours.
If you set the machine to record 2-channel stereo files (which are always channel pairs- ch1&2, ch3&4, ch5&6), the machine will start new files for each channel pair approximately every hour
Either way, the stereo digital-input / monitor-mix file will always be a 2-channel stereo file and will start a new file approximately every hour.
If recording at 24/48 the data rate is half that of 24/96, so the file slits happen at just over 2 or 4 hour intervals instead of 1 or 2 hour intervals.
[edited to correct the data rates]
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thanks, great explanation!!
yah ive understood "how" it all works i was just hoping to find a way to have a cumulative "recording time" shown regardless of how i setup the recorder, and if it does or does not do a seamless new file start/creation while it is running.
it just seems logical that it should still give me a running total of the time it has been recording even if it does a background process like starting a new file. It appears that this is not possible with this unit.
.
thanks again for all the info
Once started recording, the machine will continue to record until the memory card is full. If recording at 24/96 it will start new files approximatley every hour or every two hours depending on how it is set. Starting new files more frequently is one hedge against file loss in the case power is lost or some other problem occurs. In that situation only the current file(s) being writen would be likely to have problems.
How long the machine will record to a file before starting a new one is determined by three factors: the bit-depth/sample-rate at which you are recording, how many channels are encoded in the file, and the file size limitation.
Here's each of those factors-
1) Recording at 24bit/96Khz produces 16.5 MB/min per channel. That's just under 1GB of data per hour.
2) The DR-680 can be set to write either single channel mono files (1 channel per file) or 2-channel interleaved-stereo files (2 channels per file) for the six analog input channels. 2-channel stereo interleaved files contain twice as much data as single channel mono files. (Note that the 'extra' digital-input or monitor-mix stereo channel is always a 2-channel stereo-interleaved file)
3) The file size limitation is 2GB. Once a file reaches this size, a new one is started by the recorder. This happens "seamlessly" meaning that there is no gap in recording, or in playback once the files are rejoined or played sequentially.
So if you are recording at 24bit/96Khz, and have the machine set to record single channel mono files, the machine will start a new file for each channel approximately every two hours.
If you set the machine to record 2-channel stereo files (which are always channel pairs- ch1&2, ch3&4, ch5&6), the machine will start new files for each channel pair approximately every hour
Either way, the stereo digital-input / monitor-mix file will always be a 2-channel stereo file and will start a new file approximately every hour.
If recording at 24/48 the data rate is half that of 24/96, so the file slits happen at just over 2 or 4 hour intervals instead of 1 or 2 hour intervals.
[edited to correct the data rates]
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Famous last words!
Hi and Lo....I am not going to try to play down any issues you have had. And I feel like you, I and a few others have been in the trenches of the 680 from the start. You are absolutely an example of worst case. I feel exactly the opposite and even have two of the decks. I do in fact use the deck for very important, paying, no second chance work. I am not going to argue your stance, I am well aware that you feel justified in your post's and the position you take. You opinion is your opinion and we are all entitled to think what we choose. We can all believe what we want, but just because a persons believes something does not make it true or right. To even the playing field, I do think this deck can perform flawlessly, time and time again. That's my opinion and that's what I believe. What I would hope that someone takes away from the pro's and cons; positive and negative reactions is that to feel safe and ensure no issues you need to be well aware of what you doing with the deck, your powering options and the fact it is not built with a cast aluminum case like some other much more expensive units so it is not something that will take a licking and keep on ticking. The deck does have it's own set of limitations that you need to be aware of. But for the amount of features you get coupled with the recording quality you can get there is nothing near it at this price point. Do we all feel that it is a safe choice for doing pro work....no, you are a prime example. Do some of us feel that it is a sfae choice for doing pro work...yes,I am a prime example. One thing I think we do ALL agree with is that for approx $500 you cannot find anything else that will give you as many channels, as good a recording quality and as many features built in. Are there decks that are better, of course. Can you buy them for even twice the price, not really. I also think that if you went back to square one of this thread, many many problems come down to operator errors. Many many others seemed to be related to battery power issues. Have you noticed that once all of us here on TS discovered the flaw with multi-voltage batteries and the 680 that the problems reported dropped way off? So if you are looking to buy a deck to do pro work on a budget I say get this deck. IF you are starting a business and you want to be as sure as you can be recording more than four tracks at once....save you money or change you business plans buying strategy and increase your budget. IF you are trying to make a decision based on the response of this thread...do yourself a favor.....set aside quite a few hours and read through the entire thread then make your own call on what to do. If you ask us...we are all gonna have a slightly different answer 8)
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One thing I think we do ALL agree with is that for approx $500 you cannot find anything else that will give you as many channels, as good a recording quality and as many features built in.
Absolutely, I am in complete agreement.
Are there decks that are better, of course. Can you buy them for even twice the price, not really.
Here's where I disagree and say the DR-680 is not fit for critical applications. Although the average taper might be OK with a botched recording ever now and then, professional applications demand perfect reliability. Just wait until July and the 100 degree temps... you will see plenty of 680 complaints. This deck has multiple points of failure and is only fit for hobbiest applications. We recommend the 680 to a local venue where we have an installed rig, because it seemed like the most logical choice, and it failed to power up after a few months of use.
I'd much rather have 4-channels at 2x the price (ala the R-44) because it will work every time.
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Nevertheless in my case I went well with DR 680, but not with R44 as that was damaged three times during 2 years while connecting it to PC via USB for transfering files, following three weeks for repair...
That was for me the reason to change to DR 680 - but I try to work with some redundancy.
Setup is no more lightweight and higher costs but more sure for recording and includes a real good metering on netbook as described earlier here:
http://taperssection.com/index.php?topic=147677.msg1943695#msg1943695
Meanwhile I use a set of two LiPo batteries (5000mAh) an RME UC instead of the scarlett.
If you go with the RME UC Mic's via SPDIF to digital In (DIN) of DR 680 you may even realize 8 Channels recording (4 with Quadmic, two via DR680 chanel 5/6 and the two UC Mic's via SPDIF, but that is a little less "redundant" in case of DR 680 failure for the 5/6 channels while RME UC runs well on if netbook crashes.
More sure but more volume and weight (and money)
Best regards
Grosse_j
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Ok so I will be running bsc1's to channels 1 & 2 and SBD to channels 3 & 4. Now my question is can you do an on the fly matrix on a separate track? While you still have the two other tracks channels 1 & 2 and channels 3 & 4. If so how do you set it up in the menu to do this? Thanks in advance as I am still relatively new to using it.
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Ok so I will be running bsc1's to channels 1 & 2 and SBD to channels 3 & 4. Now my question is can you do an on the fly matrix on a separate track? While you still have the two other tracks channels 1 & 2 and channels 3 & 4. If so how do you set it up in the menu to do this? Thanks in advance as I am still relatively new to using it.
Yes, by default your monitor mix will get written to the stereo track (7+8 if you want to call it that). Just enable that track and set you mix using the pan and monitor level controls.
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Just received my new DR-680 today :)
So far it seems to be working just fine, but there is one thing that is bothering me. The speaker on top hisses all the time, regardless of the volume level. When I insert a headphone jack then it does indeed turn off, as the manual says. But is there a way to turn that speaker off (kill that hiss) without inserting a headphone jack? I have a loose headphone jack which I will use for that purpose if there isn't another way, but that is a workaround I would prefer not to need.
Regards, Christine
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Just received my new DR-680 today :)
So far it seems to be working just fine, but there is one thing that is bothering me. The speaker on top hisses all the time, regardless of the volume level. When I insert a headphone jack then it does indeed turn off, as the manual says. But is there a way to turn that speaker off (kill that hiss) without inserting a headphone jack? I have a loose headphone jack which I will use for that purpose if there isn't another way, but that is a workaround I would prefer not to need.
Regards, Christine
Welcome aboard!
I just insert a 1/4" headphone adapter in there, as I never want that speaker to work.
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I just insert a 1/4" headphone adapter in there, as I never want that speaker to work.
That's what I was planning on doing. Good to know I didn't receive a bad unit (would be just my luck on the 13th of the month). Thanks for reassuring me :)
Regards, Christine
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That's what I was planning on doing. Good to know I didn't receive a bad unit (would be just my luck on the 13th of the month). Thanks for reassuring me :)
I don't recall offhand how hissy my speaker is, but I also just keep an adapter in the jack all of the time to mute the speaker.
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I suspect most DR-680 users leave an adapter in there. I actually use the headphone output occasionally when I record, so I just plug my headphones into the adapter when I need to. FWIW, that headphone output is crap too. It's very noisy.
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after having the first 680 in the world to have the main board take a crap and writing them off a few years ago
I broke down and bought one cheap on eBay the other day... just got it in the mail
sold it months ago and never plan on looking back ;)
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after having the first 680 in the world to have the main board take a crap and writing them off a few years ago
I broke down and bought one cheap on eBay the other day... just got it in the mail
Hi, my name is darby I'm a DR-680a'holic. It's been a few years since I last used a DR-680... 8)
Glad to see you back in the fold darby. I've had three of them crap out on me and I still like them. You just have to recognize the limitations. ;)
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Just received my new DR-680 today :)
So far it seems to be working just fine, but there is one thing that is bothering me. The speaker on top hisses all the time, regardless of the volume level. When I insert a headphone jack then it does indeed turn off, as the manual says. But is there a way to turn that speaker off (kill that hiss) without inserting a headphone jack? I have a loose headphone jack which I will use for that purpose if there isn't another way, but that is a workaround I would prefer not to need.
Regards, Christine
Just looking for some clarification here. I have two 680s and have no issues with the speakers hissing, and I use the headphone out frequently to play recordings( through my car stereo) on the ride home after a gig.
Although I seem to recall some past discussion about the speaker/headphone quality here, I question whether or not Christine's unit may be defective?
edit to add: yeah the speakers crap out on loud playback. If you are playing through a stereo via headphone out good results will come from keeping the 680 'phones volume' at a moderate volume and adjusting gain on the stereo.
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I have noticed (with slight alarm) input audio coming out of the speaker while recording. I either use a headphone adapter or just keep the headphone gain pot turned all the way down to silence it. Still suprised there isn't a switch (menu or physical) to disable the speaker though, since there is on other Tascams like the DR-2d, and it would be more likely to cause a feedback condition for applications this machine is designed for, so it seems like something of an oversight to me.
The headphone output isn't great but is usable. Although it uses a smoothly variable pot, the gain changes are not smooth but exhibit rather coarse digital steps which are noisy. It has enough output to drive Senn HD650s decently enough though and as long as I don't need to adjust playback level with the headphone knob while listening it's okay. Actually, cranking it up the headphone knob and adjusting level using the monitor mixer stereo master instead is smoother and more acceptable for me. However doing that might get too hissy with lower impedance phones which will be louder at any given knob position than those 600ohm Senns.
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after having the first 680 in the world to have the main board take a crap and writing them off a few years ago
I broke down and bought one cheap on eBay the other day... just got it in the mail
Welcome back, pioneer.
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Sorry to come after so many things seem to have happened with this Tascam since it was released.
But today I found myself recommending the DR-680 on the BMCuser forum, but not having never actually used it.
On that forum they were considering using a Sound Devices Mix Pre-D to feed a Tascam DR-60D. IMHO you could get more capabilities by simply using a DR-680.
But the reliability question was not something I was considering, and that's why I came here to get some feedback on the DR-680 field use.
You are certainly aware that double-system audio has been reinstated since the use of DSLR video cameras became so widespread. Some insist on recording on the video cameras, but most are aware that if you want high quality audio you need to record it on an external digital recorder.
The Tascam DR-60D is probably becoming a standard, at least on that forum, which is a good sign, as that Tascam deserves it, in spite of its hum issues and limited battery times.
A superficial reading here has already shown we to potential problems: battery issues and temperature potential problems.
Can anyone please elaborate a bit more on these two and tell what other problems have been reported for location audio?
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Sorry to come after so many things seem to have happened with this Tascam since it was released.
But today I found myself recommending the DR-680 on the BMCuser forum, but not having never actually used it.
On that forum they were considering using a Sound Devices Mix Pre-D to feed a Tascam DR-60D. IMHO you could get more capabilities by simply using a DR-680.
But the reliability question was not something I was considering, and that's why I came here to get some feedback on the DR-680 field use.
You are certainly aware that double-system audio has been reinstated since the use of DSLR video cameras became so widespread. Some insist on recording on the video cameras, but most are aware that if you want high quality audio you need to record it on an external digital recorder.
The Tascam DR-60D is probably becoming a standard, at least on that forum, which is a good sign, as that Tascam deserves it, in spite of its hum issues and limited battery times.
A superficial reading here has already shown we to potential problems: battery issues and temperature potential problems.
Can anyone please elaborate a bit more on these two and tell what other problems have been reported for location audio?
Carlos, Are a Tascam Rep? Are you an employee of Tascam? It might help, at least, if you could identify yourself if you are. I would love to talk to someone from Tascam, someone who will actually listen and then address the issues. I can give you an extensive list of issues since I now own two of them and have been involved in this discussion from the begining. But....If you are someone that is a relative newcomer to tapersection and recording in general looking to glean information from over 7 series of posts I'd say take a weekend and read the posts. To just breeze in and ask for the cliff notes of the information we have all figured out the hard way doesn't show much respect for the people involved. BUt really hope that you are officially here from Tascam. And I will keep my finger crossed that if you are we don't get any of the standard put offs or canned responses that are pretty much the standard from Tascam, Kirk
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To just breeze in and ask for the cliff notes of the information we have all figured out the hard way doesn't show much respect for the people involved.
Arguably telling somebody to suck it and read a 6 part thread to get an overview of the problems shows far less respect.
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Of course I am not a Tascam rep or have anything to do with Tascam.
I am not a relative newcomer to Tapersection. If you read the posts I wrote (175) may not be as many as you, but I did have my share a few years ago, when I was involved in some recording projects.
What I do remember too is that people here were more cordial and helpful than you are being. If you don't want to help, it's your prerrogative, but to treat me as disrespectful is not. You are the one being so.
I have been in technical forums for a long long time, since they were mail connected, and people have always been generous and open, and helpful. And I always was so too, here and everywhere I was in. Speaker forums, solid state audio forums, location audio forums, and many many more.
Nobody is expected to read pages and pages of posts, and if you think someone would you are completely wrong. And that's certainly not unrespectful. You are the first person I find in these forums that showed a lack of generosity, very rude indeed.
Of course I will not be expecting any opinion on the extensive list of issues you mentioned about this recorder, and I certainly do not want it.
Carlos
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Sorry to come after so many things seem to have happened with this Tascam since it was released.
But today I found myself recommending the DR-680 on the BMCuser forum, but not having never actually used it.
I can't figure why anyone would do this. ???
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I read what I wrote on the BMCuser forum, and I didn't actually recommend the DR-680. Simply suggested it as an alternative option to what someone was considering using.
You see, I was a film location recordist for a very long time, in the '70s and '80s. Later on, in the '90s, I started an audio equipment rental business for film & TV, which lasted until the early XXI years.
You start to vow for products that you haven't used yet, probably because in my old times some brands seemed to warrant the quality when they released some product, and such products were more sparse... and expensive. If you ever handle a portable analog Nagra you will know what I mean.
But that did start to change in the '80s and '90s, and it's probably not what it was anymore. Tascam (formerly Teac) started releasing pro-audio products, and their DAT machines and 8-channel studio Hi-8 tape machines became a hit. I did own one of their portables, and rented it, even if it was rather fragile.
Perhaps I continue to do what I did on my business: let my recordist experience read the specs between the lines, if that's really possible, and pick what might be a good product. That's probably what I did when I suggested the DR-680.
The tide is turning on the affordable portable video market, and doing double system audio is becoming the norm. As double system audio is something I know quite a bit of, my experience may help.
Another thing is that there has been a spread in pricing too. And also several new combinations, putting together multi-channel mixers with digital recorders.
Brands like Sound Devices, Nagra and others occupy the multi-thousand niches, and others like Tascam and Roland are offering products that might become useful tools, as long as they are reliable.
Portability, size, weight and plenty of channels is key for location audio. Unfortunately these recorders demand more attention that 2-channel ones. And so require a sound person to handle them.
But that might also change with these DSLR video cameras popularity. The camera assistant is coming back, and probably the sound recordist too.
In any case, the Tascam DR-680 seems to fill above requirements.
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To just breeze in and ask for the cliff notes of the information we have all figured out the hard way doesn't show much respect for the people involved.
Arguably telling somebody to suck it and read a 6 part thread to get an overview of the problems shows far less respect.
In your opinion. IF I ever want to find out some knowledge about something I take the time to do the research myself. then I might ask some questions. Everything you would want to know about the issue a multitude of people have had...pro and con...is all here in those pages it's just that easy. If you don't want to bother reading them............
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Sorry to come after so many things seem to have happened with this Tascam since it was released.
But today I found myself recommending the DR-680 on the BMCuser forum, but not having never actually used it.
I can't figure why anyone would do this. ???
X2 then there is recommending something because of what someone else said without doing any in depth research...
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Has anyone seen the service DR-680 service manual to see how the mic balanced inputs are implemented? Chips used and all that?
From what I read and some YouTube videos around, the preamps seem not to do that bad when compared to Sound Devices pres, which is a very good thing to say.
There are not that many weak points to see. Perhaps the whole bunch of keys at the top, but they don't seem to me as something you will often need to mess with every moment. You set them (or check) in the morning, and off you go.
From a practical point of view, based on my recordist experience, perhaps the only weak point is having just one pot to set levels one at the time for each channel.
OTOS that might not be a major problem it you always use limiters (which you should on a film location recording) and if you set the levels properly on the wireless mics you will probably use on at least 5 inputs out of 6.
Then you can let the "wild" pot free to take care of the boom mic, which is the one that might get you into trouble.
The battery question, which I saw mentioned somewhere on this forum as a troublesome one, I would handle as I did with my TC slates, that carried similar 8-AA battery packs. Get some NimH 12v packs and problem solved.
As I believe in time-code as the safest way to keep things in sync, what I wonder is how to possibly adapt a time-code generator on one of the audio channels. Perhaps Input 6 could be used for that on track 6, also mic radioing the stream to the video camera(s).
Some people think TC is not necessary if you are organized and do slate your shots as in film times. But maybe TC is a a more efficient way if your crew is very small.
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Tascam (formerly Teac) started releasing pro-audio products, and their DAT machines and 8-channel studio Hi-8 tape machines became a hit.
Tascam is not formerly Teac is that what someone told you?
In 1953, a pair of engineers in Japan - the Tani brothers - created the first TEAC product line, which was comprised of open-reel audio tape recorders. In the late 1960's, the Tani brothers and Dr. Abe, a senior engineer at TEAC, formed a special R&D group named TASC (TEAC Audio Systems Corp.) for the purpose of researching ways to apply TEAC's recording technology for musicians and recording studios. Tascam started out as a research and development group to research how to use TEAC's recording technology in musician and recording studio products. The group was called TASC (TEAC Audio Systems Corp). The founders included Mr. K. Tani, one of the founders of TEAC-Japan and Dr. Abe, a senior TEAC-Japan engineer. In 1971 TASCAM (TASC AMerica Corp.) was founded to distribute TASC products in the U.S. Tascam's first products were TEAC brand multitrack recorders. In 1972 it introduced the first low-cost mass-produced multitrack recorders with Simul-Sync. On March 4, 1973 TEAC merged the Tascam Corporation into TEAC Corporation of America (TCA). TEAC-Japan retains the exclusive worldwide rights to the TASCAM brand name for their professional audio related products. Ironic that Tascam started as a company that did it's own research on products. :facepalm:
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Service manual can be found in the user manual archive section of this site, posted in multiple sections:
http://taperssection.com/index.php?topic=147560.0 (http://taperssection.com/index.php?topic=147560.0)
http://taperssection.com/index.php?topic=147559.0 (http://taperssection.com/index.php?topic=147559.0)
I have a copy the full manual in a single PDF (~2MB) which I can email you if you like.
Switching between channels to adjust input gain with the pot is fast and easy using the channel selection buttons located under the meters. It's more difficult in the dark by feel, but still doable. Ganging adjacent channels together to adjust them all simultaneously with one knob while retaining there relative differernce settings is well worth having to select using the buttons IME, but I'm not a video guy.
The limiters (and low-cut) are implemented post ADC, so presumably they will not prevent clipping the analog input stage or converter unless there is a feed-back loop to the analog stage which isn't indicated. There are circuit diagrams showing that in the service manual as well as a useful diagram showing levels though the system.
I use the Tascam snug cover which prevents accidental movement of the switches on top, but I do always make an effor to double check them before recording. The top switches can be set to be locked out seperately from the front swtiches if desired, but I don't use that feature.
The short story on external battery powering seems to be not to use a supply that self-senses voltage output needs, but is either single output voltage only or hard-switched.
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1973, the time Tascam merged with Teac, was more or less the time I started in the film business.
In those times Teac was more known than Tascam, at least in the music audio circles, due to their reel tape-recorders.
Only much later did the name Tascam started to appear.
In the '80s Teac went into other areas, always in home audio, and Tascam launched their 1/2" multi-track tape-recorders, which I never used but made the Tascam name widespread for pro quality products.
Nowadays, AFAIK, the name Teac has dropped below Tascam's, as Teac Professional. Tascam is all over the place.
I do own a Tascam DR-07, BTW, which I have been using for backup audio on my video doc setups. But when I use multiple mics, I output audio from the external recorder or the mixer to the video camera.
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Service manual can be found in the user manual archive section of this site, posted in multiple sections:
http://taperssection.com/index.php?topic=147560.0 (http://taperssection.com/index.php?topic=147560.0)
http://taperssection.com/index.php?topic=147559.0 (http://taperssection.com/index.php?topic=147559.0)
I have a copy the full manual in a single PDF (~2MB) which I can email you if you like.
Unfortunately the most important part, the schematic, is not there anymore. Perhaps you can e-mail me that part?
One thing I would like to do is to compare the input with the DR-60's.
Switching between channels to adjust input gain with the pot is fast and easy using the channel selection buttons located under the meters. It's more difficult in the dark by feel, but still doable. Ganging adjacent channels together to adjust them all simultaneously with one knob while retaining there relative differernce settings is well worth having to select using the buttons IME, but I'm not a video guy.
What's a video guy? ;)
As I haven't yet handled a DR-680 there's little I can say. Ganging all together sounds like a good option too.
The limiters are implemented post ADC, so they will not prevent clipping the analog input stage or converter. There are circuit diagrams showing that in the service manual as well as a useful diagram showing levels though the system.
That's not good. I think that was mentioned here when the 680 was released, and it did seem like Tascam missed the whole of limiting doing that.
On their comparison list, Tascam does specify all their units having limiters, but I wonder if it's also implemented after the converter on the PS82.
I use the Tascam snug cover which prevents accidental movement of the switches on top, but I do always make an effor to double check them before recording. The top switches can be set to be locked out seperately from the front swtiches if desired, but I don't use that feature.
The Tascam CS-DR680 seems properly designed. Is that the one you have? There's a Portabrace too. You need a case that can open the top, preferably transparent, like the one that was used on the Shure FP-33, where the battery door and some switches were on top.
The short story on external battery powering seems to be not to use a supply that self-senses voltage output needs, but is either single output voltage only or hard-switched.
What I meant was something like this:
http://www.batteryspace.com/custom-nimh-battery-flat-pack-12v-2200mah-10xaa-with-2a-polyswitch-and-thermostat.aspx
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In your opinion. IF I ever want to find out some knowledge about something I take the time to do the research myself. then I might ask some questions. Everything you would want to know about the issue a multitude of people have had...pro and con...is all here in those pages it's just that easy. If you don't want to bother reading them............
There's a lot of information in these threads, but there's also a lot of random chatter. Finding the key information within dozens of pages of posts isn't always easy.
If somebody doesn't want to chime in when somebody asks a question, that's their prerogative, but it's pretty absurd to claim the person asking the question isn't showing respect. Don't want to help them? Don't help them. But often people who have "been there" can give an overview and easily point to the more relevant information. Asking questions is part of doing research.
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Ganging all together sounds like a good option too.
You can also gang more than one group, as long as each group contains adjacent channels.
The Tascam CS-DR680 seems properly designed. Is that the one you have? There's a Portabrace too. You need a case that can open the top, preferably transparent, like the one that was used on the Shure FP-33, where the battery door and some switches were on top.
Yes, it was included as a package deal when I bought the recorder, it's a protective case only, except for the storage pocket on top, but is well designed and allows access to everything including the battery compartment on the bottom. It had a clear front cover and the top flap covers that. I keep the protective front flap tucked under the top (not designed to do that but it velcros in place) and the top flap is then easily lifted without velcro rip noise, or folded around the back and underneath out of the way entirely. I put the whole thing in a larger recording bag containing cables and battery for use.
Can't guarantee anything but that battery pack you linked should be okay, you may want higher capacity though.
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There's a lot of information in these threads, but there's also a lot of random chatter. Finding the key information within dozens of pages of posts isn't always easy.
I gathered up some comments and advice from the various 680 threads and put them in a Word document to help me figure things out. I don't know if this will help or not. I didn't plan on posting it, and not all the threads are attributed (apologies), but perhaps it might help newbies to the machine like me. Here goes:
*****************************************
UPDATE - I tried this: In Menu under REC > ST REC, I selected 'DIN'. Under I/O > INPUT SEL, I selected ANALOG (this refers to channels 5/6, even though it doesn't say so on the display) . Plugged in the S/PDIF out from an outputting DA-P1 , both set to 44.1, and the 680 indicates 'DIN LOCK'. So this looks good so far. But I'm not seeing any indication of input level on the 680 or hearing anything from the headphone monitor.
UPDATE II - OK, I figured it out: Under HOME/FUNC you have to select DIN MONITOR (i.e. un-select MIX MONITOR) to hear the S/PDIF input. Yahoo! 7/8 have mic inputs!
If I read it correctly, go to HOME/FUNC > I/O > INPUT SEL, the 'ANALOG/DIGITAL' option there is for channels 5 and 6. Channels 7/8 of course don't have an analog option and can only be either the 'DIN' stereo track or the stereo 'MIX' track (selectable in 'MENU' under REC > ST REC > DIN/MIX).
If you select 'ANALOG' there, you merely retain the 1/4" analog inputs to channels 5 and 6, but if you select DIGITAL, the digital input then goes to 5/6 instead of 7/8, and (I presume) channels 7/8 default to 'MIX'.
The setting in the DR-680's I/O menu for choosing digital or analog input on 5/6 is bypassed if you select digital for 7/8 by setting ST REC in the REC menu to DIN. If you've got DIN in 7/8, 5/6 will automatically default to analog input.
And while you can route digital to 5/6 and see/hear incoming audio on those channels without having to change the Functions/Monitor setting to DIN, you DO have to change that Functions/Monitor setting in order to see digital audio on 7/8. Squirrelly!
The person reporting that has already recanted. 1.12 does not remove spdif > track 7/8 recording capability. You have to change the monitor settings to see the levels from digi-in because they are reset every time you power it off.
To record on 7/8, you need to go the the recording menu off the main menu, and set "ST REC" to "DIN", meaning digital input (as opposed to "MIX", for recording a mix of channels 1-6 on the 7/8 channels).
This is all you need to record digitally on 7/8. According to the manual, if set to "DIN" on "ST REC", on the I/O menu, it will not matter if you select "ANALOG" on INPUT SEL" or if you select "DIGITAL", either way, the 680 will record the analog channels on 5/6, since you have set it up in the REC menu to record the digital channels on 7/8.
I don't think the DIN MON settings or D OUT MODE have anything to do with whether you can record digitally. They will affect whether you can monitor your digi input channels on the headphone, or output the digi input to the spdif coax output (for a patcher, etc), but I don't think they affect whether you can actually record off the digital input.
For recording the digital input on channels 5/6, the "ST REC" setting on the recording menu must be set to "MIX" (sending the 1-6 mixed signal to 7/8), and then also set on the I/O menu the "INPUT SEL" to "DIGITAL".
Thanks guys! Teddy actually filled me in on how to do this. To record digi-in I needed to set the 680 to Digital and Mix on the I/O instead of DIN, Stereo - Mix on the Main Menu shutting off 1,2,3,4, channels, and Mix Mon on the function screen. I now get the digi-in reading on the 5 & 6 channel, and I can monitor the levels there on 5&6, and the levels display in the mix side of the screen also.
Yes, upon further testing I find that if the Digital Output setting of the SD702 is "Professional" (which their manual also refers to as AES), then the DR-680 can receive and record digital data at either 44.1 or 48k. (If the rate on the 702 and 680 don't match then I see "DIN Unlock" on the 680, which is good practice.)
If the 702 is set to "Consumer", and the 680 and 702 are both set to 48K, signal passes and recording happens OK.
But if the 702 is set to "Consumer", and the 680 and 702 are both set to 44.1, the 680 shows DIN Unlock and won't pass audio.
What actually appears to be happening is that the 702 is outputting, or the 680 is misinterpreting/enforcing, a SPDIF subcode bit that triggers the old "no digital copying at 44.1" copy prohibit protocol.
As for the other things in my previous post: yes, the 680 will record an incoming 44.1 stream from the 702 (with the 702 set to "consumer") when the 680 is set to 48k. But no sample rate conversion is going on. And no indication is heard in the audio that anything is amiss during recording.
But if I then play that file back, the data that streamed in at 44.1k is played back clocked at 48k, so it's higher in pitch and faster in tempo than the original audio. This is bad design practice. If the rates don't match, then DIN Unlock should show up to announce a problem.
Another point to note is that I initially found the monitoring on the DR-680 to be quite confusing, probably because I didn't read the manual closely enough for quite a few outings. My mistake was that I thought the Solo function was what I needed to use to monitor specific stereo pairs, but all that function does is output a single channel to both left and right headphone channels. Instead, you need to use the front panel "Mix Level" button and lower the mix level to zero of any channel you don't want to hear. That's quite a tedious process because if you only want to hear channels 1&2, you've got to manually turn down 6 other channels (or 4 or 2 depending on how many total tracks you've got going). I much prefer the monitoring on the R-44 where a simple button press rotates between 1&2, 3&4, or all four mixed together.
Quote from: achalsey on April 18, 2013, 04:16:59 PM
What signal does the digital out send? Just the digital input, or can it send a mix? Is there any way to send individual channels (analoge out set to "direct") then send two specific channels through the digital out?
The digital out sends whatever monitor mix you set as a panned and level adjusted mix of the 6 inputs. If you set the machine for 'direct' output of each individual channel to it's analog RCA out, set the monitor mix levels to only output the two specific channels you want, one hardpanned left and the other right, with the other channel levels set to zero, it will do what you describe.
Quote
Also, if I set analoge out to "mix", and have a board feed into channel 5/6, then set all the other channels in the mix at 0, I would be have 3 separate outs giving the board feed, correct?
Assuming you don't have very old firmware, you would actually have board feed to all outputs. If you don't pan anything you'd get the identical mono mix of ch5 & 6 on all outputs. If you pan ch5 hardleft and ch6 hardright in the monitor mix, ch 5 would be routed to all the odd numbered RCA outs and ch6 to all the even ones. The digital out would consist of ch5 left / ch6 right.
If you set the analog out to 'direct' you'd get the ch5 input on the ch5 output, the ch6 input on the ch6 output (regardless of monitor panning or levels), and 5 & 6 on the digital out, panned however you set that in the monitor mix.
« Last Edit: April 18, 2013, 05:56:06 PM by Gutbucket »
You can only monitor ch1-6 off the machine. You can only play back either ch1-6 or the stereo channel, but not all eight channels simultaneously. It's really a six track machine with an 'extra' stereo channel and that's where that limitation shows up.
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Service manual can be found in the user manual archive section of this site, posted in multiple sections:
http://taperssection.com/index.php?topic=147560.0 (http://taperssection.com/index.php?topic=147560.0)
http://taperssection.com/index.php?topic=147559.0 (http://taperssection.com/index.php?topic=147559.0)
I have a copy the full manual in a single PDF (~2MB) which I can email you if you like.
Switching between channels to adjust input gain with the pot is fast and easy using the channel selection buttons located under the meters. It's more difficult in the dark by feel, but still doable. Ganging adjacent channels together to adjust them all simultaneously with one knob while retaining there relative differernce settings is well worth having to select using the buttons IME, but I'm not a video guy.
The limiters (and low-cut) are implemented post ADC, so presumably they will not prevent clipping the analog input stage or converter unless there is a feed-back loop to the analog stage which isn't indicated. There are circuit diagrams showing that in the service manual as well as a useful diagram showing levels though the system.
I use the Tascam snug cover which prevents accidental movement of the switches on top, but I do always make an effor to double check them before recording. The top switches can be set to be locked out seperately from the front swtiches if desired, but I don't use that feature.
The short story on external battery powering seems to be not to use a supply that self-senses voltage output needs, but is either single output voltage only or hard-switched.
I know there are at least two version of the top board, maybe three. I found that out the hard way ordering one. I "think" it is based on manufacture date and I can't remember what the difference's were. I have to dig back through my Emails to Tacam's tech dept. And sadly I have a ton of them. I want to say it has something to do with the power input design, but I am not positive after all these years.
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I apologize if I missed it, but I don't remember reading in any of the DS-680 topics how exactly the gain on the DR-680 works. I was just wondering how the math works on the DR-680. Since you can go from -31 to +31 and low/high I am a bit confused how much gain it has.
I also own a DR-100mkII and seem to remember seeing somewhere that the Low-Medium-High button on the back adds 25dB at each step and the dial on the side adds 30dB at maximum level 10. So at "High" and "10" that would give 25+25+30 = 80dB of gain (does that sound plausible?). I usually run my SDC's (MJE-384K) at Medium-6. So that would be about 25+18 = 43 dB (dial at maximum is 10, which apparently is 30 dB. So each whole number is 3dB. For level 6 that would make 3x6=18dB).
Basic testing with my voice seems to suggest that the DR-680 at Low-6 gives about the same as the DR-100 at Medium-6, but I would like to have a ballpark idea of how it scales.
Regards, Christine
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I can't comment on specific numbers, but as I've noted before, if you're capturing low level performances (in my case, acoustic with dynamic mics), it's better to set the switch to High and turn the gain down than to set it to Low and turn the gain up. This is exactly the opposite of what the manual indicates, but the results will be significantly noisier with the switch set to Low and the gain turned up.
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I can't comment on specific numbers, but as I've noted before, if you're capturing low level performances (in my case, acoustic with dynamic mics), it's better to set the switch to High and turn the gain down than to set it to Low and turn the gain up. This is exactly the opposite of what the manual indicates, but the results will be significantly noisier with the switch set to Low and the gain turned up.
So far I have only recorded acoustic instruments (either a 40-person community band or a big band). I don't own any dynamic mics (yet, but turning things over in my mind), but your answer suggests that the DR-680 does have enough gain to deal with these.
I do indeed seem to remember reading somewhere here about anything above Low/+20 it being better to switch to high and that you will find about the same amount of gain something like 15 points lower? So Low/+20 would be roughly equivalent to High/+5?
If I remember this correctly it suggests that the low and high scales have about 3/4 overlap between them. Assuming that each number is 1 dB, then -31 to +31 on low scale = 63dB + additional 15db on high scale = 78dB max gain. Which is pretty close to the presumed 80dB of the DR-100.
If my math on my DR-100 and basic testing between the DR-100 and DR-680 is correct, that comes pretty close. DR-100 medium 6 = ± 43 dB (apparently). DR-680 at low/+6 = -31 to +6 = 38dB. Plus perhaps some kind of offset. With my rough test that is probably close enough to be in the ballpark.
Or am overlooking something?
Regards, Christine
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I just checked using one mic and a splitter. High @ +23 = Low @ 0.
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I know virtually nothing about the technical side of things, so forgive my ignorance...but wouldn't 'unity' just be at zero gain?
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More like giant can of worms...
To answer the question directly, I don't know. The DR-680 gain range is +/- 32 with zero being default. If zero was unity, then negative gain values would be attenuation while above zero is applying gain. I don't think this is the case and you should not be afraid to run below zero if required.
The concept of unity gain has been grossly misunderstood on this forum for years, myself included. It's a false assumption that 'equal in / equal out' sounds best because once you start understanding how opamps work and the basics of preamp design, the notion is immediately dispelled. Typically, opamps perform "best" (lowest noise and distortion) at higher gain ranges and their performance can be much worse at their minimum gain setting. In no way are they a straight wire at low/no gain and could actually be adding significant small amounts of noise.
The concept of Unity Gain took off [on this forum] because of a few recorders with poorly implemented mic/line inputs. Examples include portable DAT recorders like the D7/D8/M1/D100 or the Edirol R-09. I think the R-09 was really when this notion of unity gain went off the tracks. These recorders were simply not built to handle high-sensitivity microphones and/or loud SPL environments and their inputs easily overloaded. For the DAT recorders, the idea was not to run below 3-4 of 10 on the dial and for the R-09 it was not to run below 8 of 100 (or thereabouts) otherwise digital attenuation would occur. When a mic/line stage is poorly implemented like it was on these units, the best solution is to simply apply attenuation prior to the Mic/Line input.
These anecdotes are useful because they give the average user a frame of reference for operating their equipment, but do not imply that unity gain is the setting that produces the best possible sound. The concept of unity gain has been grossly misinterpreted by the entire community, to be honest. There's still the notion that a mic input stage colors the sound whereas a line input stage is completely transparent (i.e. a straight wire) and that that the best sound is produced at unity, which is simply wrong. Both mic and line inputs pass the signal through opamps prior to the A/D and will always affect the sound in some way. It's actually worse running at 'unity' once you understand that many/most opamps perform better at higher gain ranges, have at least some minimum gain (i.e 10 dB), and that in order to get 'equal in/out' you might actually be applying unnecessary attenuation.
The better question to ask is 'what gain value sounds best' rather than 'what value is unity' and the most likely answer to this question is the gain value that gets as close to zero dB without clipping.
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Thanks for the comprehensive answer. I do remember DSatz posting an informative comment similar to yours and was considering mentioning the possible misnomer of 'unity gain' but don't know nearly enough about it all to be of any help.
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^^^ http://taperssection.com/index.php?topic=152494.0 (http://taperssection.com/index.php?topic=152494.0)
Not sure if this is the thread you mean, but it has good explanations of both minimum safe gain setting and the "straight-wire" myth...
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Here's the DR-680 level diagram extracted from the service manual (page 19)-
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Perhaps I would risk using two words to what I think might simplify the 680 level settings: overload and noise. Please do not forget this is an oversimplification I'm doing.
Overload usually happens at the first stage, due to wrong gain settings. In the 680 the only gain settings are High and Low. Dynamic mics usually need High gain. Care should be taken with condenser mics, as some need High gain and others Low gain.
Noise usually happens when levels set by volume pot are mishandled. Shure had to go deep into this with their portable mixers, and explain quite well how to set levels, because users opened their level pots too much, and made a fame of Shure mixers being noisy. Later on they improved on this, by using a pot that set gain and levels together. But people still misused the Master pot. Peak leds helped warn you of overloads, and the 680 also has clip warnings.
I don't think distortion levels are increased if the gain levels are set too low. What does increase is noise levels, which might be considered some kind of distortion too, in extreme cases.
One good thing about the Tascam is that the level setting (Trim) is after just one chip stage, so at least it doesn't add up the noise of another stage. The other good thing is that there is a way, explained in the manual, that shows when distortion is happening at the input or further on in the chain. So it's a question of setting the gain level and see what happens below on the meter, and then setting the volume pot and see that the meter levels don't surpass the upper line.
But..., and on this there might be people who disagree, I believe eventual overload peaks for the first stage won't always be something bad and should help keep noise low.
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I can't comment on specific numbers, but as I've noted before, if you're capturing low level performances (in my case, acoustic with dynamic mics), it's better to set the switch to High and turn the gain down than to set it to Low and turn the gain up. This is exactly the opposite of what the manual indicates, but the results will be significantly noisier with the switch set to Low and the gain turned up.
Just bear in mind that the possibility to overload the inputs exists like this. A mistake I learned the hard way. I had been running things exactly like you stated, and in the midst of a day long event and multiple performers, I had a group and after seeing the levels being too high turned them down. The first clue should have been that I was down to the 1 and 2 range. Since I didn't monitor with headphones it "looked" ok but when I played it back later, there was significant distortion. As you get used to the 680's levels you'll find this easy to avoid. And while the 680 has less than stellar readout for levels, that line around -14 is well placed and works great for a rough mark especially in the 24 bit realm. Nowadays, I almost always have my input levels between -4ish and plus 5ish. If I see that my levels are too high then I back down the sensitivity switch on the top. Using this method I have never had an overloaded input. As for the statement "eventual overload peaks for the first stage won't always be something bad". My expereinces with Tascam recording devices, like the 680 & HD-P2 (which are newer recorders relatively speaking) is that they do not respond well to "overs". And when these recorders do clip they sound ugly and are unforgiving. I'd say they hove no "headroom" but since it's the digital world that is an outdated and highly inaccurate term. The best recordings I have pulled with the tascam decks is to run them at higher input levels, be it sensitivity or trim pots, but not to push then past that.
And the "unity gain" talk...I agree with the posts that say it's misunderstood and too much value is placed on it. I believe this topic became a hot button when there were many more people running external preamps and the goal was for the recording deck to not"add" anything to the signal it was receiving. hi and lo nailed it with 'what gain value sounds best' instead of what is unity on my deck or deck xyz.
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Just bear in mind that the possibility to overload the inputs exists like this. A mistake I learned the hard way.
I should probably run tests at various levels sometime to compare, but it's worth noting that at lower levels the difference in noise isn't subtle. There's a *definite* advantage to recording using the "High" setting, at least within a particular range.
I had been running things exactly like you stated, and in the midst of a day long event and multiple performers, I had a group and after seeing the levels being too high turned them down. The first clue should have been that I was down to the 1 and 2 range. Since I didn't monitor with headphones it "looked" ok but when I played it back later, there was significant distortion. As you get used to the 680's levels you'll find this easy to avoid. And while the 680 has less than stellar readout for levels, that line around -14 is well placed and works great for a rough mark especially in the 24 bit realm. Nowadays, I almost always have my input levels between -4ish and plus 5ish. If I see that my levels are too high then I back down the sensitivity switch on the top. Using this method I have never had an overloaded input. As for the statement "eventual overload peaks for the first stage won't always be something bad". My expereinces with Tascam recording devices, like the 680 & HD-P2 (which are newer recorders relatively speaking) is that they do not respond well to "overs". And when these recorders do clip they sound ugly and are unforgiving. I'd say they hove no "headroom" but since it's the digital world that is an outdated and highly inaccurate term. The best recordings I have pulled with the tascam decks is to run them at higher input levels, be it sensitivity or trim pots, but not to push then past that.
Are you saying your recording was clipped? Or that peaks were below 0dBfs but were still distorted? Because overload in the input stage wouldn't have anything to do with digital clipping (unless of course the trim was set high enough to cause clipping). Are you saying you had the gain set to "High" and the trim set to +1 or +2? And the meters were well below 0dBfs? Or am I not understanding your scenario?
At any rate, digital "overs" and overloading the input stage are two different things.
I just checked using one mic and a splitter. High @ +23 = Low @ 0.
Oops. I guess that should be Low @ +23 = High @ 0.
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Interesting. I don't have my 680 anymore, but seem to recall that the "High" gain setting was considerably noisier throughout the entire gain range and would run on "Low" with the gain cranked up. Maybe I'm just not remembering things correctly.
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Interesting. I don't have my 680 anymore, but seem to recall that the "High" gain setting was considerably noisier throughout the entire gain range and would run on "Low" with the gain cranked up. Maybe I'm just not remembering things correctly.
On mine it is *significantly* noisier on "Low" with the trim cranked up. I can make some clips if anyone is interested.
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Your findings should be right on. If you use Low gain and crank the level pot up, all the chip noise will be amplified by the following chains.
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Interesting. I don't have my 680 anymore, but seem to recall that the "High" gain setting was considerably noisier throughout the entire gain range and would run on "Low" with the gain cranked up. Maybe I'm just not remembering things correctly.
On mine it is *significantly* noisier on "Low" with the trim cranked up. I can make some clips if anyone is interested.
This seems to go against how most recordists use the DR-680.
I use the low gain (MIC IN) for recording live music with condenser microphones.
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This seems to go against how most recordists use the DR-680.
I use the low gain (MIC IN) for recording live music with condenser microphones.
Above a certain input level (between source volume and microphone sensitivity) it would obviously make sense to use Low gain, but for lower levels High gain is definitely a necessity unless you want a noisy recording. Fortunately I only made that mistake once.
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I have considerable noise on high gain mic in. Ruined two separate recordings a couple months ago. Tested it at home through the headphone out and there was a lot of hiss. This is obviously not scientific, but personally will avoid using high gain if at all possible in the future.
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I have considerable noise on high gain mic in. Ruined two separate recordings a couple months ago. Tested it at home through the headphone out and there was a lot of hiss. This is obviously not scientific, but personally will avoid using high gain if at all possible in the future.
What do you mean by "a lot of hiss"?
When monitoring/recording, if you have the input switch set to "Low", and then switch it to "High", you'll definitely hear more noise, but that's only because the signal is now significantly louder. If you level match via the Trim function, that should not be the case.
For example, if you have 2 identical mics plugged into channels 1 & 2, with channel 1 set to "Low" and channel 2 set to "High", and the Trim is set to 0 for both channels, channel 2 will be noisier than channel 1, but it will also be significantly louder. If you then adjust the Trim on channel 1 to +23 so the input signal is at the same level on both channels, there should now be more noise on channel 1.
Make sense?
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General best practice-
Adjust the gains in order: first the high/low switch gross adjustment (first stage), then the trim fine adjustment (second stage). In setting the first stage high/low switch, choose the position which allows a trim setting closer to its +/- zero position. If you find you need to adjust trim towards the extremities of its range and switching the first stage input switch the otherway allow you to keep the trims closer to zero, do that.
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General best practice-
Adjust the gains in order: first the high/low switch gross adjustment (first stage), then the trim fine adjustment (second stage). In setting the first stage high/low switch, choose the position which allows a trim setting closer to its +/- zero position. If you find you need to adjust trim towards the extremities of its range and switching the first stage input switch the otherway allow you to keep the trims closer to zero, do that.
Yeah, I think the only question (one which I haven't really investigated yet) is what the best "switchover" point is. That is to say, I know that High/0 sounds better than Low/+23, but what about High/-10 and Low/+13, or High/-20 and Low/+3?
Is anybody interested in an example file?
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High with a heavy dose of negative trim may well have a lower noise-floor but also reduces headroom though the first gain stage. If the first stage clips then that's usually more objectable than a bit more noise down at the quiet end, partly because if the source is loud enough to clip the input stage, you'd need to use pretty heavy amounts of dynamic compression to bring the noise-floor up into the range of audibility at normal listening levels.
In regards to using the line on the meters as a good level setting target, I find that to be a pretty good safe zone for most things and just shoot for that. I did once see momentary peak clipping of the input stage with the meters otherwise riding the 'line', but that was with on stage mics in close proximity to a drum kit with large transients and a relatively low average level (wasn't close mic'ing the kit but only ~2-3 feet away, the mic pointed at the snare). The clipping indicator lit momnetarily although the meter read on or just over the line.
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I would be interested. I definitely understand what you're saying between high/low difference at the same trim level though.
Here is a track of one of the ruined sets. I just amplified it starting at 40 seconds.
This was AKG CK 63 > Naiant coupling/PFA > (adapters) > 680 mic in, gain on high, trim pretty high, I want to say like over 20. I will say however, that I was running to channels 5/6, 1/4" in with XLR > TS 1/4" adapters. Not sure if that affected it. Someone mentioned the unbalanced signal might have been the culprit?
https://soundcloud.com/achalsey/680-self-noise
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I would be interested. I definitely understand what you're saying between high/low difference at the same trim level though.
Here is a track of one of the ruined sets. I just amplified it starting at 40 seconds.
This was AKG CK 63 > Naiant coupling/PFA > (adapters) > 680 mic in, gain on high, trim pretty high, I want to say like over 20. I will say however, that I was running to channels 5/6, 1/4" in with XLR > TS 1/4" adapters. Not sure if that affected it. Someone mentioned the unbalanced signal might have been the culprit?
https://soundcloud.com/dynamically/680-self-noise
FWIW, I get an error when I click that link.
If you had the gain set to High and the trim at +20, I doubt the gain had anything to do with it. That is, if you still weren't clipping with the trim set to +20, there should have been *plenty* of headroom for the High input.
Now, if you had the trim set very *low*, like -20, I could see the possibility of the input stage overloading.
I'll try and remember to create a sample clip tonight.
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fixed. Also, I wasn't under the assumption of anything overloading. I was just talking about self noise from the machine. It seems very noisy to me with the gain turned up too high.
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I will say however, that I was running to channels 5/6, 1/4" in with XLR > TS 1/4" adapters. Not sure if that affected it. Someone mentioned the unbalanced signal might have been the culprit?
Haven't listend to the sample, but that's likely to be your issue. Besides being unbalanced and more sensitive to interference noise, with one leg of the balanced signal shorted to ground the signal level is reduced by about 6dB which effectively increases the noise floor by that amount (I'm pretty sure). Switch to a TRS instead of a TS plug, the cable remains balanced and you maximise signal to noise.
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fixed.
Thanks.
By "ruined", do you just mean it's noisy? Because while I hear plenty of noise, I don't hear any overload or distortion.
If you had recorded on Low and bumped up the level in post to match, it would have been at least that noisy, if not more so. It seems that the problem was very low input levels from your mics, not anything with the High gain setting on the 680.
This was AKG CK 63 > Naiant coupling/PFA > (adapters) > 680 mic in, gain on high, trim pretty high, I want to say like over 20. I will say however, that I was running to channels 5/6, 1/4" in with XLR > TS 1/4" adapters. Not sure if that affected it. Someone mentioned the unbalanced signal might have been the culprit?
I don't understand this. How could you have been using phantom power adapters with a 1/4" TS adapter? The TS plug would have shorted out the +48V between the sleeve and ring.
As far as my clip goes, I remembered that I had already recorded one:
https://dl.dropboxusercontent.com/s/qdjam3wyepnup4e/SM57_st.flac
SM 57 -> ART SplitCom Pro -> DR-680 ->
channel 1: Low gain, Trim set to +23 (mixed left)
channel 2: High gain, Trim set to 0 (mixed right)
Note that the signal level is equal on both tracks, but there's significantly more hiss on channel 1.
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This was AKG CK 63 > Naiant coupling/PFA > (adapters) > 680 mic in, gain on high, trim pretty high, I want to say like over 20. I will say however, that I was running to channels 5/6, 1/4" in with XLR > TS 1/4" adapters. Not sure if that affected it. Someone mentioned the unbalanced signal might have been the culprit?
I don't understand this. How could you have been using phantom power adapters with a 1/4" TS adapter? The TS plug would have shorted out the +48V between the sleeve and ring.
That....is a good point. I had not considered that. Not sure how it worked, but could obviously be the issue.
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At any rate, that amount of input gain is definitely a lot. I want to say with my AT853as with phantom adapters I set the gain to High and keep the trim around 0, at least for quiet environments. For louder shows I probably switch to Low, but I don't recall offhand. Either way, I wouldn't think you would need the trim at +20 with your mics if they were properly connected.
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From the Tascam DR680 Man-well
When adjusting the input level first set the MIC GAIN to LOW and adjust the level in the TRIM screen. If the level is still too low even when the trim is set at maximum, set the MIC GAIN to HIGH
The one time I ran the trim pots down on and used HIGH gain...I saw the dots below the channel turn to outlines when it peaked above the dashed line on the meter. I assume that was telling me I was about to over load the Mic Gain...never went back to check because it was a opening band I was using to set levels on.
I just started using the Mic Pre on my DR680 having ran a Grace V2 almost exclusively in at Line Level. Prefer the Tascam Pre to my V2 with the Gefell M200's....but...that is not the case with Milab Links or Avenson STO-2's...they seem to benefit from the Clean V2 sound.
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Well that is good to know. I'll chalk it up to user error. Thanks for the impromptu diagnosis.
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From the Tascam DR680 Man-well
When adjusting the input level first set the MIC GAIN to LOW and adjust the level in the TRIM screen. If the level is still too low even when the trim is set at maximum, set the MIC GAIN to HIGH
The one time I ran the trim pots down on and used HIGH gain...I saw the dots below the channel turn to outlines when it peaked above the dashed line on the meter. I assume that was telling me I was about to over load the Mic Gain...never went back to check because it was a opening band I was using to set levels on.
Yeah, the manual says that, but if you use it that way, your recordings may be a lot noisier than they need to be.
If you're recording something loud with sensitive mics, then no, there's no reason to use the high gain setting, but if you're recording something lower in level and/or with mics that aren't as sensitive, there's a distinct advantage to using the high gain setting. Why the manual states that is unclear.
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Hay anyone tried Frank Busman's mod for the 680?
Did it improve things to what level? That is it gets closer to Sound Devices levels?
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I bought one of these this week so I could transfer a few dozen DAT Masters but just hit a wall as I have a handful that were recorded in 32k.
Any work around? Seems most decks only offer Sample Rates of 44.1, 48, 88.2, 96 kHz.
Also the only way I can get my 48k DATS to record on the 680 is Channel 5 & 6 + ST "7 & 8" + MIX. Is this right?
I tried DIN but it is a no go.
I am going Tascam DA-20 MKII > S/PIDF > DR-680
Thanks,
Datfly
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Yeah, that doesn't sound right. You should be able to set the dr680 to 16/48 and record din.
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I bought one of these this week so I could transfer a few dozen DAT Masters but just hit a wall as I have a handful that were recorded in 32k.
Any work around? Seems most decks only offer Sample Rates of 44.1, 48, 88.2, 96 kHz.
I use an old JB3 for 32Khz xfers - the files end up being 16 bit 32 Khz (extra 4bits padded) but it works
32Khz DAT's are recorded in 12bit not 16bit
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When I transferred DATS the DR-680 picked up whatever bit/sample rate it was fed. It's the same when you use an outboard pre-amp AD converter too. The DR-680 locks to the signal it's getting.
I remember using the main menu to choose DIN on the ST channels. To view what's happening on the meters you need to change a setting on the front of the recorder. I think it's a monitor setting?
edit to add:
Maybe Gutbucket will post. He's the DR-680 guru. :coolguy:
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When I transferred DATS the DR-680 picked up whatever bit/sample rate it was fed. It's the same when you use an outboard pre-amp AD converter too. The DR-680 locks to the signal it's getting.
I remember using the main menu to choose DIN on the ST channels. To view what's happening on the meters you need to change a setting on the front of the recorder. I think it's a monitor setting?
edit to add:
Maybe Gutbucket will post. He's the DR-680 guru. :coolguy:
Strange, I just transferred several DATs to my dr-680 and it would not record when the sample rates didn't match. Got the DIN Unlovk message.
Really sucked for DATs with multiple recordings and filler at differing sample rates.
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Sorry, can't help with this one. I've only recorded digital data into the DR-680 from an R44 or a V3 and always at 24/48.
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When I transferred DATS the DR-680 picked up whatever bit/sample rate it was fed. It's the same when you use an outboard pre-amp AD converter too. The DR-680 locks to the signal it's getting.
I remember using the main menu to choose DIN on the ST channels. To view what's happening on the meters you need to change a setting on the front of the recorder. I think it's a monitor setting?
edit to add:
Maybe Gutbucket will post. He's the DR-680 guru. :coolguy:
Strange, I just transferred several DATs to my dr-680 and it would not record when the sample rates didn't match. Got the DIN Unlovk message.
Really sucked for DATs with multiple recordings and filler at differing sample rates.
Is there a way to turn of channels 5 & 6 and only have the ST channel?
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If you push the MULTI/STEREO button on the top of the recorder it switches the machine to two-channel-only mode (which is the only way to record or play at 192kHz). I never record in that mode so I'm unsure of the signal path but I think it always routes to the stereo channel, regardless of how the machine is set to behave in multichannel mode. I'd try that if using it for 2-channel digital transfers.
When in standard multichannel recording mode and recording a digital input to the stereo channel instead of recording it to ch5&6, you can only enter rec/pause if one of the 6 analog channels are also armed to write a file. In otherwords you'd need to record a third channel in addition to the stereo channel, which you could just delete later of course. The only way to have it write only a single file for the digital input when the machine is in multichannel mode is to route the digital input to ch5&6.
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Anyone try transfering files back onto an SD card from a computer for multichannel replay on the DR-680?
I've tried copying the compete directory structure from a card to the computer and back without otherwise editing anything, including the MUSIC and UTILITY folders and the dr-680.sys and dr-680.tak files in the root directory, but the machine doesn't recognize the files. If I then select the folder containg the wav files and execute a REBUILD operation the machine will recognize and play the files, but only sequentially as individual tracks, not simultaneously as mutichannel files which is what I'm looking for it to do.
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Has anyone tried out these Lexar cards in the 680?
http://taperssection.com/index.php?topic=167254.0;topicseen
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I will be testing one in my DR-680 tonight.
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I will be testing one in my DR-680 tonight.
Look forward to hearing the results, Mr. Guinea Pig!
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I have no reason to doubt that they will work. Lexar is an established name in SDHC cards. The cards I got were delivered in professional packaging. I have already checked the storage volume of the cards and they are all the proper size.
I'll report back if I have any troubles with them.
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I have no reason to doubt that they will work. Lexar is an established name in SDHC cards. The cards I got were delivered in professional packaging. I have already checked the storage volume of the cards and they are all the proper size.
I'll report back if I have any troubles with them.
Update:
Those Lexar SDHC cards work fine.
I used one in my DR-680 last night, recording 6 channels each for four bands over the course of 6 hours. No problems.
I also put on in my Lumix camera and took 200 pictures and it work great in that application as well.
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Error log-
Had the 'fluctuation' problem arise weekend before last- the cyclic gain 'pumping' which begins with a fast momentary level bump, followed maybe 15 to 30 seconds later by a level dip and smaller bump every few seconds across all channels until the recording is stopped. This is the typical manifestation of the suspected over-heat field problem, perhaps facilitated by providing only the minimum 9V from an external battery (that’s never been confirmed, and is only my personal suspicion).
It happened for 2 sets out of about 12 recorded over the weekend at a 4 day Americana festival, specifically day 2 of my 3 days of taping. Two late-afternoon/early-evening sets affected both approximately one hour & 15 minutes into the recording. I did not realize it at the time and only discovered it on playback. Deck was either simply stopped or powered down for a while between each set. Whether I just stopped or powered down before, between and after these two sets I can't recall. I can probably tell by looking at the time stamps on the files and checking if there was a file recorded during the intervening soundcheck. The final two sets of that day were not affected, nor were the two sets recorded the final day.
Temperature was warm but not hot, sunny shorts and T-shirt weather. Gorgeous day. Recorder was in its tight-fitting Tascam protective case, placed in a moderately large soft-sided cooler bag along with DVD batteries, wiring, a couple PFAs and a second rig consisting of a DR-2d, plug-in-power preamp and external lithium battery, none of which run hot. DR-680 was the only primary source of heat inside the bag. Lid was unzipped but closed loosely and I threw a t-shirt over it to keep the sun from shining directly on the bag.
Phantom power was on for all 6 channels- Microtech Gefells in 4 channels, Naiant PFAs powering DPA 4061s in the other 2.
I did reach in at some point over the weekend (probably that afternoon) and notice that the underside of the deck was hotter than normal. Looks like I’ll need to provide additional ventilation next time. One new data point on this phenomenon- the cyclic dip rapidly moves across all channels in something akin to a wave. I noticed this because these are surround recordings and on playback the manifestation of the distortion sort of wraped around the room.
Unfortunately affected something like the last 15 minutes of Sam Bush Band and Del McCoury Band (or as Del says Bill Monroe used to pronounced it- “Dale McCurry”)
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So the post above was mostly an informational heads up to other users, with an known work-around ( keep it cool)..
On my active problems list, I’m still hoping someone might have insight on a way to do this-
Anyone try transfering files back onto an SD card from a computer for multichannel replay on the DR-680?
I've tried copying the compete directory structure from a card to the computer and back without otherwise editing anything, including the MUSIC and UTILITY folders and the dr-680.sys and dr-680.tak files in the root directory, but the machine doesn't recognize the files. If I then select the folder containg the wav files and execute a REBUILD operation the machine will recognize and play the files, but only sequentially as individual tracks, not simultaneously as mutichannel files which is what I'm looking for it to do.
bump.
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Error log-
Had the 'fluctuation' problem arise weekend before last- the cyclic gain 'pumping' which begins with a fast momentary level bump, followed maybe 15 to 30 seconds later by a level dip and smaller bump every few seconds across all channels until the recording is stopped. This is the typically reported manifestation of the suspected over-heat field problem, perhaps facilitated by providing only the minimum 9V from an external battery (that’s never been confirmed, and is only my personal suspicion).
Unfortunately affected something like the last 15 minutes of Sam Bush Band and Del McCoury Band (or as Del says Bill Monroe used to pronounced it- “Dale McCurry”)
were you running 9v or 12v?
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9V
Not to rehash the whole thing again, but the consensus is that this problem is heat-realted, not 9V related.
My lingering thought is that a higher volatage supply such as 12VDC might postpone onset though not prevent the manifestation. I simply want to manage the issue in a practical way to prevent it's occurance when likely and wonder if a bit more voltage might reduce current demands and make a difference. It may not make any difference, or it may make it better or worse. It's nothing more than speculation.
The known preventive measure to is to better ventilate the deck, and to stop recording and preferably power down between sets to let the deck cool and to stop the problem and reset things if it does happen.
Mine has been fed nothing but a steady diet of 9VDC DVD-battery power backed up with the internal AA's since birth. I think I used the AC supply once. The AA's in the deck were dead by that afternoon but I didn't get a chance to change them out. I checked to make sure and after that was simply careful to make sure I did not need to hot-swap DVD batteries during a set.
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I wasn't trying to start that whole thing up again, just curious. :) I'll be running in some warm/hot weather soon enough so hopefully there won't be any issues.
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A few all day warm weather festival tips-
Keep it cool with shade and good ventilation (had it shaded, was lacking in optimal ventilation).
Stop and start a new recording at the end of each set.
Power off between sets instead of simply stoping recording and switching off phantom.
Keep an eye on it during warm mid-day sets over an hour long. If you notice the cyclic fluctuation happening (or hear it though headphones if your monitor while recording) stop the machine and restart the recording, provide increased ventilation and maybe fan the machine with forced air for the remainder of the set.
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I recorded this weekend and ran into the same problem... i took the unit out of the case... in the corner on the bottom it was very hot I turned it over and set my iced drink over the spot... we.re talking Phoenix Arizona... maybe a non wet coldpack for in the bag...
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Yeah I was considering something like that. Unfortunately the humidity here in FL makes anything cold turn rapidly wet with condensation, and would only work for a while, but might be a good short-duration strategy out there in the dry desert. A couple of those first-aid cold compress packs that chill down once you crush them may be a good idea to keep in the bag to remedy the problem once it happens I guess.
I'm thinking that adding a large passive heat-sink might be the best solution if it works. I might grab a piece of 1/4" aluminum plate out of the scrap bin at work, shear it to basically the same dimensions of the recorder and place it directly underneath. I should probably loose the form fitting case, or at least open up the bottom of it around the power supply board hot zone area, but I like the protection it provides. Maybe I can fit the aluminum plate inside the case.
Anyone know of any sort of thermal transfer conformal pad, or even some sort of gel which won't make a total mess of things like traditional thermal paste? Might not be necessary for sufficient heat transfer to the large surface area aluminum plate, but woudn't hurt. Thinking something like a metalized foam pad or sponge thing might exist out there somewhere.
Thinking out loud though my fingers here..
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Has anyone from Tascam confirmed that that amount of heat (and the issues associated with it) is normal?
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I looked into then abandoned the gel type laptop heat sinks. I now have a 12v fan I pulled from an old computer. I put velco on it and the bottom right of the recorder. It works, but it does make noise. I haven't been in a situation yet that it was too noisy. But, I've only ever used it three times.
Actually all this talk about that problem has got me re-thinking what to do this summer. If anyone comes up with a good solution, I'd love to hear it.
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Question for those of you who've felt up the 680's hot bottom ;), how localized was the heat in the bottom right corner? Hot there and considerably cooler across other sections of the bottom of the machine?
After some searching there are conformal heat transfer pads (usually soft silicone/ceramic compounds, sometimes called T-pads) designed to fill air gaps in the interface between a heat producing component and a metal heat sink which appear to be much cleaner than thermal grease. I can't tell if they remain intact or melt and make a mess once applied and are for one-time application only, apparently some are designed to melt in place during the first heat cycle which would not be good. Many of them seem to have peal-and-stick adhesive on one side so they could adhere to the aluminum plate and hopefully the other side would conform to the contours of the bottom of the DR-680 but only stick to it very lightly, or just conform and not stick at all. If so, something like that might work.
I'd guess PC builders on this board may be familiar with that stuff.
Checking out the 3M offerings..
Hmm, they also have Heat Spreading Materials, basically copper or copper and aluminum based confromal tapes with thermal adhesive and a black or copper colored top surface. That could do the job on it's own, eliminating the need for the aluminum plate heat sink, simply adhering to the bottom of the recorder itself and spreading heat across a larger area of the bottom surface more evenly. May stick permanently and become a permanent mod to the machine.
http://solutions.3m.com/wps/portal/3M/en_US/Electronics_NA/Electronics/Products/Product_Catalog/~?N=7234426&rt=c3 (http://solutions.3m.com/wps/portal/3M/en_US/Electronics_NA/Electronics/Products/Product_Catalog/~?N=7234426&rt=c3)3M™ Heat Spreading Materials are designed to provide excellent in plane heat spreading for quick heat dissipation on a preferential heat transfer path between heat generating components and designated cooling devices such as heat sinks, fans, heat pipes and metal frames.
Our heat spreading tapes provide robust mechanical strength with good long term stability at higher temperatures, making them ideal for applications such as spreading heat from hot spots on backlights, batteries or PC boards.
Might have to open the case and take a look. Maybe the answer is a conformal T-pad on the inside plus the heat dissapating tape on the outside of the case. I'd really like to find a simple passive solution which lets me rest easy in these situations and doesn't require a fan.
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The heat is localized about two inches from the back and two inches from the right side. It's definitely not over the whole back of the recorder,
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I bet if you cut away the casing you'll find a chip that you could drop a heatsync on to remedy this issue - you need direct contact with the chip for this to work
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I was recording um last night and during the recording the 1st channel started flashing from black to empty circle the level never dropped and the flashing continued throughout the show randomly (I think) any ideas? thanks in advance Bryan
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Remember there are three points of gain control on the way out the headphone jack, if any one of them is too low, no umph- channel level, mix level, phones level. Is your mix level at 100? In the past that caused the same problem for me until I realized what was going on.
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9V
Not to rehash the whole thing again, but the consensus is that this problem is heat-realted, not 9V related.
My lingering thought is that a higher volatage supply such as 12VDC might postpone onset though not prevent the manifestation. I simply want to manage the issue in a practical way to prevent it's occurance when likely and wonder if a bit more voltage might reduce current demands and make a difference. It may not make any difference, or it may make it better or worse. It's nothing more than speculation.
The known preventive measure to is to better ventilate the deck, and to stop recording and preferably power down between sets to let the deck cool and to stop the problem and reset things if it does happen.
Mine has been fed nothing but a steady diet of 9VDC DVD-battery power backed up with the internal AA's since birth. I think I used the AC supply once. The AA's in the deck were dead by that afternoon but I didn't get a chance to change them out. I checked to make sure and after that was simply careful to make sure I did not need to hot-swap DVD batteries during a set.
I picked up a couple of Tekken MP3300 battery packs that have a switchable power output from 3v-14v. Reading here I understand the auto voltage sensing units have wrecked a few DR680s and shoudn't be used at all, but would I be safe to run my DR 680 using the MP3300 and if so, should I set it at 9v or 12v?
If you carry your 680 in a cooler bag, guess you could put one of those cooler ice packs in there, too, or would that cause condensation?
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Glad that was it Chuck.
Never used a Tekken myself. That seems to be the consensus though. I'd guess you'd be fine at either the 9V or 12V switchable setting. I'd probably use 12V as that's more comfortably in it's range, unless you find that makes it run hotter.
Rocks- The cooler bag makes for a functional, inexpensive gear bag. Pockets on the outside, good shoulder strap, and the internal plastic tub is water-tight up to the zippered lid and retains its shape making it easy to arrange and manage gear. Haven't used a cold pack in there, it would condense water out of the air here, and probably warm up before it was really needed. I leave the lid open and put the small umbrella on top or something to shade it an let heat out. It's soft sided which works better as bump protection than cooler insulation.
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Just noting that B&H has the 680 price back down to $440, although they're closed for the holiday for a week. Free shipping!
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The multi-voltage battery I bought from China has not yet arrived.
When it does I will comment about it here.
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So the level meters absolutely suck on this deck, the last show I ran with my Aerco was brick walled & I had absolutely no idea. Besides running really low where should I see my peaks on this turd?
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So the level meters absolutely suck on this deck, the last show I ran with my Aerco was brick walled & I had absolutely no idea. Besides running really low where should I see my peaks on this turd?
Brick walled? What did the meters look like when recording? I've had mine for close to a year and have never had that problem.
I generally try to aim for around the reference line (although I'm forgetting the value at the moment). Also, from the manual:
"When recording, if a signal is distorted at the input stage, the area beneath its meter is highlighted.
If a recording level exceeds 2.0 dB below the maximum level, a mark appears at the top of its meter, indicating an overload."
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I find the meters to be pretty good too. If I get an over (which I think is actually -2dB from actual clipping) every ten minutes or so I figure I got the levels just right.
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Yep no indicator on the channels but the mix levels were high, I was expecting some sort of warning. After the fact I found the the gain was too high on my preamp but to not see an over indicator of any kind on the deck was a major disappointment, at that point why bother with level indicators?
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Yep no indicator on the channels but the mix levels were high, I was expecting some sort of warning. After the fact I found the the gain was too high on my preamp but to not see an over indicator of any kind on the deck was a major disappointment, at that point why bother with level indicators?
I have never experienced any 'brickwall' results with my deck, other than times when it was clearly apparent on the display, as referenced by lukpac above.
I always record 16/44.1, aim to get the peaks @ the line on the display, almost always amplify tracks in post, and never have any issue with noise from the units preamps.
Curious about your dilemma, and wondering what 'mix levels' you are referring to. All I can think of is that your brickwalled track might be a saved mix file, (which is labeled as ...000_st) but I guess that can't be it because I don't think you can record a mix file if you are using 7/8 as DIN. I'm not much help, but also wonder if this was the first time you ran the Aerco with the deck?
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Your mix level doesn't have any effect on your recording. Unless you're recording your mix track as well, but still that's separate from your 'main' recorded channels. Your mix level just governs your headphone out, mix track (if you're recording that) and your outputs (if your output is set to send your stereo mix to each set of outs rather than individual channels).
I guess it's anecdotal, but I've never experienced this issue either. So far I've found some slightly frustrating errors all to ultimately be user error. Not using fresh batteries, not formatting the SD card, not having the right adapters, ect...
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The 680 has alerts for going over zero and for overloading the mic inputs. See page 16 of the owner's manual for what they look like, but if you're overloading the input stage it shows it underneath the level meter of the channel that's overloading by putting a box around the circle at the bottom. It's pretty easy to miss if you're not looking for it.
http://tascam.com/content/downloads/products/43/E_DR-680_OM_vA.pdf
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The 680 has alerts for going over zero and for overloading the mic inputs. See page 16 of the owner's manual for what they look like, but if you're overloading the input stage it shows it underneath the level meter of the channel that's overloading by putting a box around the circle at the bottom. It's pretty easy to miss if you're not looking for it.
http://tascam.com/content/downloads/products/43/E_DR-680_OM_vA.pdf
^ This.
The reference line for the meters is at -18 dB and although the metering isn't as elegant as a traditional LED meter, it's definitely has features to indicate both 0 dB clipping and input stage overload.
Were you running line in? It's damn near impossible to clip the input stage of the 680's line in, so I imagine you were running mic-in with the gain all the way down (like -20)?
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Also, experiment. Run some tests in your living room with your stereo cranked or something.
If there's some combination of things that would result in overload without any sort of indication, it would be interesting to know, but as above, I certainly haven't run into it myself.
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I was running line in from my Aerco, there were no indicators at all to indicate it was clipping or brick walled or whatever. It was really weird, I know to look for the indicator under each channel & there was nothing which is why I was frustrated when I heard the recording. Without a doubt the Aerco was too hot, it was running about 45 on the gain, my fault for not checking it because I assumed it was still at 25db. Just a disappointing experience with the 680 overall based on the meters. I just don't understand how they can go from great meters on all the old decks to this new garbage but I guess that's what happens when you take five years off from taping, new technology right? :facepalm:
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The only unresolvable problem in the DR680, also shared by the DR60 and all other lower priced Tascam recorders, is that the limiter is not in the first stage, as it should be. A poor decision from Tascam.
I wonder how they did that on their top recorder P82, as it is supposed to compete with Sound Devices, and they do put their limiters where they should be.
It's strange that nobody seems to complaint or even mention this (not minor) thing in this or other Tascam forums. Perhaps complaining over this might make Tascam correct that, at least on some of their models.
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I was running line in from my Aerco, there were no indicators at all to indicate it was clipping or brick walled or whatever. It was really weird, I know to look for the indicator under each channel & there was nothing which is why I was frustrated when I heard the recording. Without a doubt the Aerco was too hot, it was running about 45 on the gain, my fault for not checking it because I assumed it was still at 25db. Just a disappointing experience with the 680 overall based on the meters. I just don't understand how they can go from great meters on all the old decks to this new garbage but I guess that's what happens when you take five years off from taping, new technology right? :facepalm:
Gotcha. My guess is you were clipping the input of the Aerco in which case the 680 (or any recorder for that matter) isn't going to give you any indication that the signal is distorted. The closest thing you'll get to a warning is if the levels look 'sluggish,' but they're going to be well below 0 dB and not necessarily clipping the 680s inputs.
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It's strange that nobody seems to complaint or even mention this (not minor) thing in this or other Tascam forums. Perhaps complaining over this might make Tascam correct that, at least on some of their models.
I don't use the limiter, so it doesn't mean that much to me how it's implemented.
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This is what I was thinking also. I have never had an issue with overloading or the meters on the 680.
Gotcha. My guess is you were clipping the input of the Aerco in which case the 680 (or any recorder for that matter) isn't going to give you any indication that the signal is distorted. The closest thing you'll get to a warning is if the levels look 'sluggish,' but they're going to be well below 0 dB and not necessarily clipping the 680s inputs.
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I don't use the limiter, so it doesn't mean that much to me how it's implemented.
Limiters are a must in location film/video recording, particularly on documentaries where you may have unexpected peaks.
In any case they are effective when they adjust the first stage, before distortion or clipping occurs.
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I didn't even know the DR-680 can determine the input signal from the preamp is overloading. I'll have to check that display out on the DR-680 because I allows get nervous when the Lunatec V2 red LED lights turn on. Then, I get home and it sounds just fine.
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I didn't even know the DR-680 can determine the input signal from the preamp is overloading.
Just to be clear, it will only tell you when the input of the 680 (mic or line) is being overloaded. It won't tell you when upstream equipment (i.e. the aerco, in this case) is the cause of signal distortion.
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I thought I was reading and understanding those comments wrong.
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I was running line in from my Aerco, there were no indicators at all to indicate it was clipping or brick walled or whatever. It was really weird, I know to look for the indicator under each channel & there was nothing which is why I was frustrated when I heard the recording. Without a doubt the Aerco was too hot, it was running about 45 on the gain, my fault for not checking it because I assumed it was still at 25db. Just a disappointing experience with the 680 overall based on the meters. I just don't understand how they can go from great meters on all the old decks to this new garbage but I guess that's what happens when you take five years off from taping, new technology right? :facepalm:
As others have asked/pointed out, are you sure the signal wasn't distorted at the output of the Aerco? It's possible that there was nothing wrong with the meters on the 680, and that any distortion was present before the signal hit the input stage.
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I didn't even know the DR-680 can determine the input signal from the preamp is overloading. I'll have to check that display out on the DR-680 because I allows get nervous when the Lunatec V2 red LED lights turn on. Then, I get home and it sounds just fine.
Don't worry too much about the red peak indicator light. From the manual:
PEAK INDICATOR The LED peak indicator, which monitors the signal between the input
and output amplifiers, illuminates the green LED at -14dBu and illuminates the red LED at
+16dB (10dB before clipping).
Anything less than constantly on, or close to it, you should be fine.
Someone was actually kind enough to mock this up for me a while ago. Forgot about it, but thought it was appropriate (though off topic) here.
http://i.imgur.com/ygUdQBv.jpg
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I was running line in from my Aerco, there were no indicators at all to indicate it was clipping or brick walled or whatever. It was really weird, I know to look for the indicator under each channel & there was nothing which is why I was frustrated when I heard the recording. Without a doubt the Aerco was too hot, it was running about 45 on the gain, my fault for not checking it because I assumed it was still at 25db. Just a disappointing experience with the 680 overall based on the meters. I just don't understand how they can go from great meters on all the old decks to this new garbage but I guess that's what happens when you take five years off from taping, new technology right? :facepalm:
As others have asked/pointed out, are you sure the signal wasn't distorted at the output of the Aerco? It's possible that there was nothing wrong with the meters on the 680, and that any distortion was present before the signal hit the input stage.
You are correct, I have no way to verify whether the signal was distorted or not, my assumption was that the Aerco wouldn't have been overloaded at the gain it was set on but obviously something wasn't right in the chain. I'm glad I posted the question here because as always you guys bring another perspective to the situation. While overall I'm still not pleased with the level meters I at least learned through the experience!
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The only time I ran an Aerco (borrowed) into the DR-680 (mine), the Aerco had problems and distorted. The DR-680 faithfully recorded that distortion without overloading. The Areco was subsequenty repaired and apparently works fine now, although I've not borrowed it again. There are no level or clipping indicators on the Aerco (at least the one I used) so there was no way to know it was distorting at the time other than listening to the output.
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If you want good level meters, the HS-P82 is the way to go. Don't even need my glasses to see them. ::) As long as $ is no object.
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Do you if the mic preamplifiers and limiter placement is better on the HS-P82 than on the 680 or other Tascam models?
That is if the preamps are less noisy and if the limiter is on the first stage, as it should be?
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So...my 680 won't turn off. Anyone have this problem before? Took a battery out to kill it and the file I saved on there seems fine. Turned it back on and still won't shut off. Hold is not engaged.
Won't on both internal or external powering.
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First time problem? I hate to be another statistic of a faulty 680, since I really like it, but it a very annoying issue that I don't know how to deal with.
Re-installed the latest firmware. That's about all I can think of to try. Also the hold switch doesn't seem to engage. When I switch it 'on' nothing seems to happen. I've never used it while recording, but seem to remember a message showing up, or at least an indicator, that it was engaged. I could be mistaken in that memory though. I'm wondering if this is part of the reason it won't turn off now.
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First time problem? I hate to be another statistic of a faulty 680, since I really like it, but it a very annoying issue that I don't know how to deal with.
Re-installed the latest firmware. That's about all I can think of to try. Also the hold switch doesn't seem to engage. When I switch it 'on' nothing seems to happen. I've never used it while recording, but seem to remember a message showing up, or at least an indicator, that it was engaged. I could be mistaken in that memory though. I'm wondering if this is part of the reason it won't turn off now.
Welcome to DR-680 support group. Just remember, you're not alone!
So I had the problem of the DR-680 not turning off when it would overheat, but not during normal operation. The hold switch button on the top panel only affects on/off and if memory serves me correct, nothing is supposed to display on the screen when it is engaged or disabled.
The hold button on the front panel is the 'true' hold button for this deck and can lock all front panel buttons, top panel buttons, or both. Notice I used the word 'buttons' as the hold feature will not lock the top panel switches (i.e. mic/line, high/low, and phantom on/off). The top panel switches cannot be locked in any way.
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Okay, thanks. Wasn't sure if there was any indication of the hold being engaged. I've never used it as I've never had problems with buttons being accidentally pushed.
Yeah, mine just simply won't turn off.
I looked through the menus and the 'auto off' feature seems to be the fix for the moment until something else comes along.
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I think I had to pull the batteries once or twice to get it to shut off a couple years ago. Always worked fine after reinstalling them and booting it up again though.
Keep in mind that the hold switch functionality is determined by a menu setting. You might double check how that's set as well as going through the rest of the menus just to double check everything else.
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Thanks Gut. Yeah, while I was scrolling through the menus I took note of what the 'lock mode' was set to. I used the 'restore to factory setting' option as well. Didn't help.
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Well, looks like my 680 is dying a sudden, slow death. Won't turn off when the button is pressed but when recording automatically tries to turn off then freezes after I press stop. Luckily looks the the files get saved but this is ridiculous. :'(
Still basically working but seems to be falling apart. Definitely won't be surprised when it stops saving files.
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Well, looks like my 680 is dying a sudden, slow death. Won't turn off when the button is pressed but when recording automatically tries to turn off then freezes after I press stop. Luckily looks the the files get saved but this is ridiculous. :'(
Still basically working but seems to be falling apart. Definitely won't be surprised when it stops saving files.
you can borrow my DR-2d!
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How old is your 680 and how much use has it seen?
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Thanks John.
I bought here it July 2012 and used it on average about once a week since then. It was like new when I bought it. Seller had bought it in Feb. 2012 and rarely used it apparently.
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So....update, my 680 is now working properly again. ??? I didn't change anything but it seems to be back to normal. No idea why or what happened before, but I will certainly not be surprised if issues arise in the future.
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I have a question I was not able to find with a search of the prior posts concerning using a cigarette light attachment to power the Tascam DR-680.
Is there any reason why I can't fabricate a male 12v cigarette lighter power plug > type "M" barrel plug to power this rascal while in the car?
The outside of the unit says 12v is needed, which by coincidence, happens to be the size of my car battery.
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Just make sure to get the polarity correct and it should work fine. FYI, the Voltage range from the car cig lighter will range from about 11-14VDC. It will be close to the top of that range with the motor running.
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I assume positive polarity, yes?
That is to say, the outside ring is negative (ground) and the inside pin is positive (hot)?
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Yep. Note the little graphic indicating that next to the power input jack on the side of the recorder.
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Thank you Gut Bucket. This recorder gets mobile vehicle use in the field, as opposed to stand alone batteries, and it is a bit of a battery hog when all 8 channels are running.
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Just make sure to get the polarity correct and it should work fine. FYI, the Voltage range from the car cig lighter will range from about 11-14VDC. It will be close to the top of that range with the motor running.
Couldn't going above 12 V be dangerous? A bunch of 680's got burned out using the Tekkeons that supposedly automatically selected 12 volts and apparently went a bit higher at times? Just asking.....
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I'd think it would be okay but I can't guarantee it. Then again, I just checked the manual and about all it says is don't plug anything in there except the Tascam AC power supply. Maybe better not to do it.
I've run mine for hours in the car on long drives home from recording powered by batteries I recharge with the power from the car. I've never plugged it in directly.
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Since a new 680 is still over $400 and given the number of reports on TS of 680s apparently fried by external power, I would be reluctant to be the first guy to try powering one from a car cigarette lighter outlet. I thought there was a post here by a member warning not to go over 12v, but haven't looked for it specifically.
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My new AA batteries meter, on the average, a tad over 1.6v, which would make the actual voltage, via AA battery power, about 12.8v.
Also, if the vehicle is not running, the actual voltage would be 12.v from the car battery. Only when the generator is running would voltage into, but not necessary out of, the car battery, exceed 12v. I'm not sure the voltage from the car battery fluctuates as suggested. Indeed immediately downstream of the battery, I have a voltage regulator which smooths out the voltage for the rest of the car. I'd have to put a meter to the cigarette lighter to verify.
But you have my curiosity going here, so I'll probably meter the cigarette lighter when the car is off and on. I would expect the same voltage, but I could be wrong.
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The alternator's voltage regulator typically has a set output of ~14V, which is what a meter across either the battery terminals or the cigarette lighter terminals should read with the engine running. With the engine off it will read around 12V to start and you probably don't want to run it down to much under 11V. You could use a cigarette lighter power adapter with a resistor bridge in it to drop the voltage a couple volts. I used to have a Radio-Shack one with switchable voltage output from 3-12v, and the power resistors in it would get slightly warm in use. Had to remember to unplug to not run down the battery when through using it, but it worked fine.
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11.9v static with engine not running. A whopping 14.8v with engine running.
I will pass on using a cigarette lighter. I do have a power inverter, which converts 12vc to 110v and I'll just use the Tascam AC Power block. It's terribly inefficient, but more efficient than blowing up the Tascam.
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Is there a way to mute the speaker and headphone outputs on a 680? Mine has decided not to play thru either ....
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No mute switch in the menu that I recall. You can either simply turn down the headphone/speaker output-level knob on the front panel, or go into the monitor menu and turn down the individual channel outputs, or main stereo mix output, or both.
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I think I asked this a while back but can't find where but with the 680 how can you check how much space is left on the memory card? Thanks
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Menu > Card > Information, I think...
The section in the menu where you format your card (last section in the menu) also has an 'information' option.
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Anyone tried a 64gb sd card in a 680? If so did it work? Thanks
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Anyone tried a 64gb sd card in a 680? If so did it work? Thanks
The 680 doesn't support SDXC.
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I'm going to post this over here since I'm pretty sure this a 680 issue. Anyone had issues with the phantom power or seen anything similar to what happened in the post linked below?
http://taperssection.com/index.php?topic=168966.0
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I'm going to post this over here since I'm pretty sure this a 680 issue. Anyone had issues with the phantom power or seen anything similar to what happened in the post linked below?
http://taperssection.com/index.php?topic=168966.0
What was the air temperature at the venue? Hot?
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I'm going to post this over here since I'm pretty sure this a 680 issue. Anyone had issues with the phantom power or seen anything similar to what happened in the post linked below?
http://taperssection.com/index.php?topic=168966.0
I've (fingers crossed) never had that problem, but I believe others have mentioned it in the various 680 threads.
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I'm going to post this over here since I'm pretty sure this a 680 issue. Anyone had issues with the phantom power or seen anything similar to what happened in the post linked below?
http://taperssection.com/index.php?topic=168966.0
sounds like it is time for a new recorder! I'll be home next week if you need to borrow the DR-2d
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I know! :facepalm: It's really annoying. Problem is I don't have anything to sell to help fund a replacement. I'm just going to wait it out for now.
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I know! :facepalm: It's really annoying. Problem is I don't have anything to sell to help fund a replacement. I'm just going to wait it out for now.
Was it really hot out?
I've have that problem when running the DR-680 when it's really hot out. The bottom, back, right side of the recorder gets very warm and produces those spikes. Now, I make sure that area is not covered and is open to the air and I haven't seen the issue re-appear.
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I've have that problem when running the DR-680 when it's really hot out. The bottom, back, right side of the recorder gets very warm and produces those spikes. Now, I make sure that area is not covered and is open to the air and I haven't seen the issue re-appear.
Do you have your 680 in a bag of some sort? Or just in the open?
I can't say I've been in any super hot environments, but my unit has always been in the open (and powered via AC, FWIW), and I (again, fingers crossed) haven't had any issues, nor noticed any excessive heat.
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I always run it in a big camera bag, but I keep the area on the bottom clear and open to the air.
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I've had it happen a few times with the recorder in the Tascam form-fitting cover, inside the big bag while recording all day out doors. I've been meaning to at least modify the snug Tascam case to let heat escape more easily from the power supply bottom corner. Maybe even open up the machine to look at the possibility of modifing the recorder housing for better heat dumping, possibly even adding a bigger heat-sink, etc.
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The problem manifests on my machine as a sudden and obvious momentary level increase across all channels immediately followed by a longer level decrease, then 30 to 40 seconds later it then settles into a cyclical momentary level decrease with a period of a few seconds. It's happend three times total. Last time I had 6 channels of phantom turned on- two PFAs, four MGs. On playback from the machine that cyclical level decrease moves across channels like a progressive wave. You can see it move across the meters from one side to the other. On playback of my surround recordings directly from the machine the level change rapidly wraps around the room from speaker to speaker.
[edit] I expect the meters will exibit that same behaviour when the problem is initially occurs while making recording, but in all three cases I wasn't watching them at the time and didn't notice it.
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Thanks all! I was inside at night, and it was warm but wouldn't say it was particularly hot.
It only happened on the channels I was running phantom on. Had mics > V2 > 680 (1/2) and SBD > 680 (5/6) that luckily weren't affected at all.
I bought a cheap-o portable USB fan that I'll stick under the 680 to get some air flowing and see if that helps. At least this seems like a manageable problem.
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Different question: what are the voltage tolerances again?
I'm running one of the naztech batteries on the 9v setting. If the battery drops some voltage for a split second randomly (is that possible?) could that kill the recorder? (The internal batteries died and I'm an idiot and didn't change them).
Had a bit of an issue and just lost a set. :(
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Not kill as in damage the machine, but maybe kill as in shut down or glitch the recording if the internal batteries weren't available to bridge a power intermittancy from the external battery.
Bummer about the set.
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Yeah that's what meant, sorry. I switched the battery up 12v. Hoping to doesn't drop out again.
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Well that answers my question! Just as I was going to ask how everyone is doing running their 680's on 9v. Guess I'll stick with 12v just to be safe.
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Guess who uses a DR-680 for their field work:
https://www.youtube.com/watch?v=hsczXAVCF9o
I was at Pickathon all weekend, and caught a couple of these sets, and was pleased to see them running all their mics straight into the Tascam.
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Can anyone help me understand why the mix level to the right of the channel levels on the display does not match? Last night was a perfect example, I ran two stereo pairs of microphones so four channels. The mix level would be over the dashed line while all four channels were under the dashed line (btw I still think these level displays SUCK, I can't believe how useless they are) I checked the mix levels & all four channels were set at 100, the recording sounds absolutely fine but I still can't wrap my head around why the mix level display is running hotter???
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Mix display is showing the sum track it. Therefore a single input could be fine for levels, but when you combine the two individual sources on the sum track with no mix adjustments, it can go over.
Not a big deal since you're recording each channel discretely.
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Mini-Me S/PDIF will play fine with the digi in, correct? Or I guess the AES for that matter?
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Mini-Me S/PDIF will play fine with the digi in, correct? Or I guess the AES for that matter?
Anyone have experience running a 680 with a Mini-Me? I should have it next week to play with, but was just curious. I'll have to teach myself all over how to even use the digital input. I've only used it a handful of times over the two years I've had the thing and always forget what settings to put it on to accept a signal. Now that I think of it...I should save the instruction book to my phone...
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I've only run one 9V DVD at a time. Maybe 4 hrs or so? When it comes time to swap DVD batts it switches momentarily to the internal AA's.
Haven't used the Mini-Me, but on the DR-680 you can record the digital input to either ch's 5&6 or the stereo channel. I always use the stereo channel. In the menu you'll set the stereo channel to DIN instead of MIX. The screen will say DIN UNLOCK or something when you put the unit into REC/PAUSE unless it senses a digital input, that message will go away when it locks to the digital input OR you push the button in the center of the jog wheel (at which case it reverts to the internal clock).
Hope that helps.
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I ran AA's in the 680 once and they drained at break neck speed. I run two 9v wally worlds via a parallel power cable. One battery is at one green after a 4hr show with the other still full. Is that normal? Will one drain before the other? I haven't run only one but suspect it would fare well in that time frame.
Yes usually one drains over the other, not sure why. Maybe it takes from the weaker source or something
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Thanks for the refresher Gut. I guess some of the point of the digital preamp is to open up the extra channels so will probably run the stereo channel as well.
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I ran AA's in the 680 once and they drained at break neck speed. I run two 9v wally worlds via a parallel power cable. One battery is at one green after a 4hr show with the other still full. Is that normal? Will one drain before the other? I haven't run only one but suspect it would fare well in that time frame.
Yes usually one drains over the other, not sure why. Maybe it takes from the weaker source or something
Same experience here. Different internal resistance?
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680 question -
When you run mics directly into the 680, what do you set your mic gain on? I've been told high is the setting - just curious if this would vary based on source and mics used...
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680 question -
When you run mics directly into the 680, what do you set your mic gain on? I've been told high is the setting - just curious if this would vary based on source and mics used...
It will vary based on mics (i.e., dynamic vs. condenser), source (loud rock vs. soft acoustic), and placement (close vs. distant).
The manual suggests using the low gain setting unless more gain is absolutely necessary, but if you have to turn the trim up more than, oh, maybe +13 or so, it's probably better to use high gain and turn the trim down, as there will be less noise. The trade-off is there's an increased possibility of clipping.
I've tried to make notes on what settings I use with various mics in various situations, but I usually still end up adjusting things each time.
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To piggy back on that...is it okay to go line-in directly into the 680? Or is it best to use mic-in when not using an external preamp? I've used line in mics > 680 with perfectly fine results (I think...) but thinking about it now that might not be the best idea.
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To piggy back on that...is it okay to go line-in directly into the 680? Or is it best to use mic-in when not using an external preamp? I've used line in mics > 680 with perfectly fine results (I think...) but thinking about it now that might not be the best idea.
With a line level source? I've done it a lot, usually from a board/mixer output or insert point. No issues that I've noticed. Can't say I've ever done any sort of measurements, but I've never noticed any sort of issue (like with noise on the "low" mic setting) that would lead me to do so.
The only "issue" I've had is sometimes I can't for the life of me figure out why the level on a mic channel is so low, until I (finally) realize the switch on that channel got flipped from mic to line.
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Use whichever input sensitivity switch position (mic/line) gets your levels closest to where you want them with the input trim set to the middle of its range, then fine tune the level using the trim control. If required, phanom power is available at either input sensitivity setting.
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Did you still have phantom power? I would think that would be the main reason for mic in setting.
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It switches input sensitivity only. P48 switching is separate and independent.
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680 question -
When you run mics directly into the 680, what do you set your mic gain on? I've been told high is the setting - just curious if this would vary based on source and mics used...
I've always ran mine on low mic gain with the input on mic. I've never used it on acoustic music though, only loud electric bands.
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That makes sense to me. I know my 184's ran hot
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Most soundboard feeds I grab are fairly hot when using line in and low gain setting. Hard to imagine how hot it would be with mic in and low gain. I would expect some overloading using the mic in unless you have a really attenuated soundboard or outboard preamp level. I only run line in for normal level line sources.
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Most soundboard feeds I grab are fairly hot when using line in and low gain setting. Hard to imagine how hot it would be with mic in and low gain. I would expect some overloading using the mic in unless you have a really attenuated soundboard or outboard preamp level. I only run line in for normal level line sources.
Just as a note, the Low/High setting doesn't matter when Line is selected, only Mic.
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How do you record digital to tracks 5/6 again? On my phone and can't open the PDF of the manual.
A dude is trying to get a patch so I need 7/8 open if he wants a mix of the 4 channels I'm recording right?
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How do you record digital to tracks 5/6 again? On my phone and can't open the PDF of the manual.
A dude is trying to get a patch so I need 7/8 open if he wants a mix of the 4 channels I'm recording right?
Should be I/O -> Input Sel: Digital.
From memory, the digital out will be exactly what you're hearing through the headphone output, regardless of what is going to 7/8. So 1) if you want something from the digital in as part of a mix, yes, you'll need to go to 5/6, and 2) don't switch the monitor between mix and DIN while recording, otherwise the digital output will change.
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OK, thanks. I'll give that a shot.
Might not matter as I realize I some how took the appropriate cable he needs out of my bag. Not sure why I did that.
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OK, thanks. I'll give that a shot.
Might not matter as I realize I some how took the appropriate cable he needs out of my bag. Not sure why I did that.
Any RCA cable will likely work. Worth a shot anyway.
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He needs rca to 1/8th, which for some reason I don't have at the moment. Used to have two in the bag. For a recorder with only rca outs I feel like I really should have the right options.
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Digital in question? (sorry if it's been covered before)
I just got a new USB Pre2 and can now (finally) use the ditial in on my 680.
I have attempted to thry this out at home.
I set the 5/6 to I/O DIN and seem to have gotten a DIN Lock indicator- this is good correct?
also I thought I could route digital in to 7/8, but perhaps not?
When sending a spdif in from the USB- Pre2 I go not levels on 5/6 but on the output stereo mix and in the headphones in could see and hear levels - is this correct?
I tried to record about a minute of music to test but have yet to listen.
Any insights are indeed a great help.
Thanks in advance
--ian
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I haven't really used the DIN recording on 5/6 to know how the levels are displayed offhand, but if you want to use 7/8, you need to change the setting for the stereo track from mix to DIN.
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I haven't really used the DIN recording on 5/6 to know how the levels are displayed offhand, but if you want to use 7/8, you need to change the setting for the stereo track from mix to DIN.
ahhhh that makes sense... I have always run in stereo pairs for ease of editing.
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I haven't really used the DIN recording on 5/6 to know how the levels are displayed offhand, but if you want to use 7/8, you need to change the setting for the stereo track from mix to DIN.
ahhhh that makes sense... I have always run in stereo pairs for ease of editing.
I only record digital in to the stereo-channel, which is separate from the main 6 channels (effectively 7/8). In that case, the file format for the 6 main channels can be either mono files or stereo pairs. Regardless, the stereo channel is always recorded as a single stereo pair file. Not sure, but if you want to record the digital input to 5/6 and set the machine up to do that in the menu, I think it may also be required to set the file format to stereo pairs instead of mono.
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I haven't really used the DIN recording on 5/6 to know how the levels are displayed offhand, but if you want to use 7/8, you need to change the setting for the stereo track from mix to DIN.
ahhhh that makes sense... I have always run in stereo pairs for ease of editing.
I only record digital in to the stereo-channel, which is separate from the main 6 channels (effectively 7/8). In that case, the file format for the 6 main channels can be either mono files or stereo pairs. Regardless, the stereo channel is always recorded as a single stereo pair file. Not sure, but if you want to record the digital input to 5/6 and set the machine up to do that in the menu, I think it may also be required to set the file format to stereo pairs instead of mono.
yes, I usually record in stereo pairs... mostly only 1/2 & 3/4 or sometimes 1/2 and 5/6 but now that I can go spdif in just looking for advice on best method. .
I still have not popped in the file I recorded from the USB pre2 > 5/6 or 7/8 or whatever it did...
no levels could be seen in 5/6 though they were engaged, and level in the stereo mix perhaps 7/8?
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Hello,
This is my first post. I'm a film sound designer based in Eastern Europe and I've been looking into the TASCam DR-680 for more than 3 years now, but all this time I focused on investing in my sound studio. For 2015 I need to get a portable work-horse, a reliable, versatile 12 channel quiet, clean, detailed recorder capable of recording at least 12 tracks, and also 2 tracks in 192Khz for SFX work, and I'm pretty set on buying two DR680's.
So for the last couple of months I've been reading all I can find about it, watching reviews and videos of comparison tests.
I've read all 2138 posts in this looong thread - feels like a novella :)
However there are still some things which are not clear to me, so I will make a short resume of what I don't understand clearly, while also writing down what I feel is important for me to learn about this recorder.
It will be a long thread I guess and I hope somebody else also benefits from it.
Major issues with these devices are reported to be:
A. File loss/corruption when decks loses power while recording, problem aggravated by not using a supported SD card freshly formatted in the unit itself.
B. Badly designed power source, in more than one way:
B1 Improper voltage regulators, no fuse for over-voltage leading to bricked units with some voltage auto-sensing batteries.
B2 Dual-powering system glitchy, sometimes switching from external battery to internal AA's even if those are almost drained, also leaving the "AC in" icon turned on on the display.
B3 Bad design leading to serious overheating of the power source (lower back side of the unit?) causing random glitches when used with Phantom Power ON in heat (random popping sounds on recording, cyclical volume pumping, channels disappearing, decks freezing) especially if set in a bag (even the recommended CS680). This was reported only with 9V "DVD" external battery as far as I know.
C. With some microphones, the DR680 is reported to have unacceptable amounts of preamp noise (+12dB) when the phantom power is set to ON, there is even a mod indicating how to get rid of this problem - apparently the filtering capacitor of 0.5 uf (2x1uF in series) used to filter the phantom power just at the node before the feeding resistors of 2x3.3 kOhms is much too low to eliminate the audio noise
generated by the internal 48 V converter, so replacing those capacitors is needed for a 12 dB preamp noise improvement with certain mics.
D. Susceptible to electromagnetical interference, some users reported hearing high frequency pop-like spikes overlapping on the recording when certain batteries were close to (not even plugged in) the recorder.
E. "The only unresolvable problem in the DR680, also shared by the DR60 and all other lower priced Tascam recorders, is that the limiter is not in the first stage, as it should be" - quoting
Carlos E. Martinez, Reply #215
F. Badly designed speaker - hissy, impossible to turn off except with a jack plugged in the auto-sensing Headphone-out jack
G. Badly designed headphone amp - very noisy, digital volume control unresponsive, lags behind and sounds like a zipper file, without zero-crossing detection
H. TRS Jacks 5/6 can cause popping on recording (one unit here had such a problem, was fixed with a contact cleaner called Progold)
I. The level indicators are weird - showing "0" where "-2dB" actually is. Also they don't show the actual value in dB. and only have a single reference point.
J. The unit will drain internal AA-sized batteries even when powered down, which could lead to batteries leaking if forgotten in the recorder for an extended period of time. However, there appear to be two ways of powering down the device: a stand-by mode presumably entered by pressing the power button once, and a turn-off mode which implies pressing and holding the power button for three seconds, waiting for the unit to shut down, and then engaging the lock switch. If properly powered down, the internal batteries should not drain.
K. Support with TASCAM is pretty bad, they apparently want to have 3 tries at repairing the unit before replacing, which could mean up to two months for somebody in the states, but a lot more for someone like me, living in Eastern Europe.
Except for a certain mr. Chang, there's nobody at TASCAM who's willing and knowledgeable to help with troubleshooting.
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Before ending, I'd like to say that I know many people have a love-hate relationship to the 680.
Perhaps some people who spent 4000$ on a Sound Devices unit think they have the right to say the 680 is not a reliable machine, but I don't think it's fair. In comparison to the SD, yes, they may have a point, but this is mostly to the fact that the TASCAM is not as fool-proof as the SD is. Yes, I meant fool-proof. Film recording mixers who work 16 or 18 hour days recording take after take in freezing cold/blistering heat/God knows what weird conditions will know what I mean - even the best professionals are prone to errors, especially when tired, and SD have designed and built their products with such situations in mind. Their price reflects this. However, I have witnessed cases of SD 744, 788 and even the new 664 freezing and needing reboots. No system is perfect.
Keeping this in mind, there was a list compiled here on this thread with good and bad 680's, and also stating the malfunctions and the apparent causes. I don't think the list has much importance, because the world of tapers is very small in comparison to the hundreds (thousands?) of units produced and sold, and also people who have working units don't tend to google forums and praise their units, but in comparison, the people whose units malfunction (including due to operator error!) google forums and rant about how bad these units are and how crappy Tascam is.
I've never seen a driver who drove a red light and crashed blame the car manufacturer though.
Hope I didn't bore you.
And Happy Holidays :)
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Some questions for the owners of DR680:
1. Cascade function. From what I gather, a SPDIF cable between two units will synchronize the transport of the slave unit to the master unit, by recording the mix of the master DR680 on ch 7/8 on the slave unit. Is it possible to cascade 4 units like this, in order to get 6x4 analog ins?
2. If I understood correctly from what I read on this forum, cascading using SPDIF cable will NOT also synchronize the internal clocks of the units. Is this true?
Does it need a "band-aid" in order to achieve a synchronized recording? I understand that I need an ADC with AES/SPDIF out which will slave to the external AES/SPDIF of the Master DR680, correct? I have read that somebody here uses a SD Usbpre2 for this, but then in such a case, the master unit will not control the REC START of the slave unit, correct?
3. What's the cheapest and most reliable way of syncing two recorders while ALSO using the extra digital in on both units? The SD USBpre2 is expensive.
4. Did anyone here ever do a comparison between a stock DR680 and a Busman modified DR680? This question has been lurking around the forums for 4 years now and nobody answered it. Such a comparison would be useful for me in quiet recordings with sensitive mics (e.g. Rode NT-1A)
5. What is the quietest way of powering this unit? Single power source, internal AA's 1.5V? Internal NI-MH 1.2 Eneloops? Just AC adapter? Just 12 V LiIon battery? Or dual power-both AA's and 12V li-ion?
External battery is pretty problematic (manual states use AC adapter only), support guys say use 12v only, there was a post back somewhere which stated 8-13V, 600mah minimum, I know many people use 9V "DVD" batteries, and I also know that some batteries cause electromagnetic interference.
Due to its low price, it's not set in a cast aluminum body and is not properly insulated from EM interference. I understand that and can work around it.
I know such sources of noise are not an issue for taping live shows but I sometimes need to record and layer quiet sounds, sort of like the ticking of a hand-watch, or soft gentle brushes on hair, so any preamp noise must be minimal, because it really adds up if there are 3 or 4 layers of such sounds.
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Checking in on thread. Just bought a refurb unit from retail section. I'm sure I will have some questions upon receipt of this unit. I've gone through all 6 threads in anticipation but I'm sure something will perplex me when it is in hand.
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One question of of the top of my head, is can you run CA-14/UBB into the DR 680? If so how? Do you need PFAs or will the UBB be enough to power the mics? I would assume that the CA-14/UBB would just need a mini > XLR adapter, I could be wrong on the difference in voltages though.
Would I need the same type of adapter cable to run my tinybox> DR680?
Thanks for any help.
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One question of of the top of my head, is can you run CA-14/UBB into the DR 680? If so how? Do you need PFAs or will the UBB be enough to power the mics? I would assume that the CA-14/UBB would just need a mini > XLR adapter, I could be wrong on the difference in voltages though.
When running my CA-14/9200 > DR680, I use a mini (out of the 9200, assume UBB would be no different) > 1/4' dual cable into two channels on the 680, set at line in/low/phanton off. works fine.
Usually start off with Rec Trim for those channels at 0, and adjust with 9200 gain. Small adjustments then made with 680 Trim as neccesary on the fly.
edit to add - I just realized your posting about the UBB, which provides no gain. But it does power the mics. Use the Rec Trim to add gain. You might experiment with channel gain (low/high) on the top of the 680. You may have to add gain in post.
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That is the good news I wanted to hear, thank you!
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Hello,
This is my first post.
I've read all 2138 posts in this looong thread - feels like a novella :)
:)
Sir, you have my respect.
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Can someone clarify for me if a Tekkeon 3450i will work to power the 680 if the dip switches on the back are set for 12v?
I have Lenmar 9v DVD batteries, Tekkeon 3750, Tekkeon 3450i, Tekkeon extended battery pack, and a slew of Tekkeonn 3300s. I would assume I could use any of these set to 12v but I want to make sure before I connect a battery to this unit when it arrives this weekend.
TIA!
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I run my 680 on 12V. Never tried to run it on 9V.
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Checking in on thread. Just bought a refurb unit from retail section.
well I guess there is one thread I no longer have to re-find (as it was tempting...)
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9V DVD batteries work (what I use). 12V is optimal, but if the battery-pack is capable of multiple voltage outputs and 'autoselects' which voltage is used, double check that it hasn't been reported to have caused issues. Batteries that have selectable output voltage which is manually switched or uses a dedicated 'hard-wired' output for 12V are safer. A straight 12VDC battery without regulation is safe.
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I know it was mentioned before, but can I run re-chargeable batteries inside the unit and also plug a DVD battery at the same time without any worries? Thanks!
Barrett
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Yep. Always a good idea to keep internals in when using an external battery in case the connection somehow gets interrupted.
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Hey Barrett welcome to the multitrack fold!
Beware the internals do not last much beyond a set or so..
(No TBF for me this year. Time an budget is going to renos...)
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Yep. The recorder will switch to the internal AA's if the external battery gets dies, if the connection is intermittent or gets disconnected, or while swapping external batteries.
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I have noticed that the back light stays lit when using external power and will turn off while using internal batteries.
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When using an external power source, the assumption it that the recorder is being powered by the AC supply providing an endless supply of power, so the backlight remains on. When using the internal batteries the back-light switches off to conserve limited battery power.
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I have noticed that the back light stays lit when using external power and will turn off while using internal batteries.
This is one of my favorite features. This is true also on the Tascam HD P-2. There have been a few times where my backlight went off which immediately alerted me that something happened to my external power source, like the cord got yanked or the external battery died. Since the internal batteries were then working, it gave me the opportunity to fix the issue and not lose a note. It's a great feature.
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is there a new model. Picture on tascam main page. (Red Handles )
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Yes!
5 minutes ago there was only the picture of the DR680mkII on tascam's homepage, now the new unit's page is up:
http://www.tascam.com/product/dr-680mkii/
Looks like they solved at least some of the problems: better battery life, better capacitors for P48, better mic pres (sounds interesting), better internal quartz, not so picky about SD cards & SDXC support up to 128 Gb, file take nr. can now be reset, dual level recording... and that seems to be it unless I'm missing something.
However operating temperature is still 32-104F.
I wonder what the price will be.
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Yes!
5 minutes ago there was only the picture of the DR680mkII on tascam's homepage, now the new unit's page is up:
http://www.tascam.com/product/dr-680mkii/
Yes indeed! Sold!
Regards, Christine
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Disappointing. I was hoping it would be a significant functional upgrade. Something like maybe eight analog inputs and the wifi control. Cool about the better pre-amps, although I feed it line sources 98% of the time.
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Improvements from the original DR-680-
Playback of multitrack recordings
The DR-680MKII can load multitrack audio files. Multitrack materials (multiple mono files) prepared with a DAW, for example, can be played back.
Most here won't care about this, but it is huge for me. One of the more frustrating things with the original 680 has been the inability to reload the files originally recorded with it onto an SDHC card for multichannel playback from the machine.
What I would also like to see is a few basic DPS routines for playback-
EQ
polarity
delay
compression
and the ability to use more than one of those effects simultaneously.
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Improvements from the original DR-680-
Playback of multitrack recordings
The DR-680MKII can load multitrack audio files. Multitrack materials (multiple mono files) prepared with a DAW, for example, can be played back.
I was thinking of your posts when I read about that function.
Improvements from the original DR-680-
What I would also like to see is a few basic DPS routines for playback-
EQ
polarity
delay
compression
and the ability to use more than one of those effects simultaneously.
A firmware update could add at least some of these, delay would be helpful even for recording (the old DR40 has it), but I suspect it might not be possible to have them simultaneously unless they upgraded the hardware too.
It would have really made my day if they would have added the possibility to use as USB sound card. Zoom has been doing this in their recorders for ages, and Edirol does it with the R8.
This way, you could use any laptop with a freeware daw to play back 5.1 recordings, add any effects you want, and play them back via RCA to any 5.1 reciever.
I wonder if they upgraded the headphone amp, there's no mention of it.
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Good call on both.
Add 6-ch USB DAC interface capability and an upgraded headphone amp to my wish list.
Actually the only thing that bugs me about the headphone amp is the annoying stepped gain when adjusting the un-stepped knob. They could just switch to a stepped adjustment knob and I'd be content. It could be better of course, but can live with it as it is fully capable of driving my 600 ohm cans to an appropriate volume.
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Of course this comes out 2 weeks after I bought my 680. :facepalm:
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Been wondering if the DR-680 and HS-P82 would get mkII revisions. One of those is answered. Seems likely to me the HS-P82 would be coming as well.
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DR-680mkii = lipstick on a pig
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It's got some cool new red knob protectors :P
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Mmmm, pig.
I see it not as lipstick but as bacon strips.
Wonder if the knob stays on better. Mine keeps popping off, although I've yet to loose it.
Probably just jinxed myself by saying that.
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My question is, will there be a firmware update for the DR-680? Seems like most of the upgrades on the mkII are firmware related, but then again, what do I know? ???
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DR-680mkii = lipstick on a pig
You're the unlucky one who had 2 or 3 units malfunction, right?
The <<lipstick pig>> might not be so bad though, bot only If they keep the street price at the current level (as low as 400$).
The truth is that, according to their own statements, they only really upgraded 3 things:
Improved audio quality - probably similar to Busman mod - and they also fixed the underspecified capacitors for phantom power. I wonder if they chose the same op-amp as Busman? That would be funny.
SDXC support, along with the <<readily available>> SD and SDHC cards probably means we won't see an updated supported media list, which could also translate to better reliability and less problems with lost recordings, especially coupled with the next upgrade
Improved battery life - which, among other things, translates to a new power supply design that could fix many of the problems they had with overheating units, freezing units, power issues leading to lost recordings and 0 KB files.
Another three upgrades are firmware related and could have easily been added to the mkI:
Dual level recording - the only real software upgrade
Reset file take no. (really? This shouldn't even be mentioned as an improvement, it was supposed to be there in the first place)
Playback of multitrack files (the same, it should have been there from the start, especially considering that mkI wouldn't even play back its own recordings once copied to the PC and back to SD again)
Another one is a faux point:
lithium battery support - lithium AA's work everywhere - and they cost 20$ for an 8-pack. Who wants to pay 20$ for a one-time usage?!
And finally there's the 'internal clock oscillator' improvement which nobody even complained about afaik.
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Of course this comes out 2 weeks after I bought my 680. :facepalm:
If bought new, are you still in the return period?
You guys have already spotted the improvements in the 680. It just seems to me that Tascam has extended the life of their existing tooling by upgrading some of the features in the 680 sort of like they did with the DR60d. Like another poster, I had been thinking Tascam might come out with an entirely new 6-8 track unit on the lines of the 70d, but now that I see what they've done with the 680mkii, I think they are trying to get more sales with an upgrade to the existing product before bringing an entirely new unit to market.
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My question is, will there be a firmware update for the DR-680? Seems like most of the upgrades on the mkII are firmware related, but then again, what do I know? ???
The entire power topology had to have been redesigned to increase run times by 150% - and probably many of the components on the logic board where replaced with newer more efficient versions (or consolidated down)
So it is highly improbable a firmware update will come out to add mk ii like features to the older 680
Look at the Marantz updates - they did MINIMAL changes (most if not all apparently software related) to the 661 mk ii from the orig 661 and never came out with an update firmware for the 661 to add these features
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The entire power topology had to have been redesigned to increase run times by 150% - and probably many of the components on the logic board where replaced with newer more efficient versions (or consolidated down)
An firmware update increased the run time of the DR2d by something like 4 times!
Who knows at this point if the topology or PCB was changed or not, components have been changed at least, and I agree that firmware updates for the original mki version are unlikely. Tascam will be focusing on moving the new units.
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The new model is great news. Everyone everywhere please buy the new unit.
Then sell me your old minty unit in the yard sale.
Seriously, the upgrades were a big so what for me.
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Seriously, the upgrades were a big so what for me.
I agree that the added features are fairly lackluster, but the biggest upgrade might have been as simple as fixing defects. That's probably not something they're willing to fess up to in their marketing.
If they did indeed fix the many known, serious defects of the original 680, then I would never settle for an older v1 version.
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It would be pretty cool if someone would add post-processing (multi-band compression, eq, etc) on playback.
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Seriously, the upgrades were a big so what for me.
I agree that the added features are fairly lackluster, but the biggest upgrade might have been as simple as fixing defects. That's probably not something they're willing to fess up to in their marketing.
If they did indeed fix the many known, serious defects of the original 680, then I would never settle for an older v1 version.
This is what's on my mind. I'd like more than 4 channels and I'm not super picky regarding preamps but I don't want to lose recordings because my recorder overheated and crashed in my bag or whatever. I won't be an early adopter but I'm very interested in field reports once it's been out a while. The reality is I should probably just consolidate to an HS-P82 and be done with it, but haven't gotten there yet. I can't imagine this mkII would really compete with that, but you never know.
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This is what's on my mind. I'd like more than 4 channels and I'm not super picky regarding preamps but I don't want to lose recordings because my recorder overheated and crashed in my bag or whatever. I won't be an early adopter but I'm very interested in field reports once it's been out a while. The reality is I should probably just consolidate to an HS-P82 and be done with it, but haven't gotten there yet. I can't imagine this mkII would really compete with that, but you never know.
Technically speaking, you're only an early adopter if it's a new product - the mkII is an updated version of an already established product (4 years old).
The HS-P82 is the flagship, Of course the 680 is a step-down, but it's the next best thing from TASCAM, and in my opinion, the cheapest high-quality multichannel recorder available at the moment. Noise floor is comparable to Sound Devices 7xx series, much better than Edirol R44, which is more expensive and only has 4 channels.
And unfortunately you're never 100% safe with a single recorder - I've seen cases of SD788's failing, and even an extreme case of an AATON CANTAR III (14000$) with lost recordings and hard drive sent for recovery in France, to recover feature film takes - every sound mixer's nightmare.
I'd keep the R44 and get the 680mkII as well. They can be synced via digital and you're sure to never lose a show, worst case scenario - one of the decks fails, and you lose some of the mics - either to the r44 or the 680. Or you could use the USBpre2 to get a full 8 channel analogue recording and try to sync the r44 to the 680... looking at your equipment list, there are lots of possible combinations to have fun with :)
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^^^
DR-680+R44 alone = 10 clock sync'd channels, you just need to align the files from both machines. I've done that a number of times. Using the USBpre2 as well you get all 12 clock sync'd channels.
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^^^
DR-680+R44 alone = 10 clock sync'd channels, you just need to align the files from both machines. I've done that a number of times. Using the USBpre2 as well you get all 12 clock sync'd channels.
What if you had only two DR680's? Would it be possible to record 12 sync'd channels via analog-in only? They advertise that the DR680 can serve as a master transport, but there's no mention of digital clock sync...
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...a little red paint and I'll have a MKII. ;)
I actually think this is a good thing. It really is a good multi-track recorder. If they fixed the minor gripes people had with it it's even better.
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^^^
DR-680+R44 alone = 10 clock sync'd channels, you just need to align the files from both machines. I've done that a number of times. Using the USBpre2 as well you get all 12 clock sync'd channels.
What if you had only two DR680's? Would it be possible to record 12 sync'd channels via analog-in only? They advertise that the DR680 can serve as a master transport, but there's no mention of digital clock sync...
That's the way it's supposed to work, master/slave sync is via the SPDIF ports. Haven't tired it. If so, it should do 14 with using an external ADC into the master machine.
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I actually find the red on the new ones quite a bit more appealing than the silver :)
A 680 + my R-44 and USB Pre2 would give me a lot of options, channel capacity, and redundancy when I feel I need it. Likely what I'll do depending on the price of the mkII once it comes out and initial reports are positive.
There are two factors I'm considering which still weigh in HS-P82's favor:
- I want to play with timecode between audio and video
- I'm growing weary of the cabling and powering of multiple devices. I like running at least one camera most of the time so the amount of gear is getting more difficult to manage.
Now if they come out with a updated hs-p82 with the red handles, well.... the decision is basically made for me :P
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It's plainly obvious that Tascam has made some upgrades to the mkii to get more life of out their existing production line tooling which means more net profit.
Timecode is something that also interests me, but I'm afraid I don't know enough about it. My GH3 has internal timecode, but there's no input to eternally sync it.
Because of the upgrades to what kind of cards the mkii will accept, I'm wondering if you can use a mkii with an original 680 or if you can only link two of the same model?
edit: now seeing the mkii at B&H for $599 preorder.
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Apparently didn't get a notification about new posts...
C. With some microphones, the DR680 is reported to have unacceptable amounts of preamp noise (+12dB) when the phantom power is set to ON, there is even a mod indicating how to get rid of this problem - apparently the filtering capacitor of 0.5 uf (2x1uF in series) used to filter the phantom power just at the node before the feeding resistors of 2x3.3 kOhms is much too low to eliminate the audio noise
generated by the internal 48 V converter, so replacing those capacitors is needed for a 12 dB preamp noise improvement with certain mics.
Can't say I've ever noticed a problem. When first turning phantom on there's sometimes some noise for a few seconds, but I've never had any sustained noise past that.
F. Badly designed speaker - hissy, impossible to turn off except with a jack plugged in the auto-sensing Headphone-out jack
You can just turn the volume down.
G. Badly designed headphone amp - very noisy, digital volume control unresponsive, lags behind and sounds like a zipper file, without zero-crossing detection
It could be better, but it works.
H. TRS Jacks 5/6 can cause popping on recording (one unit here had such a problem, was fixed with a contact cleaner called Progold)
I did have an issue once where there was a problem with contact and phantom and/or the mic signal was getting interrupted. Since then, I've made sure to move the plugs in and out and around a bit before recording to prevent any such issues.
K. Support with TASCAM is pretty bad, they apparently want to have 3 tries at repairing the unit before replacing, which could mean up to two months for somebody in the states, but a lot more for someone like me, living in Eastern Europe.
Some people have certainly had issues, but (fingers crossed), I've been using mine a few months short of 2 years, and haven't run into a problem yet. AC power, a few different brands of SD cards (none listed in the approved media list AFAIK), combination of line-in and mic (both dynamic and electret). The biggest issue I've had is running digital in on 7&8 can be confusing. It works fine, it's just easy to do something wrong.
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"...it's just easy to do something wrong." Agree. Otherwise, there wouldn't have been so many prior posts asking the question "how do I ?" of one kind and another. Although Tascam has apparently addressed some of the physical issues with the unit, it is not the easiest recorder to use, IMO. That's why mine sits unused so much. You can't expect to just start recording without studying the thing.
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"...it's just easy to do something wrong." Agree. Otherwise, there wouldn't have been so many prior posts asking the question "how do I ?" of one kind and another. Although Tascam has apparently addressed some of the physical issues with the unit, it is not the easiest recorder to use, IMO. That's why mine sits unused so much. You can't expect to just start recording without studying the thing.
Well, I agree for using the digital input. But I thought it was pretty straightforward using the analog inputs.
It will be interesting to see how the preamp setup compares with the MKII. As I previously noted, if you're recording relatively low level stuff like acoustic instruments, especially with dynamic mics, setting the gain switch to LOW and turning up the gain results in significantly more noise than setting the switch to HIGH and turning down the gain. Not a big deal once you are aware of the issue, but it doesn't seem like the best design.
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Somebody asked about op-amps on the mkII.
The mkII uses a better sounding op-amp compared to the mkI. (BA2115 instead of BA4580)
Only the DR-70D has the NE5532 op-amp.
I thought I mentioned this before, but the Headphone knob is improved on the mkII as well - better taper and finer control.
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Somebody asked about op-amps on the mkII.
The mkII uses a better sounding op-amp compared to the mkI. (BA2115 instead of BA4580)
Only the DR-70D has the NE5532 op-amp.
I thought I mentioned this before, but the Headphone knob is improved on the mkII as well - better taper and finer control.
Thanks for sharing the info tomuo!
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Somebody asked about op-amps on the mkII.
The mkII uses a better sounding op-amp compared to the mkI. (BA2115 instead of BA4580)
Only the DR-70D has the NE5532 op-amp.
I thought I mentioned this before, but the Headphone knob is improved on the mkII as well - better taper and finer control.
Tom - Has Tascam addressed the chronic overheating problems with the DR-680 and if so, how? Also, as the power input protection been enhanced? The first two 680s I owned were immediately fried or had firmware corruption due to improper dc input power protection.
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Hello everybody, long time reader.
I've read whole 6 threads about DR680. I'm also owner MK I for a year now. I did a capicator mod myself and i must say i can see a difference.
Also about battery power i build a power supply myself. Can say that its checked for more than a 6 months of use and its very good and stable. No problem with overheating and its very efficent.
I used LI-PO battery 11,1 V 25C 10000 mAh (like this one: http://www.ebay.com/itm/10000mAh-11-1V-ZOP-25C-Li-poly-Lipo-Battery-For-Car-Boat-RC-Airplane-Plane-/271587643995) connected to power stabilizer/converter (http://botland.com.pl/przetwornice-step-up-step-down/2127-przetwornica-regulowana-step-upstep-down-s18v20alv-4-12v-2a.html). This converter is essetnial: you can put any DC voltage from range 2,9 V ti 32 V and you can set (useing multimeter) a desire output voltage in a range between 4V to 12V, and its also giveing max 2A power on the end. I know that its 500mA less that's on the AC power adapter, buts by my test its a bit more thats realy needed - checekd on working unit, all 3 phantoms ON and recording.
This battery give me about 7-8h of recording at least. Never used all. Also there is one more thing thats good with that stabilizer. If the battery is fresh after charge it will give you a bit more (still less than 12V, but in case of other kind of battery - perhaps higher voltage - it will drop the voltage down to 12V). With time when the battery discharge the voltage will drop, so the converter will upscale the voltage to 12V - cost of capicty of the battery. If the converter will not able to scale the voltage more, and the output voltage will drop more than 1V it will cut off the circut - so the unit will switch to internal battery.
Edit: Also the good idea is to add LiPo Buzzer, not to destroy battery cell with overdischarging.
Hope that helps.
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Where does everyone set there mix levels for independent channels and stereo mix too??
Todd
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Preorder 599.
http://www.bhphotovideo.com/bnh/controller/home?O=&sku=1116651&Q=&is=REG&A=details
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Hello everybody, long time reader.
I've read whole 6 threads about DR680. I'm also owner MK I for a year now. I did a capicator mod myself and i must say i can see a difference.
Hmm, are you taking orders?
;D
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What capacitor did you swap?
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Capicators for P48 mod. Ill search the gerslutz for revelant topic with instruction. Its realy simple task. Takes about 30 min if you got just basic skilla wit soldering.
Ok, so the oryginal post of Bert from Audiomaster: http://www.audiomastersforum.net/amforum/index.php/topic,8654.45.html
Also the schema: https://www.gearslutz.com/board/remote-possibilities-acoustic-music-location-recording/889971-tascam-dr680-opinions-alternatives-2.html
And how its look like to do (photos by @pwhodges)
(http://cassland.org/images/AM/DR680mod1.jpg)
(http://cassland.org/images/AM/DR680mod2.jpg)
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Awesome. Thanks!
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So if I understand correctly...the capacitor mod is to reduce noise when using phantom power? Can somebody indicate when the noise occurs? Only with certain mics? True condensers? Electrets? Dynamics?
I know - at least with my unit - that if I'm using a dynamic mic, there's no perceptible difference in noise if phantom is on or off. I don't have any true condensers, and I haven't (yet) done any comparisons with the electrets I do have (which of course would require another preamp for comparison).
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Anybody use metal film caps instead of electrolytic caps?
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So if I understand correctly...the capacitor mod is to reduce noise when using phantom power? Can somebody indicate when the noise occurs? Only with certain mics? True condensers? Electrets? Dynamics?
The extra noise occurs only with *some* condenser mics. I saved the article describing which (I thing it was just older mic models? Or mics that take more current? can't recall. Will edit the post if and when I find the link)
EDIT: Here is the original post: http://www.audiomastersforum.net/amforum/index.php/topic,8551.0.html
Dynamics don't care much about P48 if they're properly wired so it wouldn't make a dfference. You can fry electrets if you apply P48 without going through a step-down PIP adapter. But you can use the internal preamps :) And if you would, then the P48 problem would likely turn up.
Here's a nice tech review of the noise floor of the original "stock" DR680:
http://www.lindos.co.uk/index.php?page=test_and_measurement&subpage=SOURCE=Results&subsubpage=/FlexiData/Results/files/filename/399/res.html
By the way, does anybody still have the service manual and schematic? Would love a drobox link if possible.. There is something uploaded in the "manuals" sections but due to size constaints it is cut into small pieces and it seems to be missing some parts.
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Where does everyone set there mix levels for independent channels and stereo mix too??
Todd
I never record the stereo mix. Chances that I get the mix perfect at the show is close to none so I just mix at home.
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I typically leave the stereo mix at 100% and adjust the individual channel mix levels to where ever they need to be.
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Also about battery power i build a power supply myself. Can say that its checked for more than a 6 months of use and its very good and stable. No problem with overheating and its very efficent.
Hello :)
Did you put the module inside the 680? Did it fit OK? Did you do any other modifications to the existing power supply?
Thank you.
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So if I understand correctly...the capacitor mod is to reduce noise when using phantom power? Can somebody indicate when the noise occurs? Only with certain mics? True condensers? Electrets? Dynamics?
The extra noise occurs only with *some* condenser mics. I saved the article describing which (I thing it was just older mic models? Or mics that take more current? can't recall. Will edit the post if and when I find the link)
EDIT: Here is the original post: http://www.audiomastersforum.net/amforum/index.php/topic,8551.0.html
Dynamics don't care much about P48 if they're properly wired so it wouldn't make a dfference. You can fry electrets if you apply P48 without going through a step-down PIP adapter. But you can use the internal preamps :) And if you would, then the P48 problem would likely turn up.
mkII has larger caps for 48V smoothing. (4.7uF instead of 1uF)
By the way, does anybody still have the service manual and schematic? Would love a drobox link if possible.. There is something uploaded in the "manuals" sections but due to size constaints it is cut into small pieces and it seems to be missing some parts.
mkI Service Manual 3493KB, Schematics 808KB. PM me your email address.
Tom.
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The extra noise occurs only with *some* condenser mics. I saved the article describing which (I thing it was just older mic models? Or mics that take more current? can't recall. Will edit the post if and when I find the link)
EDIT: Here is the original post: http://www.audiomastersforum.net/amforum/index.php/topic,8551.0.html
Dynamics don't care much about P48 if they're properly wired so it wouldn't make a dfference. You can fry electrets if you apply P48 without going through a step-down PIP adapter. But you can use the internal preamps :) And if you would, then the P48 problem would likely turn up.
Here's a nice tech review of the noise floor of the original "stock" DR680:
http://www.lindos.co.uk/index.php?page=test_and_measurement&subpage=SOURCE=Results&subsubpage=/FlexiData/Results/files/filename/399/res.html
By the way, does anybody still have the service manual and schematic? Would love a drobox link if possible.. There is something uploaded in the "manuals" sections but due to size constaints it is cut into small pieces and it seems to be missing some parts.
Thanks for the link. Seems to be a quirk with the 680 combined with quirks in certain mics.
BTW, by "electret" I meant mics that are marketed as standard condensers but are permanently charged and require phantom. Of course some of those can be run via PIP if the phantom adapter is separate and can be removed.
Also, while I haven't experienced the noise issue with my 680, I *did* notice a quirk. I got a pair of Rode M5s, and I was having issues with hum when the 680 was plugged in to AC. Touch the grille on the M5s and there would be tons of hum, unless you also touched a grounding point on the 680. It turned out the grille wasn't grounded. But that was *not* an issue with any of my other interfaces. Clearly there was some difference with how the 680 handled grounding. Regardless, I sent my mics back and got new ones that have the grilles properly grounded, and hum is no longer an issue.
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So if I understand correctly...the capacitor mod is to reduce noise when using phantom power? Can somebody indicate when the noise occurs? Only with certain mics? True condensers? Electrets? Dynamics?
The extra noise occurs only with *some* condenser mics. I saved the article describing which (I thing it was just older mic models? Or mics that take more current? can't recall. Will edit the post if and when I find the link)
EDIT: Here is the original post: http://www.audiomastersforum.net/amforum/index.php/topic,8551.0.html
Dynamics don't care much about P48 if they're properly wired so it wouldn't make a dfference. You can fry electrets if you apply P48 without going through a step-down PIP adapter. But you can use the internal preamps :) And if you would, then the P48 problem would likely turn up.
mkII has larger caps for 48V smoothing. (4.7uF instead of 1uF)
By the way, does anybody still have the service manual and schematic? Would love a drobox link if possible.. There is something uploaded in the "manuals" sections but due to size constaints it is cut into small pieces and it seems to be missing some parts.
mkI Service Manual 3493KB, Schematics 808KB. PM me your email address.
Tom.
Tom:
That's so cool that you'll provide schematics! PM sent!
I wonder if you guys have figured out why some of the first gen DR-680's overheated and killed the power supply. I've had three of them and mine is out of warranty. But, I'd like to make a circuit change to fix that overheating problem. Other than that, I love mine. I stuck with the DR-680 even though my first two died. It has all the features I need. I haven't found anything i like better for 6-8 channel recording in the field yet.
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Why not buy a MKii that is already factory modded and use the 680 for those instances where more channels are needed via the cascade function?
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Why not buy a MKii that is already factory modded and use the 680 for those instances where more channels are needed via the cascade function?
:) I will probably do that.
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So what are the prospects for repair of broken 680s?
I BO'ed that broken one from ebay last week - should be here before the weekend.
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Also about battery power i build a power supply myself. Can say that its checked for more than a 6 months of use and its very good and stable. No problem with overheating and its very efficent.
Hello :)
Did you put the module inside the 680? Did it fit OK? Did you do any other modifications to the existing power supply?
Thank you.
nope, i didn't put battery inside. A did a battery pack external, connected by plug to the 12v connector on the side. This way I also made another cable plug to powet up my Saffire Focusrite 26 PRO for additional preamps that im sending to Tascam 7-8 track by SPDIF if needed.
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mkII has larger caps for 48V smoothing. (4.7uF instead of 1uF)
That is exacly what i did.
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I did some testing tonight with phantom power noise and difference between hi-lo gain.
Test setup was this:
mic -> Radial ProMS2 "direct 1" -> DR680 channel 1 +P48.
Radial ProMS2 "direct 2" -> DR680 channel 4.
Channel 1+4 = left and right of stereo mix. Direct 2 on the ProMS2 is transformer isolated from the direct 1 that carries the phantom power.
The idea was that if there is a difference between a channel with and without phantom power in terms of noise this test ought to show it. If I am making a mistake in my reasoning, feel free to let me know.
Mics tested: Shure KSM141 cardioid and omni, Rode NT5 omni capsule, Rode NT4 MJE-384K capsule, AT4081, Advanced Audio CM1084 (cardioid) in Bright and Vintage settings.
Result: I could not prove that there is any noise difference between the phantom and non-phantom channels. I did see on most mics a distinct hump around 100Hz on all mics except the AT4081. Perhaps due to environmental noise that the figure 8 ribbon blocks out.
Perhaps when one channel has phantom power all channels suffer the same amount of noise? Or none of the tested mics suffer from this problem, as I understand it varies from one mic to the next.
A second thing I tested was the suggestion that switching from low +15 to high -8 (for example) would make a difference in noise. Here I can say that yes, there is a noticeable difference. On most mics the dfference started around 1000Hz and widened to around 2~4 dB at 10KHz. So in a quiet section of the recording (me not saying anything in my quiet living room) the low +15 channel might be at -114 dB, the high -8 channel would be at -116dB at 10KHz.
The biggest difference was with the CM1084 in vintage setting, Low +20 and High -3. There the difference at 10Khz widened out to around 8dB.
Regards, Christine
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Thanks for that Christine. I've never done a measurement of the LOW/HIGH noise difference, but it was very audible, to the point that I'm slightly embarrassed at the (thankfully one) recording I made with the switches set to LOW. I wonder if that will still be an issue with the mkII.
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I did some testing tonight with phantom power noise and difference between hi-lo gain.
Test setup was this:
mic -> Radial ProMS2 "direct 1" -> DR680 channel 1 +P48.
Radial ProMS2 "direct 2" -> DR680 channel 4.
Channel 1+4 = left and right of stereo mix. Direct 2 on the ProMS2 is transformer isolated from the direct 1 that carries the phantom power.
The idea was that if there is a difference between a channel with and without phantom power in terms of noise this test ought to show it. If I am making a mistake in my reasoning, feel free to let me know.
If I understand correctly, the small capacitors in the stock unit lead to a raise in the self-noise level of the microphone. So your method would not work, given the fact that the problem is not a ground loop hum, but one of mic self-noise - the signal from the mic coming out of your splitter should be identical.
A proper test would involve an external phantom power adapter, and using only one input channel of the DR680 with a condenser mic with P48 turned on VS P48 OFF with the mic powered by the external phantom adapter box (such as the millenium PP2B)
Users on a french forum tested the DR680 when it came out and found that the noise floor is slightly different on each channel on both mic and line.
The HI-GAIN- LOW-GAIN noise difference is not a problem, it's a feature actually - works as a PAD - helps you NOT to overload the inputs with high-level SPL. I'm pretty sure it's the same in the MKii - so there was nothing to fix there!
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The HI-GAIN- LOW-GAIN noise difference is not a problem, it's a feature actually - works as a PAD - helps you NOT to overload the inputs with high-level SPL. I'm pretty sure it's the same in the MKii - so there was nothing to fix there!
But the gain control is in the analog domain, isn't it? Why would a pad even be necessary most of the time?
And wouldn't it make more sense if it was actually called a pad, like on mics and other interfaces?
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Input gain is via a digitally controlled analog PGA, AFAIK. I'm pretty sure the switch changes the gross range of gain-adjustment of the PGA, and is a function built into it rather than being implemented prior to it. If implemented in it, it's shifting the gain range up and down and is not a pad, which would instead be attenuating the signal prior to it reaching the PGA.
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Input gain is via a digitally controlled analog PGA, AFAIK. I'm pretty sure the switch changes the gross range of gain-adjustment of the PGA, and is a function built into it rather than being implemented prior to it. If implemented in it, it's shifting the gain range up and down and is not a pad, which would instead be attenuating the signal prior to it reaching the PGA.
That would make more sense, but at the same time, if that was the case, why would the noise level differ at the same overall gain depending on where the switch was set?
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It may do it by changing the gain of a separate gain stage
within the PGA but before the more granular trim, or in a completely separate stage prior to the PGA. For best noise performance, most of the gain should be made in the first stage. The gain of the second amplifies the noise of the first.
Wait, I'm forgetting about the HDA discrete stuff, that would be the first stage. It probably has two gain levels, line/mic switched, and after that, the fine trim adjustment is made via the digitally controlled PGA. So if the PGA is having to make more of the gain (input LOW, trim cranked) input noise increases. With input HIGH, the quieter HDA stage is doing a greater share of the work, so input noise is lower.
Improved first-stage HDA performance over the first gen machine would translate as better noise performance either way, but you still want to keep the gain of the first stage HIGH and the second stage lower, unless you'd clip the first stage, in which you'd set it to LOW, in which case the signal will be hot enough that input noise isn't an issue.
Start HIGH, then go LOW if necessary to avoid overloading the input.
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A proper test would involve an external phantom power adapter, and using only one input channel of the DR680 with a condenser mic with P48 turned on VS P48 OFF with the mic powered by the external phantom adapter box (such as the millenium PP2B)
In other words, have the mic go into a DR680 channel with phantom turned on, but that phantom power isn't actually powering the mic. That would require the P48 channel on the DR680 to be attached to the transformer isolated side of the mic splitter to avoid the risk of the mic receiving double P48.
The non-P48 channel of the DR680 can be hooked to the non-isolated side of the mic splitter.
Or am I understanding you incorrectly?
Regards, Christine
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A proper test would involve an external phantom power adapter, and using only one input channel of the DR680 with a condenser mic with P48 turned on VS P48 OFF with the mic powered by the external phantom adapter box (such as the millenium PP2B)
In other words, have the mic go into a DR680 channel with phantom turned on, but that phantom power isn't actually powering the mic. That would require the P48 channel on the DR680 to be attached to the transformer isolated side of the mic splitter to avoid the risk of the mic receiving double P48.
The non-P48 channel of the DR680 can be hooked to the non-isolated side of the mic splitter.
Or am I understanding you incorrectly?
Regards, Christine
If I'm understanding correctly, you wouldn't be able to use a splitter, you'd either have to do two separate tests with the same mic, or use two mics of the same model, one powered by an external source and the other powered by phantom from the 680.
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If I'm understanding correctly, you'd either have to do two separate tests with the same mic, or use two mics of the same model, one powered by an external source and the other powered by phantom from the 680.
This is what I am envisioning one would want to do/acccomplish
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In other words, have the mic go into a DR680 channel with phantom turned on, but that phantom power isn't actually powering the mic. That would require the P48 channel on the DR680 to be attached to the transformer isolated side of the mic splitter to avoid the risk of the mic receiving double P48.
The non-P48 channel of the DR680 can be hooked to the non-isolated side of the mic splitter.
Sorry if I was a bit vague last night.
What I meant was, I don't think the mic splitter would be useful in this case. OFC it's a great tool to compare mic pres, but in this instance, a simpler approach is needed, because we need to test the influence of the phantom-48 on the mic's own noise floor. So we're not actually testing for the differences in mic pres, but the quality of the electrical current provided to the mic.
Hence, using a mic splitter would yield almost identical results on different channels of the DR680, since the signal it's splitting is already "polluted" by the +48V the DR680 is feeding it.
So a proper test would include just a condenser mic, the DR680, and an external phantom power supply.
You would make two consecutive recordings:
First, mic -> DR680 P48 ON
Second, mic-> phantom power box -> DR680 P48 OFF.
There is no case I can think of, even including the mic splitter, which could accomodate a single recording of two different channels which would show the difference in this case, but then again I'm not familiar with mic splitters.
As far as I know there are many sources which influence the noise in a recording:
- mic self-noise (except dynamic mics, but it's a different topic),
- the p48 supply,
- the mic pre EIN,
- the a/d converter "resolution" or EIN
- electromagnetic interference (this is why expensve recorders have all-metal chassis - it creates a faraday cage
- the quality of the power feeding the recorder (esp. in case of AC power - the AC outlet never gives a perfect sine wave, it may be distorted, there may be many harmonics and parasitic noise, some people even hear radio...)
Here are some links:
http://www.beat-net.blogspot.tw/2014/08/dr-680-phantom-power-noise.html
http://outrecording.com/tascam-dr-680-noise-test/
This is only ever a problem with low-SPL recordings. It would likely never influence anything going over 70 dBFS (normal speech), and in NO CASE would it matter for taping amped shows.
It might make a difference in recording quiet acoustic performances in quiet places though..
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http://www.beat-net.blogspot.tw/2014/08/dr-680-phantom-power-noise.html
http://outrecording.com/tascam-dr-680-noise-test/
Now...both of those links seem to suggest the issue does *not* have to do with mic self-noise. For instance this quote from the second link:
"Testing phantom power noise was as simple as turning up the gain and flicking it on and off. Of course with no mic plugged in. You can hear a stark difference when turned on and off at a HIGH GAIN."
So now I'm confused.
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Now...both of those links seem to suggest the issue does *not* have to do with mic self-noise. For instance this quote from the second link:
"Testing phantom power noise was as simple as turning up the gain and flicking it on and off. Of course with no mic plugged in. You can hear a stark difference when turned on and off at a HIGH GAIN."
So now I'm confused.
Technically speaking, it is distinct from the mic self-noise, but it influences it. And there's no way to tell it apart unless you have an external +48V supply. I was using the term as an example to why using a splitter wouldn't work in this particular case (testing mic ->splitter -> dr680 two channels -one with +48 on, the other with +48off)
However testing the noise level of an input with nothing attached to it doesn't work - the noise will be very high as it's not terminated, not balanced and picks up em interference. The author of the second link does say that his test "is hardly scientific or conclusive" though :)
Turning phantom power on with nothing attached to the input should produce some more noise on any machine as long as there is no termination. An impedance termination is needed, and for +48V, a resistance as well. The noise, even if you hear it with no mic attached, should balance itself out when the mic is connected (all the blah-blah of the balanced connection, 90 degree delayed phase of the signal on one of the wires, noise being the same on both wires thus being phased out etc.)
This is what the author of the first link did: "On one leg I (nr. of the XLR input) connected a resistance to ground to simulate mike consumption. On the other leg I connected a capacitor to ground. This forms a RC filter to keep out high frequency signals."
P48 aside, some people recommend using a paper clip to short pins 2 and 3 of the XLR input to test the EIN of the mic pre:
http://www.daking.com/?cat=15
PS. I'm not an an engineer, I learned this all by myself, so if I'm wrong I'd love to be corrected :)
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OK, new test.
This is the setup:
KSM141 -> Radial ProMS2
Radial ProMS2: Direct 1 output --> RME Quadmic II
Radial ProMS2: Isolated output --> DR680 (gain: L +15)
In other words: The KSM141 is connected to the input of the mic splitter. The output that does pass P48 is connected to the RME Quadmic II (which I assume to be the most reliable phantom power I own). The isolated out of the mic splitter (which does not pass P48) goes into the DR680. Phantom power on the DR680 is turned off. So the Quadmic is providing the phantom power, the DR680 gets the signal. Theoretically totally separate from any P48.
I took the mono channel of this test and the mono channel of the same test except the DR680 providing phantom power and put them through Adobe Audition CS6. In the attached mp3 the left hand channel is DR680 with phantom power, the right hand channel is DR680 without phantom power (phantom provided by RME Quadmic II). The attached jpg shows the frequency analysis; green is left (DR680 +P48), blue is right (DR680 -P48).
The recording is of my reasonably quiet living room.
The only conclusion I can draw from this specific test is that I cannot prove that anything is wrong with the DR680 (Mk I)'s phantom power. Which doesn't mean there isn't a problem, just that I haven't yet been able to replicate such a result.
Suggestions welcome.
Regards, Christine
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You are already well past my level of understanding, but there was a post on GS that I think may be relevant to your question.
https://www.gearslutz.com/board/9675848-post29.html
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This is the setup:
KSM141 -> Radial ProMS2
Radial ProMS2: Direct 1 output --> RME Quadmic II
Radial ProMS2: Isolated output --> DR680 (gain: L +15)
In other words: The KSM141 is connected to the input of the mic splitter. The output that does pass P48 is connected to the RME Quadmic II (which I assume to be the most reliable phantom power I own). The isolated out of the mic splitter (which does not pass P48) goes into the DR680. Phantom power on the DR680 is turned off. So the Quadmic is providing the phantom power, the DR680 gets the signal. Theoretically totally separate from any P48.
Based on the mic's and rated Self-noise, your mics have the following self-noise floor in dBu.
KSM141 Self noise -114 dBA
RODE NT5 self noise -115 dBA. Modified capsule says improved SNR but no specs available...
RODE NT4 self noise -116 dBA
AT4081 -108 dBa
Advanced Audio CM1084 - can't tell (incmplete specs- sensitivity not specified)
TASCAM DR680's mic pre EIN is -128 dB A-weighted, which is a good 12 dB lower than the most silent mic of the bunch, the NT4. So in theory it shouldn't influence any of the mics.
Looking at the noisefloor, the Shure rates even better than specced at -115.5 dB.
It seems like your unit is free of this problem!
Did you buy it new? I wonder what year it was made in, maybe they fixed this silently. It would have been a really easy factory fix - just replace 3 capacitors on the production line :)
I will soon recieve a DR680 MKi and do the test with an NTG3 as well as an NT2A, with and without external phantom box, can't wait to see the results.
By the way, Christine, didn't you post a comment back in August on outrecording.com's article on dr680 phantom power noise article? Looks like your signature :P
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Well, I bought my DR680 (mk I) around March 2014 on eBay from a shop in the UK. I replaced the UK cord for the power supply with one with a European plug (I am in the Netherlands). My impression was that I received a new unit, although I suppose it might have been modified. As far as I can tell from the outside it is entirely standard. Serial number is 0350xxx. But nice to hear that mine doesn't appear to have that problem :)
Yes, I seem to remember something about posting a comment on the DR680 phantom power noise article.
By the way, my impression is that the senstivity of the CM1084 in "Bright" mode is not very different from the KSM141 or the NT5. It's in the same ballpark at least. In "Vintage" mode it is needs 5 dB more gain. The MJE-384K capsules to the NT5 take about 1dB more gain (that is, if I run the standard NT5's at L+6 then the NT5's with MJE-384K capsules need L+7).
Regards, Christine
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Can you use a Sound Devices 702 or 744 to use channels 7 & 8 on the 680 via a S/PDIF? I'm sort of thinking I want one of the SD units for my main two mics, but would be content to use my 680 for spot mics if these units will work together like this.
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Yes. The clock on the 680 will slave to that of the SD across the SPDIF link. You'll need to start the transports separately, and line up the resulting files recorded on the two different machines afterwards, but they will share clock-sync. You can either choose to record the stereo SPDIF input on the 680 or not, it's not required that you do so to sync the machines. If you do, you won't have to do any file aligning between that SPDIF stereo channel and the others recorded on the 680.
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Thank you.
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Hi all, I'm running the new 680 for the first time tonight and have a couple questions. Forgive me if these have already been covered, I haven't had a chance to dig in to the other 680 threads yet.
What's the process to gang/un-gang channel's? I've read the manual but it's not explained very well. Is it really as simple as pressing the the 2 channel buttons on either end of the group you want to gang? Can this be done at any time or only while monitoring/recording? I assume you reverse the sequence to un-gang groups?
I plan to run a 5 channel mix, 2 mics onstage, a stereo SBD feed and a 3rd mic towards the back of the room. Is it possible to create stereo files for the 2 stage mics (1/2) and 2 SBD channels (5/6) then record the 3rd mic (3 or 4) as mono? I'd like mix everything to stereo on the fly but need to be able to split the 3rd/mono mic between left and right.
Is there a way to add delay to any of the channels? The 3rd mic will be about 30-35' away from the stacks. If I can't add delay I'll need to remove it from the stereo mix and add it back in post.
Apologize for all the questions. ??? ??? ??? I'm a little stressed about running tonight, I'll practically be un-boxing 680 at the venue before the show starts. :facepalm:
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What's the process to gang/un-gang channel's? I've read the manual but it's not explained very well. Is it really as simple as pressing the the 2 channel buttons on either end of the group you want to gang? Can this be done at any time or only while monitoring/recording? I assume you reverse the sequence to un-gang groups?
Press and hold the button for one channel, press the button for the other channel, let go. Order doesn't matter. Can be done any time.
I plan to run a 5 channel mix, 2 mics onstage, a stereo SBD feed and a 3rd mic towards the back of the room. Is it possible to create stereo files for the 2 stage mics (1/2) and 2 SBD channels (5/6) then record the 3rd mic (3 or 4) as mono? I'd like mix everything to stereo on the fly but need to be able to split the 3rd/mono mic between left and right.
Page 24:
With this unit, a group of audio files that are recorded at the same time are collectively called a “take.”
Depending on the file recording mode setting, the files included in the recorded take will be as follows.
Example 1: 6 tracks and a stereo track recorded
• MONO format set: 6 mono WAV files and a stereo WAV file
• STEREO format set: 3 stereo WAV files of inputs pairs 1-2, 3-4 and 5-6 and a stereo WAV file of the mix
• 6ch format set: a 6-channel WAV file and a stereo WAV file
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There's no way to have stereo files for 1&2 and 3&4 and a mono file for 5.
Is there a way to add delay to any of the channels? The 3rd mic will be about 30-35' away from the stacks. If I can't add delay I'll need to remove it from the stereo mix and add it back in post.
Nope. I'm not sure what you mean. Why can't you just record the 5 tracks as-is and shift around (if necessary) when mixing?
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Is there a way to add delay to any of the channels? The 3rd mic will be about 30-35' away from the stacks. If I can't add delay I'll need to remove it from the stereo mix and add it back in post.
Nope. I'm not sure what you mean. Why can't you just record the 5 tracks as-is and shift around (if necessary) when mixing?
No delay available on the recorder. You'll need to shift/delay as required later. Set it to write 6 mono files to make doing so easier.
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What's the process to gang/un-gang channel's? I've read the manual but it's not explained very well. Is it really as simple as pressing the the 2 channel buttons on either end of the group you want to gang? Can this be done at any time or only while monitoring/recording? I assume you reverse the sequence to un-gang groups?
Press and hold the button for one channel, press the button for the other channel, let go. Order doesn't matter. Can be done any time.
I plan to run a 5 channel mix, 2 mics onstage, a stereo SBD feed and a 3rd mic towards the back of the room. Is it possible to create stereo files for the 2 stage mics (1/2) and 2 SBD channels (5/6) then record the 3rd mic (3 or 4) as mono? I'd like mix everything to stereo on the fly but need to be able to split the 3rd/mono mic between left and right.
Page 24:
With this unit, a group of audio files that are recorded at the same time are collectively called a “take.”
Depending on the file recording mode setting, the files included in the recorded take will be as follows.
Example 1: 6 tracks and a stereo track recorded
• MONO format set: 6 mono WAV files and a stereo WAV file
• STEREO format set: 3 stereo WAV files of inputs pairs 1-2, 3-4 and 5-6 and a stereo WAV file of the mix
• 6ch format set: a 6-channel WAV file and a stereo WAV file
---
There's no way to have stereo files for 1&2 and 3&4 and a mono file for 5.
Is there a way to add delay to any of the channels? The 3rd mic will be about 30-35' away from the stacks. If I can't add delay I'll need to remove it from the stereo mix and add it back in post.
Nope. I'm not sure what you mean. Why can't you just record the 5 tracks as-is and shift around (if necessary) when mixing?
Thanks lukpac!
So it sounds like my only option to mix the 3rd mic into the stereo mix, without recording an unneeded 6th channel, is to use the MONO format with the stage mics and SBD channels panned full left/right and the 3rd mic center panned. Does that sound right? Can you gang channels running in mono mode?
For my delay question, I'd like to get the stereo mix as close as possible at the show. With the 3rd mic being a ways back from the PA there will probably be a noticeable delay in the mix if I can't delay to the other channels while recording. Channels can be individually delayed on the DR-70D so it seems like this would be an option on it's big brother, the 680.
I know my chances of getting my the stereo mix right at the show are slim but I run a lot of matrix's at this venue and if I can get mixing dialed in over the next few shows it'll save me a ton of time in post.
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So it sounds like my only option to mix the 3rd mic into the stereo mix, without recording an unneeded 6th channel, is to use the MONO format with the stage mics and SBD channels panned full left/right and the 3rd mic center panned. Does that sound right? Can you gang channels running in mono mode?
Yes to both questions.
For my delay question, I'd like to get the stereo mix as close as possible at the show. With the 3rd mic being a ways back from the PA there will probably be a noticeable delay in the mix if I can't delay to the other channels while recording. Channels can be individually delayed on the DR-70D so it seems like this would be an option on it's big brother, the 680.
It should be a feature on the 680, along with polarity invert and EQ, but unfortunately is is not. These features are not included in the new 680mkii version either, which is my primary gripe with the new version.
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So it sounds like my only option to mix the 3rd mic into the stereo mix, without recording an unneeded 6th channel, is to use the MONO format with the stage mics and SBD channels panned full left/right and the 3rd mic center panned. Does that sound right? Can you gang channels running in mono mode?
The stereo mix (essentially the 7th and 8th tracks) is independent from 1-6 as far as format goes. That is to say, it doesn't matter how you save 1-6. The format for those will have no effect on the stereo mix. You need to use MONO to prevent recording an unused 6th channel, but that has nothing to do with the stereo mix.
Yes, you can gang channels in any mode.
For my delay question, I'd like to get the stereo mix as close as possible at the show. With the 3rd mic being a ways back from the PA there will probably be a noticeable delay in the mix if I can't delay to the other channels while recording. Channels can be individually delayed on the DR-70D so it seems like this would be an option on it's big brother, the 680.
I know my chances of getting my the stereo mix right at the show are slim but I run a lot of matrix's at this venue and if I can get mixing dialed in over the next few shows it'll save me a ton of time in post.
Yeah, no option for delay on the recorder.
Usually that amount of delay is probably welcome, although I suppose it depends on the venue and personal taste. And I almost always use EQ and compression anyway, so for me the stereo mix wouldn't be anything more than a rough mix to sample until I did the real thing.
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So it sounds like my only option to mix the 3rd mic into the stereo mix, without recording an unneeded 6th channel, is to use the MONO format with the stage mics and SBD channels panned full left/right and the 3rd mic center panned. Does that sound right? Can you gang channels running in mono mode?
Yes to both questions.
For my delay question, I'd like to get the stereo mix as close as possible at the show. With the 3rd mic being a ways back from the PA there will probably be a noticeable delay in the mix if I can't delay to the other channels while recording. Channels can be individually delayed on the DR-70D so it seems like this would be an option on it's big brother, the 680.
It should be a feature on the 680, along with polarity invert and EQ, but unfortunately is is not. These features are not included in the new 680mkii version either, which is my primary gripe with the new version.
I'm running the mkii tonight, wish I knew these options were missing before I bought it. Cue buyers remorse... >:(
This seems like a major oversight on Tascam's part, maybe they can add some of these options via Firmware updates?
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Is there a way to add delay to any of the channels? The 3rd mic will be about 30-35' away from the stacks. If I can't add delay I'll need to remove it from the stereo mix and add it back in post.
Yep, get two of these, and give one to me ;D
http://hallresearch.com/files/manuals/AD-340.pdf
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Tascam updated the original 680 firmware at least once and issued new firmware for the mkii when it was released. Sorta doubt they will add more features. If you're serious about buyer's remorse and still have the option to return it, you might consider this.
680 operation is not as easy as the 60d or 70d, IMO. I was hoping for a mkii version more along the lines of the 70d myself.
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I'm running the mkii tonight, wish I knew these options were missing before I bought it. Cue buyers remorse... >:(
This seems like a major oversight on Tascam's part, maybe they can add some of these options via Firmware updates?
That's my somewhat doubtful hope. Not sure if it's possible, and if it is if they will have enough incentive to do so.
Please complain about this loudly to Tascam! If enough users do so, they may work to implement it if possible. As a new owner of the mkii, Tascam should be more open to your concerns than the rest of us!
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Tascam updated the original 680 firmware at least once and issued new firmware for the mkii when it was released. Sorta doubt they will add more features. If you're serious about buyer's remorse and still have the option to return it, you might consider this.
680 operation is not as easy as the 60d or 70d, IMO. I was hoping for a mkii version more along the lines of the 70d myself.
There were 3 or 4 firmware revisions on the 680 (I never implemented the last, which was mostly an mp3 update as I recall). The early updates offered a few major feature improvements however, which included the channel ganging feature. This all hinges on if there is the DSP processing power and architecture available in the machine to implement those features. If not, Tascam should have added that with the mkii hardware updates. The 70d doesn't feature channel ganging, which is a deal-killer for me (and I need more than 4 channels too). It doesn't appear channel ganging could be added easily to the 70d due to it's analog gain pots. If they were infinitely rotatable digital control pots instead, they could add it while retaining the easy "twist a separate physical knob for each channel" interface.
2manyrocks, If you return it please tell Tascam why you are doing so, not just the retailer, so that they are made aware of this stupid design mistake.
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Is there a way to add delay to any of the channels? The 3rd mic will be about 30-35' away from the stacks. If I can't add delay I'll need to remove it from the stereo mix and add it back in post.
Yep, get two of these, and give one to me ;D
http://hallresearch.com/files/manuals/AD-340.pdf
Cool box. Except he'd need both of them delay the other 4 channels, and preamplification/phantom for two of those as well.
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Thanks for the input gents! The UPS man delivered a little early so I was able to play with settings on lunch. It seems like a fairly straight forward unit to operate, ganging channels and adjusting trim/pan/mix levels is about as simple as it gets.
I'll be making my voice heard to tascam about the mkII not having delay and polarity settings, total BS if you ask me. Would you recommend PM'ing the rep on the forum (tumuo?) or talking directly to customer support? I'd like to bring it to their attention before dropping a bad review bomb on B&H. >:D
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I dunno who to contact at Tascam.
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Tom Duffy (http://taperssection.com/index.php?action=profile;u=53688)
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I'm experiencing about 2 to 2 1/2 hour battery time on the 680 using fresh alkaline batteries with some occasional turning on of the back light. I think the best I've ever done was 2 hours 55 minutes. The unit mercifully shuts down automatically when the batteries die, and keeps the files intact, which is a nice feature. I do have an external battery source, which I like, but it is not always possible to run it inside venues.
I wish the unit would give me warning of a few minutes before shutting down. My experience with operating the unit on one bar (of three) of battery power is that last bar can mean 30-60 minutes battery time.
Has anyone used lithium batteries on this unit to extend the battery run time?
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^ why can't you use an external battery, at indoor venues
???
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For my delay question, I'd like to get the stereo mix as close as possible at the show. With the 3rd mic being a ways back from the PA there will probably be a noticeable delay in the mix if I can't delay to the other channels while recording. Channels can be individually delayed on the DR-70D so it seems like this would be an option on it's big brother, the 680.
If they had the delay it would be an easy to calculate the amount of delay needed with velocity formula, just in case.
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General rule of thumb adjustment of 1 ms per foot is pretty close.
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I'm a stealth taper, so in large degree, stuff has to fit in my crotch. While I have a huge penile member and package, there is a limit to the amount of stuff I can shove down my crotch and still have room for my fun stick.
I realize that crotching and stealthing a DR680 is not the norm. And using it in open taping situations or in the car, I have a nice external battery pack.
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I don't think the 680 was designed to be stuffed in the crotch. Maybe that will come in the mkIII.
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^^ Oh I didn't think of that, makes sense now!
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General rule of thumb adjustment of 1 ms per foot is pretty close.
Gutbucket your not going to adjust for air temperature and barometric pressure? Just messing with you.
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Heh, nope. Though I always chuckle whenever I see one of the good Captain's txt files, which never fail to note the temperature and relative humidity. ;)
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Max thread limit reached. Time for a new one.
$7: http://taperssection.com/index.php?topic=173392.0