Overdubbing on D-160v2

markert

New member
Got my d160 recently. Layed down eight drum tracks the first day out. However, I seem to be having trouble dubbing the rest of the tracks. I have drums on tracks 1-8. I then make track 9 "record ready" and and start recording a guitar, but instead of the guitar, it seems to be putting what's recorded on track 1 onto track 9. Maybe I am patched wrong, but the Fostex manual is not very helpful on this and other topics.

Any suggestions? Any help is appreciated.
 
Markert,

Specifically what mixer are you using.

I encountered the same problem a few months ago and after becoming increasingly frustrated with a lack of a solution and having annoyed anyone who may have had an answer (including Skippy) we eventually found it was a problem with the console in the studio we were in at the time.

Peace..........ChrisO :cool:
 
Yup- I'll bet you a beer that it is a patching problem. It also might possibly be a clock probem, but only if you are either using an external clock source, external converters, or both. If you just got the box I doubt that that's the case- so let's focus on the patching thing.

The D160 as delivered has only 8 A/Ds. So to record on track 9, you connect your input to the D160's input 1, and enable track 9. Sounds like you have that part done. Only question is, what is on the *input* to channel 1? If you've left it patched up from monitoring the drum mix, you may well be simply feeding the output of track 1's playback right back into the #1 input. This is especially likely if you are driving the Fostex inputs with channel direct outs from channels 1-8 on your board.

Think about it: if the outputs from tracks 1-16 on the D-160 are put into line inputs 1-16 at your board (a likely setup), then the direct out on channel 1 will have the playback from tape track 1 on it *by default*, unless you explicitly repatch something somewhere. It gets worse: on some boards, like my Alesis Studio 32, there is no mic/line switch. So whatever is on the line input, and whatever is on the mic input, get *mixed together*. Meaning, of course, that the mic signal loses, since it is probably 40dB or so lower than the line signal. This can certainly drive you _nuts_ until you discover it.

It can be moderately painful to work out a working style with this going on. Even with an inline board like the Studio32 (which has entirely separate tape return/monitor inputs), the mic/line commoning problem is a downright pain in the ass. I want a freakin' switch there!

Anyway: repatch the D160's inputs to the direct outs on the appropriate channel, don't just reuse channel one on the mixer to try and track with *in addition to monitoring* (unless you have an inline-monitor board, which is probably not the case). If you are using direct outs, move the Fostex input 1 to the direct out of (for example) channel 9, and just use channel 1 to monitor what's on track 1, and use channel 9 for your overdub.

The bottom line is that you need to separate your monitoring function (the channels used for existing tracks) and the recording function (the new material intended for unused tracks) to separate channels on the mixer, and that absolutely requires repatching. I personally let the recorder track assignment drive my channel assignment on the board: board channel 9 is _always_ associated with Fostex track 9. Thus, I'm constantly changing the patch for the Fostex input #1 between direct out of channel 1 and 9 at the board. That is, until my AC2496 converter box showes up, giving me 1 A/D per channel and simplifying that part of my life to no end.

Hopefully, that makes some sense... If not, post more details about your rig. However, the only way that the D-series machines will duplicate one track onto another is by external analog feedback, or by using the internal "track dupe" function. Whatever is happening here is external to the box.
 
Thanks Ausrock and Skippy. You guys have been a lot of help, especially since the fostex manual is a little cryptic.

Yeah, Skippy, I think maybe you are right about the board. I was recording remotely on some guy's fostex mixer and we had to monitor through the same channels as the inputs. I probably didn't press the mic/tape input button.

I think I have solved the problem by getting a 16 in/2 out fostex line mixer from ebay. It's rack mounted and will be used for monitoring. I wll probably use my M-audio DMP-2 dedicated mic preamps. Although, musician's friend has a new ST Audio eight preamp rackmount thingy that would make the whole system quite portable.

By the way, I got the d160 to sync quite well with my Digi001/protools system on my mac. Dumped eight tracks at a time with ease.

Thanks again.
 
Glad to hear that you got it to sync up with your DAW easily. I fly 16 at a time up and down into Cubase (W98/Hammerfall), and it works like a champ as well, after an initial period of minor wordclock annoyances.

Those days of simple operation are coming to an end, though: I'll no longer be able to just leave the Fostex's ADAT I/O patched to the Hammerfall all the time. Once the external converter box arrives, I'll need to repatch the lightpipe inputs: either 16 in from the Hammerfall, or inputs 9-16 *only* from the converter box. That's to work around the Fostex digital-input gotcha (which isn't very well documented in the manual, and which you'll eventually need to know about).

When you set the D-series recorders up for digital input, the analog inputs are still live, and are used to "fill in" inputs that are not locked to incoming data on the associated ADAT port. So, let's say you're using an external 8-channel converter like the AC2496, so that you can record 16 tracks simultaneously. You plug your external converter into the input 9-16 ADAT port, and leave the input 1-8 ADAT port empty. Set the fostex up for digital input and ADAT sync. And the unit will use its internal converters anyway for tracks 1-8, since they don't have any incoming lock. Piece of cake, right?

The problem is that the input 1-8 ADAT port has to be dark: you can't be shipping digital 0 in from some other item on that port (like leaving it patched to your DAW), or it'll detect sync and use that digital 0 signal for 1-8, rather than the internal converters. Gotcha!

So, until I come up with a few extra bucks for an ADAT patchbay, I'll be crawling around on the floor occasionally, reconfiguring for tracking or for receiving data from the DAW.

One other gotcha: apparently, not many supposedly "ADAT-compatible" digital patchbays can actually handle the bandwidth of 24 bit recording. If you end up buying an ADAT switcher, proceed with extreme caution: the only one that I'm aware of that actually can consistently handle the bandwidth requirements without data munging and/or dropouts is the Z-sys z8.8a, which I can't afford just now.... Caveat emptor!
 
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