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Gear / Technical Help => Ask The Tapers => Topic started by: drumminj on January 25, 2004, 12:41:56 PM
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So a few weeks ago I taped a show both mic'ed and off the soundboard, to two different JB3s. The sources were SBD->JB3 and at853->ua-5(optical out)->jb3. Trying to matrix these in Wavelab, at first it appeared there was some drift of maybe 20 milliseconds. No biggie, probably not even noticable. It seemed the mic'ed source fell behind. I used wavelab to stretch the mic'ed source, and things seem to be fine, for the most part.
HOWEVER, as I'm listening through the show just making sure there isn't any destructive interference with how i have things lined up, I've noticed that at some places, the mic'ed source jumps ahead a few MS, and quickly goes from being behind about 5-10ms (how I aligned things) it's now ahead by 1. I would expect that any differences between the clock in the UA-5 and the JB3 would be constant, and not vary throughout the course of the show. So the only conclusions I can come to is that 1) Wavelab's stectching function is bad, or 2) there are actually some dropped samples in the mic'ed source.
The second conclusion wouldn't surprise me terribly, but I'm curious if anyone has a different take? Any way to be sure this what happened? At this point I've spent enough time on this (and still have more to go) that I don't want to have to go back to the unstretched source, line it up, and test, but it'd be nice to know if the JB3 was misbehaving.
Sorry for the long ramble. My thoughts aren't very coherent this morning :)
J
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I think it's clock differences, I had the same problem working with DAT sources before. Now I won't touch a matrix unless the word clocks between the converters are sync'ed. The worst nightmare I ever had was syncing two analog cassette sources, constant varying drift, ahving to re-sync every 4 seconds at the worst parts....I just wanted to pull my hair out and quit......
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I guess with analog casette the varying drift makes more sense as it's an issue with the mechanical drives on them, right? With A/D converters I would think it should be constant. Am I wrong on that assumption?
Regardless I'm gonna finish this matrix. I'm too stubborn not to (unfortunately). But it's definitely getting frustrating :(
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the problem is with the accuracy of the clocks. Apogee is known for their rock solid clocks, the reason for this is they do not drift and are locked to a crystal. Less expensive a/d's like the one in the JB3 and M1/D100, etc. don't have crystal lock and might run at 44100hz for a while and then drift to 44106hz and then back the other way. This is why apogee sells studios units such as the "Big Ben" so that all their gear (with w/c in) runs exactly the same on each track
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damn id love a big ben :-)
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damn id love a big ben :-)
just put your email address in your setlists and soon your box will be full of emails telling you how to get a "big ben"!!....or at least how to add three inches LOL
I fucking hate spam
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the problem is with the accuracy of the clocks. Apogee is known for their rock solid clocks, the reason for this is they do not drift and are locked to a crystal. Less expensive a/d's like the one in the JB3 and M1/D100, etc. don't have crystal lock and might run at 44100hz for a while and then drift to 44106hz and then back the other way. This is why apogee sells studios units such as the "Big Ben" so that all their gear (with w/c in) runs exactly the same on each track
Good info. +T
-Kevin
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I think it's clock differences, I had the same problem working with DAT sources before. Now I won't touch a matrix unless the word clocks between the converters are sync'ed. The worst nightmare I ever had was syncing two analog cassette sources, constant varying drift, ahving to re-sync every 4 seconds at the worst parts....I just wanted to pull my hair out and quit......
I just went through this. Two DAT sources, the SBD source was off by .5 seconds if you lined them up end to end, yet it hopped all around throughout the source. I ended up cutting a 90 min SBD source up into about 40 samples, and lined them up piece by piece with the AUD source to get it to work.
Now that I am done with this, I am getting a mixer to do this in the field ;D
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Yeah, I feel the pain of doing this post-show. I have a UA-5 and normally matrix real-time, but didn't want to mess this show up, so I figured I'd set the mix afterwards. In retrospect, it's not worth it :/
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In retrospect, it's not worth it :/
Don't get me wrong...the end result has always been worth the pain for me.....nothing beats a tape like that, that I will enjoy for a long time after that....I just want to make it a little easier on myself going forward ;D
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the problem is with the accuracy of the clocks. Apogee is known for their rock solid clocks, the reason for this is they do not drift and are locked to a crystal. Less expensive a/d's like the one in the JB3 and M1/D100, etc. don't have crystal lock and might run at 44100hz for a while and then drift to 44106hz and then back the other way. This is why apogee sells studios units such as the "Big Ben" so that all their gear (with w/c in) runs exactly the same on each track
That's very interesting. I've experienced this as well but had a different theory...
We generally like to synch sources on some kind of sharp peak. Many of these peaks are direct sound, reflected sound (delayed) and p.a. sound (direct and reflected)
Keep in mind, when you're talking about a few milliseconds, you're talking about a few feet difference in location. The drummer being 6 feet behind the guitar - there's 6 ms. Both come out of the p.a. at the same time but the direct sound from the drums/guitar amp could offer different 'match points' for 2 sources. I think many house p.a. systems do introduce a slight delay so the drummer's natural sound (behind the p.a.) has a chance to 'catch up' to the p.a.
Reflected sound could also explain it - I've noticed on some drum stuff that you can actually see a 'double image' in the wave form - the direct sound followed by the lower amplitude reflected sound (you have to be seriously zoomed in to see this). Again, knowing that every extra foot of travel will result in a 1ms delay, it doesn't take much to create a few ms gap.
I wonder what would happen if we only looked at the same instrument for time alignment i.e. check the hi-hat at several points in the recording.
So anyways...that was my theory. I haven't done the above where you'd only look at the same instrument at various points in the recording. Didn't really think of it until I was writing this.
I didn't obsess over the variations because my ears seemed to tell me that +/- 5ms was undetectable when mixing 2 different sources. Instead, I just choose a happy middle ground and as long as I wasn't out by more than a couple of milliseconds, all was good. The reason I say different sources - for fun - try multitracking the same source twice and introduce a 5 ms delay - it gets really phasey and weird.
I suppose the other way to find out is to have 2 devices that can sync (either master/slave or both to another clock) and then see if the same thing happens - if so, then the above explanation seems reasonable.
Anyone have any real live experience with synched clocks? I was actually gonna ask about this anyways.
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that's the whole purpose behind synching clocks...
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SD 744T
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So more information on this thing. Taped again last night mics>UA-5>JB3. I glance over just to see how long he's been playing, and see the counter acting weird. I thought it was the angle I was at, so I get a better look, and the counter is going 1...1...2...3..4....4....4....5.6....7...
Counting up at irregular intervals. Really weird. The show was really late last night so I haven't had a chance to do any testing, but I imagine it'll be easy to pinpoint whether it's the UA-5 or the JB3 acting up. The sound itself seems fine.
Since the JB3 is being fed a digital signal, this surprises me. I guess I don't know enough about how the sampling works with a digital source. Wouldn't the UA-5 do the A/D and thus the clocking, and the JB3 just receiving samples at a certain bitrate?
J
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Counting up at irregular intervals. Really weird.
I've noticed irregular counting on my JB3 as well. I wouldn't worry about it, just a dufus counter. According to my testing it doesn't have any impact on the recording itself, strictly a display issue.
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I wonder what the counter is based on, though. If it's the JB3's internal clock, or the # of samples it's received from the external source at a given samplerate.
Regardless, I agree it won't affect the sound of the recording. I'm just curious if it's indicative of a larger problem. I will test tonight to see if I can reproduce it w/ and w/o the UA-5, and with the UA-5 and a different JB3.
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Just tested this. Both JB3's did it with analog input, as well as optical input from the UA-5 as well as my DVD player. It seems to correspond with disk access, but not always(at least I couldn't hear/feel disk access).
At any rate, yeah, it seems the counter display is just irregular. Go figure.
J