Hi RM, Corny again with a latency question.
I have a DMXR100 mixer and a HDR24/96. They
are tied via Lightpipe. I like to use the old trick of running all (or select)drum track and/ or bass tracks to a stereo pair which is sqaushed, pumped a bit and mixed under. So, I decide to simply bus them to 23/24 on the Mackie and record. What I got in return sounded really phasey, so I checked to make sure I did not have reversal on anything, then when I zoomed in, I ended up have to nudge those tracks over. Is this a common problem? Should there be this much delay in a system like this? What is happening when I am overdubbing? Same thing? Do I need to time allign everything? I checked all my frame delays too, everything was ok.
Thanks in advance,
Corny
ps, those 1176 bass settings are perfect! thanks! :)
Tags
Comments
Thanks RM, I'l try #2 and see what happens. I am about to chuck
Thanks RM, I'l try #2 and see what happens.
I am about to chuck the DMX for a Herrison or something and a Otari MK series or maybe a MCI. Your right, I see quite a bit more benefits to analog, I have bad cases of digitalitis and bass and guitar sounds are a serious struggle, I don't remember beating my head against the wall that hard back when I had my 1/4 Fostex 8 track, and a reel of 456... BTW, What is your main console and media? Lemme guess... a 9000 and a Studer with a side of PT? :D
Hey...don't chuck that DMX so fast....did you check the aux situ
Hey...don't chuck that DMX so fast....did you check the aux situation?
As for me....whatever. I do alot of writing/demo's/tv/budget-stuff all in PT.
When there's money and I have a choice, I prefer old neve's (80series), API's, ect. But I'll work on literally anything if I have to (even a mackie).
FlyingFaders is my favorite automation...but I know ssl well ( altough I haven't worked on them much since the J came out...mostly old neve's and neve vr's since then)
I actually just got done doing that. I sent it though the aux an
I actually just got done doing that. I sent it though the aux and patched it straight back into the return, contrary to runnning through an outboard- cause the only stereo comp I have is a 96k and the rest are mono's (1176 clone, 737, etc.) using the Comps on the Sony at the return. Same comb-ee sound!!! Wierd, it is starting to make me really wonder. Has to be in the board. So, I went back to Number one. It works, it just would be easier to do 2 as you suggested.
Do you then prefer the sound of the 80's over SSL?
Also, (off the topic again)I was wondering about the steroid drum sound, ala Bob Rock, maybe you could tell me what signal chain he is using to get that really smooth yet cracky mids on the drums. Is that going through API/ Neve?
Thanks for all your help.
Originally posted by Corny: Do you then prefer the sound of t
Originally posted by Corny:
Do you then prefer the sound of the 80's over SSL?
Neve;ala 8028,8048,8068,8078,ect
Also, (off the topic again)I was wondering about the steroid drum sound, ala Bob Rock, maybe you could tell me what signal chain he is using to get that really smooth yet cracky mids on the drums. Is that going through API/ Neve?
Neve 1081's as far as pre's/eq's on the drums.
hi corny, as far as i know the lightpipe is the problem. conver
hi corny,
as far as i know the lightpipe is the problem. converting the data to optical and back introduces a significant delay. just how much you've probably found out nudging those tracks. as it's only 2 tracks i would try going aes/ebu or coaxial spdif and check the difference.
Hello Corny, I just recieved an email from m.wagner. He's a
Hello Corny,
I just recieved an email from m.wagner. He's an owner/user of the sony and he said this, (relating to my belief that you could route analog aux send to gear to analog aux return):
"Unfortunately not on the DMX. As far as I know the Oxford takes care of
that for you. If I need to sub compress or treat the signal analog, I
come out of the M/A converters of the R-1 and then bring the signal back
in through the analog inputs of the Sony. It requires to delay all the
digital signals going straight into the board from the R-1, in my case
by 65 samples, other converters may differ in time. There is a delay on
each channel of the console, so it's not that big of a deal."
Thanks, I thought that was specific to my board, thats a relief,
Thanks, I thought that was specific to my board, thats a relief, but at the same time a bummer! I can't afford an Oxford. I was ready to send it to Burbank. I appreciate you going to the trouble of emailing and finding out, RM. That does mean though, that the Aux sends are delaying the signal if I am using analog effect, say a little reverb (I guess if I want a little pre-delay, it's ok). It also means if I am triggering a midi kit I am getting a load of delay as well. So I had better change to dital only interface (preferably AES) or nudge? Aslo, if I were to switch to a better converter then do my analog, could I get better results? Or am I still looking at the same problem, but less?
Thanks very much!
Corny
Corny Everytime you go through an A/D or D/A conversion you g
Corny
Everytime you go through an A/D or D/A conversion you get latency. The exact time will differ depending on which converter you are using. You might want to check on the timing of your specific setup. If you take a signal with lots of high end content, like a snare or click track, and send it digital from the HDR to the DMX. now send the same signal (same track on the HDR) out of the analog outputs of the HDR to the analog inputs of the DMX. Via the input routing page bring both inputs up on channles next to each other (makes it easier to adjust). Now phase reverse one of the channels and bring both levels as close as possibel to each other. Switch your channel delay to "WORD". Wile listening in MONO, slowly turn up the delay on the DIGITAL channel until the signals cancel each other. If they don't cancel completely you might have to match the levels better (pan one channel left and one right and check the PGM meters). If you still hear something, it's probably the difference between the HDR and the DMX converters. Once the signals cancel check how many samples it shows for a delay. That's your latency.
All your digital inputs should be set to that delay time (mine is 65 samples at 48K = 1.2 ms ) and the analog input should not have a delay. Now, when you bring back a snare through an anlog compressor and mix it together with the "digital" snare, it should be in time. Be cartain to check the phase (polarity) of the analog inputs on the DMX, as far as I know some of the earlier versions of the DMX had phase reversed analog inputs.
Hope that helps. Sorry Iif I'm stating the obvious.
Holy smokes! That helped a helluva lot. Thank you for such an ex
Holy smokes! That helped a helluva lot. Thank you for such an extensive and explanitory reply. I will folow this out tonight. As for the phase on the analog, I do have an older model and sure enough, they are swapped. Would you go in there and physically correct it? Or just leave it alone?
My gratitude :w:
Corny
leave it alone, just switch the phase on the analog channels. Un
leave it alone, just switch the phase on the analog channels. Unfortunately one more thing to think about. Actually two, because the analog AUX redturns might be out of phase with the analog inputs. You can do a similar test as described above to find out. Don't get me wrong, I really like the console and I'm getting great results with it, just a few quirks on the older versions.
Hey Corny! Hopefully someone with an DMXR100 can chime in he
Hey Corny!
Hopefully someone with an DMXR100 can chime in here (I know the big bro-OXFORD-but have only played with the DMXR100 on demo's of it). I think that there are analog aux's returns on that board...so that you could do just what you're trying to do....bus to an "effect" (in this case a compressor) and return it to the consloe with zero latency.
The problem with digital (in this situation) is the digital (quite literally). The act of computing, still takes more time ,thanelectrons running at the speed of light through a maze of copper and silicon (re: analog).
So until I ask someone I know, or someone chimes in ....see if there isn't an fx return that's analog on that board. That's the way to do it. Stereo Aux out - compressor(s)-analog stereo aux return.
If not then you have to either:
1. Print the compressed version and offset (like you have done already).
2. Have duplicate tracks of the source material pre-delayed (offset). Sending only this to the compressor and returning on a couple of faders and balancing against the original tracks.
Option#1 lock you in and slows down the mix; what with having to re-bounce to compenstae for any changes you may wish to make after you've doen it the first time.
Option#2 let's you do it in "real time" but just uses up way to many tracks/channels for my taste.
This is STILL one of the benefits to mixing in analog. :w: