SSlmixed.com Forum
Tech discussion => 4K,6K,8K Series => Topic started by: madmuso on August 19, 2012, 12:03:06 AM
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Hey guys,
new here, was sent here by Mata haze after he helped me out via email regarding a friends 6048 console. Hopefully i can spend some good time in here learning heaps.
My friend and I have been slowly figuring out how to track and mix on his console, we are making some great progress but there are a few things im not sure about. Bare in mind the console is at his place so I can sometimes forget the finer details when posting but i'll do my best.
First thing I wanted to ask is regarding the patch section, it seems the tape returns (from DAW in this case) are routed to 2 rows on 2 patchbays, one bay consists of "multitrack returns and channel line inputs" the other bay consists of "multitrack returns and tape monitor inputs". What is the reason for having the same signal from the DAW going to 2 rows of 2 different patchbays both labelled "multitrack returns"?
The other question is, which is the most common way of recording the "main stereo" output of the console back into your DAW? The former owner of this console had written "mix" on slots 21 and 22 of patchbay row L. So I assumed that this was the main mix out of the console, I ran patch cables from these 2 "mix" slots to slots 1 and 2 of the multitrack sends (row H) patch bay, but we werent getting any signal back to the DAW's converter inputs.
well, hopefully we can sort this last one out so we can do a few test mix's to get used to this beast! I gotta say, these consoles are great, complex yet kind of simple at the same time, if that makes sense. I can certainly see why they would have been so popular when released.
cheers guys,
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Howdy!
The reason there are 2 sets of returns is to accommodate both modes (RECORD and MIX)
In MIX mode, they're feeding the Channel line ins, in record mode they're feeding the Tape mon ins.
For printing back to 2 track, you'll see the patch row with outputs (On the top row of the patch) called LF, RF. They'll take the L and R outs to wherever you send them.
Stef.
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To elaborate on what Stefan mentioned, the multitrack returns via DL's 11 and 12, are hardwired to the patch in two different places. They go to row C which is 1/2 normalled down to line inputs on row D. Row C is then also linked to row J which is 1/2 normalled to the tape monitor inputs on row K so the multi returns appear in two places, either to feed line inputs or tape mon inputs depending on whether you are recording or mixing and which fader (large or small) you want them to be controlled by.
Looking at a printed 6K patch layout I have here, I notice that L21 & 22 are not actually labelled which surprised me as these two jacks along with the A, B and C stereo outs are all normalled to the 8 track inputs on M15 to 22 which end up on DL31, circuits 15-22. The answer lies on that patchbay but, how about plugging into one of the stereo outputs that feed recorders for now, such as L27 and 28, L29 and 30 etc? Also, investigate L10 and 11 as there seems to be two lots of inserts on the patch layout I have here which also doesn't make sense! Finally, I've just looked up DL 31 which has Prog L and R on circuits 21 and 22. These are fed from M7 and 8 which are normalled down from L7 and 8 so, my tip is to try L7 and 8.
I will be posting an article on SSL patchbay wiring in due course so look out for that in the how to do technical section. Hope this helps for now. Finally, I have a DL blank and DL 33 layout on my website in the SSL documentation menu if they help with other wiring issues (www.profcon.co.uk).
I should be finding an SSL power cable for Stefan rather than writing this but it's too dark and late to search in the outside shed plus I needed some patchbay therapy!
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Go Jim :-)
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Jim's the man! we're very lucky to have him here.
Stef.
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wow, thanks for the quick informative replies guys, realllllllllllllly appreciate it. :D
Stefan, I have a photocopy here of the patchbay layout but cant seem to find the LF and RF ports you mentioned. strange.
Jim, will try L7 and 8 and also L27 and 28, etc.
So, provided that the main out signal is present on ports L7 and 8 (or any of the others mentioned) simply patching it to the desired multitrack sends ports on patchbay H should see that signal go to the AD converters right?
Is this the most common way of printing your final stereo mix because I have heard of people running the stereo out to some nice outboard first (to slightly enhance it) before it goes back to the DAW. Im assuming then that the main stereo out of the console has an insert available or are they simply patching from L7 and 8 to their desired user patchbay port which has their fav outboard, then from the outboard patching back to the multitrack sends?
While on the subject of the patchbay I just want to make sure that my friend has his monitor/speaker switcher plugged into the correct port, currently it is connected to the "sls 39 and 40" ports on row N. He can switch between 2 sets speakers by pressing a couple buttons in the master section of the console so Im assuming its correct?
my head is spinning from all this patchbay talk! but I love it! kind of!
thanks again guys,
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SLS is the studio loudspeaker mate, that resides in the studio if you have it.
Your studio monitors are normalled to the DL 33, they are typically lablelled MONS.
Stef.
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Me again! As Stefan stated, the SLS (Studio Loud Speaker) feeds are just that. They come out of N39 & 40, are 1/2 normalled down to P39 & 40, go to DL 33 and head off to the studio live area so the musicians can hear what they have just recorded on a set of speakers in that room. This was common practise years ago but I have a feeling that they are hardly used these days. Musicians have to take their headphones off to listen and would perhaps like to hear the takes over headphones or even in the control room as the monitors there would be better and more inspiring in a more acoustically controlled environment.
I don't own an SSL anymore so, I'm writing this stuff from memory but I have a feeling that the reason you can't find the LF and RF outputs is because your console is a 6K. These dealt with 3 x stereo busses, A, B and C for music, effects and dialogue plus a straight L & R stereo mix which seems hard to find so far! The 4K was a 'normal' recording studio console involving stereo and quad feeds (left and right, front and back) so, LF and RF may not existing on your console.
The centre section patchbay wires (rows L&M and N&P) should, I think, all be labelled so, if you see a wire idented as P39 on a DL (33 in this case), it originates from patchrow P, jack number 39. Wires from patchbay points which go to the 2 centre section 104 way 651 Mrac connectors will also be labelled with the patchrow jack point references.
Mini outputs: Your 6K may just have 1 set of mini outputs (L39 & 40). To listen to an alternative set of mini's, you wire the 2 wires out of DL 33 and into a relay box provided by you. The Alt button on the 651 is just a straight switch contact which you need to wire out to the relay box via DL 33 I think. Once pressed, it would trigger the relay to switch between the different sets of mini speakers connected to it. SSL later had mini 1 and 2 outputs without the need for a client supplied relay box but it seems like your console may not have this facility depending on its age.
Stereo distribution: SSL supplied 5 stereo outputs for mastering or client copies on patchrow L27-36 on the patch drawing I have here. These were linked (or probably fully normalled) from M7 & 8, the L & R programme outputs. I will look through the info I have here on the looming specs and post something on that later.
The trouble with this stereo distribution is that, once you connect an unbalanced machine (a cassette in the old days) into one of the 5 outputs, it unbalances all the others so, beware of that. Some studios sent the stereo signals to a separate distribution amp which then fed the various 2 track machines.
I would suggest having your DAW stereo inputs wired separately on the patch. That way, you can use the console L & R inserts to go into a compressor for instance, like you mentioned, and cross patched into your DAW bypassing the group sends/multi inputs patchrow, H1-48. Unless you rewire the patchbay a bit, which could easily be done, this would take up 6 additional patchcords as it involves 2 cords from the insert sends to the compressor or other device, 2 back from there to the console insert returns and 2 from the main L & R outputs into the DAW inputs.
Again, this is from memory so please correct me anyone if I've got my thoughts wrong on this. I'm from the old school before DAWS were invented so I am somewhat ignorant regarding this matter! Haven't DAWS got a set of stereo inputs?
Hope this helps (and I'm accurate with my information) and doesn't confuse you even more!
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ok cool, its all starting to make sense, slowly! Im finding that the "old school" language in the paperwork and even on certain parts of the console compared to modern day descriptions is also playing a part in the confusion, but im getting there! Different working methods I guess, the SLS is a perfect example.
You mentioned that the 5 stereo distribution points would all become unbalanced even if one set of those outputs was connected to device that wasnt balanced such as "a cassette tape machine", I get that, thats all cool, but if I was to patch from M7 and 8 (the L and R programme outputs) into H1 and 2 (which feeds a new stereo track in DAW) isnt this keeping the signal balanced because we are patching within the consoles patch bay? And if I wanted to add a compressor before hitting the converters I could simply cross patch into the comp first, patch out of the comp back into H1 and 2? Or is it a MUST that you have to use the consoles actual insert point when wanting to strap outboard across the main stereo/programme output?
As far as the mini outs go, I think we will need to do a little bit of wiring as you have mentioned because the owner has 3 sets of monitors, NS10's, genelecs and questeds which are all in the control room.
There was another thing I was thinking about today in regards to fx returns. I used to have a small tascam console years ago and I used to simply return the outboard outputs to the line ins on any spare stereo (or mono) channels. Is it ok to do this with this console or do the fx returns HAVE to come in via a particular section of the console such as the "echo returns ports on M41 to 48?
Its about time i answered one of your questions now Jim! hehe. I use Nuendo and it allows you to create as many stereo inputs as you like, however the amount of channels your AD converter has will determine how many you can use at one time. I have 16 individual ins on my converters so for example, I can record the L and R outputs of 8 keyboards into 8 separate stereo tracks all at the same time. Or assign the converters inputs to 16 mono tracks within the software and record 16 separate guitars at once, which is 14 too many, I can live with double tracked guitars! ha!
Stefan, thanks for the SLS clear up!
Thanks heaps guys, you're making this learning curve less painful!
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As far as I can recall without looking up SSL looming documentation, M7 & 8 on row M are half normalled and not just linked to the stereo distribution L27-36. That means that, if you plug into M7 & 8, it breaks the feed to the stereo distribution (as above) on the patchbay right hand side so you can route the stereo signals into a compressor and then to multi inputs H1 & 2. You could also use the console insert points as well but may need to plug some bantam plugs into M7 & 8 to stop the feed to the distribution if you already have an unbalanced device connected into the stereo inputs on M27-36.
SSL clients had to supply a mini relay switching box years ago which was a pain as it involved more work for the installation people!
On the 651, there are 4 echo sends and stereo returns (via DL 33 I think). 30 years ago, that was fine but people started to return outboard through channels instead to take advantage of the many facilities on the modules plus the TR capabilities. The echo returns on the 651 are not recognised by the computer TR system. So, to answer your point, you don't have to return stuff through M41-48 but you could still use the 651 4 x echo sends and returns for your favourite outboard stuff and cross patch them on the patchbay.
Maybe I can explain the thinking behind the SSL patchbay layout in more detail at some point but not now. Basically, it was designed so that the minimum number of patchcords needed to be plugged in except for, perhaps the SMPTE code going to multitrack channels 1 or 24 on a 2" machine (another subject!).
There are many more stereo distribution units on the market these days which helps. I remember Metropolis in London specifying a change of patchbay wiring in about 1989 when they had their first SSL console to accommodate a balanced stereo distribution amp. Thanks for your info on the Nuendo.
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ok, I now have plenty of things to try out! so, I most likely am heading to his place on Sat to fiddle with the console some more, I will let you know how we go! fingers crossed!
Thanks again guys, awesome stuff :D
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Hey guys,
I went over to my friends place today and we managed to insert a stereo comp to the main mix. However im still a little unsure of a few things and heres why.
On row L21 and 22 the previous owner has written "mix" and beneath that on row M21 and 22 he has written "insert". I patched from this insert point into a stereo comp then patched back from the comp into the L21 and 22 "mix" ports. works fine. If this is all correct why wouldnt the patchbay be labelled from factory like that? or is this a sign of some modifications maybe? All I need to do now is find a set of L21 and 22 duplicate ports else where on the patchbay to send back to the DAW, this is proving difficult!
There were two other main issues I was hoping to solve, firstly, I still cannot find where the control room monitors should be plugged into in the patch, the only ports I could get any signal from was from the "programmme" L7 and 8 ports so I plugged it back into the SLS ports. Secondly, as mentioned, I still cant get the main stereo mix from anywhere in the patch to go back to the DAW via H1-48, so we cant print the stereo mix.
Depending on time I may head over there again tomorrow. So after today, the things I am wondering about are:
1) Is it ok to leave the control room monitors plugged into the SLS ports or do they HAVE to be plugged into some sort of dedicated monitor ports. I am thinking that due to the different modes of the console and the way the patch interacts in different modes that they MUST be plugged into the correct specific ports.
2) getting the main mix back into DAW.
Jim, I thought of what you said, having the stereo inputs of the DAW on a totally separate set of patchbay ports so i could, for example, patch from the "stereo mix" compressors outs straight into the DAW's stereo inputs, but then I cant hear what the compressor is doing to the song because the stereo track within the software cant be monitored or I will get a feedback loop. So I have to get this to work via the stereo mix buss' insert points so i can then simply make sure that im not clipping on the way back into the DAW, I can do this via the converters signal displays (as well as the softwares input displays) so the actual software track doesnt need to be heard/monitored, this is why I have to send the DAW an exact duplicate of L21 and 22 (which is what we ARE hearing in the control room).
I really wanted to print a stereo mix today but no luck! ah music, gotta love it hey.
thanks,
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Hi, I'm baffled by the lack of a straight L & R mix out from patchbay L/M as well. My SSL drawing shows L21 & 22 unlabelled but they are normalled down to M and then to DL 31 cct 15 & 16. On the same DL, it shows Prog L & R outs (from M7 & M8) go to DL cct (circuits) 21 & 22 . Try taking the patchbay out and look at the cable idents as this will give a clue. Also look at L/M 10 and 11 for more clues. I'm sure the answer lies with patch L/M as N/P deals mostly deals with returns to the console.
CR monitors should come out of L37 & 38 and mini's out of L39 & 40. I would imagine that, if you plug the CR mons into the SLS outputs as a temporary measure, they would cut out if you were recording as these are a playback function and, logically, would not be activated during a recording as they would spill onto the live room recording mics creating feedback.
To hear a compressor in the final mix before going to PT, you need to hear the results over the monitors of course before the signals leave the SSL as you suggested. Err, so far, I'm running out of suggestions but I would certainly recommend taking the patch out to identify the individual wire idents. Maybe L21 & 22 were rewired by the previous owner. If that is the case, you should easily see the difference in wiring and looming.
Hope this helps a bit more but, bear in mind I don't have an SSL anymore and my thoughts are based on memory and the odd bit of written documentation I have available. If I had your console here, I could very easily sort out these problems.
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thanks Jim, you've been a massive help. Im gonna head over there next week so looks like we will have to take the patch bays out to see what is going on.
I forgot to mention yesterday that for some reason on channel 17's insert return was working, all the insert sends work though, strange.
Also, beside the main mix console compressor, there are two identical vertical rows of buttons, beneath the left row is a button labelled external to studio and beneath the right row is "external to monitor", for some reason the entire right row has been disconnected.
Well, more problem solving to do next week, let ya know our progress, thanks,
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Regarding loosing an insert return, this is 99% the 5534 in the "Patch Return" circuit, on the Channel Amplifier card.
IC9 on a G series
IC5 on a transformer E series
Pop a new one in :-)
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It might be worth looking on the inter web for big pics of E Series patchbays? that's how I got a little heads up about mine prior to the console arriving..
Stef.
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Regarding loosing an insert return, this is 99% the 5534 in the "Patch Return" circuit, on the Channel Amplifier card.
IC9 on a G series
IC5 on a transformer E series
Pop a new one in :-)
thanks for the reply, so, im assuming this is a part that requires each of the actual channel strips to be raised out of the console and fitted? Doesnt sound like it would be too difficult. :-\ The owner of the console has already taken them out a few times to change other parts so hopefully we can get it sorted.
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Hi, I'm baffled by the lack of a straight L & R mix out from patchbay L/M as well. My SSL drawing shows L21 & 22 unlabelled but they are normalled down to M and then to DL 31 cct 15 & 16. On the same DL, it shows Prog L & R outs (from M7 & M8) go to DL cct (circuits) 21 & 22 . Try taking the patchbay out and look at the cable idents as this will give a clue. Also look at L/M 10 and 11 for more clues. I'm sure the answer lies with patch L/M as N/P deals mostly deals with returns to the console.
CR monitors should come out of L37 & 38 and mini's out of L39 & 40. I would imagine that, if you plug the CR mons into the SLS outputs as a temporary measure, they would cut out if you were recording as these are a playback function and, logically, would not be activated during a recording as they would spill onto the live room recording mics creating feedback.
To hear a compressor in the final mix before going to PT, you need to hear the results over the monitors of course before the signals leave the SSL as you suggested. Err, so far, I'm running out of suggestions but I would certainly recommend taking the patch out to identify the individual wire idents. Maybe L21 & 22 were rewired by the previous owner. If that is the case, you should easily see the difference in wiring and looming.
Hope this helps a bit more but, bear in mind I don't have an SSL anymore and my thoughts are based on memory and the odd bit of written documentation I have available. If I had your console here, I could very easily sort out these problems.
Tried the suggested CR monitor patch numbers, no luck, I tried the ones suggested as well as a heap of other ports on the patch and nothing, seems the only monitor outs working are the ones designated for the Studio. i will tackle this again soon, I have another question that I have been confused on for a bit, bear in mind im back at home and not in front of the console when writing here so I may get a few things mixed up, sorry.
as you guys are aware, each channel has 5 separate cue send knobs ( a stereo one, top of the row) followed by 4 mono ones, the mono ones are labelled 1-4 and the top knob is just labelled "stereo". Am I correct in thinking that during tracking these "cue sends" are used to send separate mixes to the artists, and during mixdown are the actual "sends" to fx units? The reason im asking is because from memory in the master section there seems to be a master "cue return area" as well as a master "echo return" area but no actual "echo sends" anywhere that I can see, so, im assuming that during mixdown the fx units should be patched into the echo returns as the master echo returns section caters and is designed to treat the fx during mixdown? Just another piece of the SSl puzzle im trying to put together, fun times!
As soon as I make sense of this then i can try to figure out where the 8 channel headphone amp gets its signal from, each channel on the headphone amp has a L and R stereo input so I have a bit of figuring out to do. yay.
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hey guys,
I got my hands on the manual and have been reading and taking notes like a madman. Im not sure if its my interpretation or conflicting details from different parts of the manual thats confusing me regarding the sends and returns, cues, etc. So here is exactly what the manual says:
SECTION 3-12 (channel strip details section)
Cue and Aux sends.
During the recording process the stereo and the first two mono buses may be used as headphone feeds. All the Aux send buses are found on the jackfield (N1-6) and can be used as feeds to effects units. The stereo and the first two mono sends are simultaneously routed back into the centre section (jacks P5-8) where talkback and additional external stereo feeds, via the CUES pot, can be added.
After the addition of talkback, stereo reverb from the stereo reverb returns, and a stereo external source the cues appear on the patch row N9-14. These outputs are then normalled to the feeds to the headphone power amps (jacks P9-14).
Cue 1L and 1R are derived from AUX 1, and CUE 2L and 2R are derived from AUX 2.
Section 5-4 (master section details)
Cue-Aux sends:
Each channel is fitted with one stereo and four mono auxiliary sends for use as foldback, echo sends or as mix minus feeds.
Auxiliary sends 1 to 4 appear on the patch N1-4 where they are normalled to the outgoing jacks P1-4 to be used as feeds to effects units.
Aux 1 and 2 together with the stereo sends are simultaneously fed back into the communication section of the master facilities module where talkback (via buttons in the communications section), stereo reverb (via the echo returns) and external stereo feeds (via the external to studio selector) can be added.
Three stereo Cue outputs then emerge on the patch N9-14 for use as headphone foldback. Note that Cue stereo is derived from the stereo send on each channel, cue 1L and 1R are derived from aux 1 and Cue 2L and 2R are derived from aux 2.
Section 8-2 (the patch section)
echo sends (N1-4):
These carry the Aux 1 to 4 outptus which are normalled down to the effects send lines (P1-4) out of the console. Jacks P1 and P2 are also wired to jacks N7 and N8 which, together with the stereo send outputs on N5 and N6 are normalled back into the console (via jacks P5-8) for the addition of talkback, stereo echo and stereo external sources. These Cue outputs then emerge on jacks N9-14 as three stereo cue sends.
So, am I correct in saying that you can only have 3 separate headphone mixes going on at any one time during tracking? And why have they stipulated 2 different rows for headphone outs?
Ive attached a copy of the patchbay layout so it helps when reading the above.
Im gonna head over there tonight to try to make more progress.
http://i135.photobucket.com/albums/q142/madmuso/patchrowsKLMNPRS.jpg
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heres the entire patchbay layout in case somebody wants it, im sure you guys already have it but I thought i'd scan it while I was at it.
thanks,
http://i135.photobucket.com/albums/q142/madmuso/6000sreiespatchbay.jpg
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Regarding loosing an insert return, this is 99% the 5534 in the "Patch Return" circuit, on the Channel Amplifier card.
IC9 on a G series
IC5 on a transformer E series
Pop a new one in :-)
thanks for the reply, so, im assuming this is a part that requires each of the actual channel strips to be raised out of the console and fitted? Doesnt sound like it would be too difficult. :-\ The owner of the console has already taken them out a few times to change other parts so hopefully we can get it sorted.
Exactly that, you can scope the channel through, but when you've had that fault 10 times before, and the fix is always the same.. hehhehe. If you run as faults book, it will help you with future problems.
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Hey guys,
we finally made some progress! :D
Inserts are all done and working.
We now have our nearfield monitors plugged into the correct ports on the patch (L37 and 38). And the monitor "source" buttons in the master section along with all the other monitor related routing (in the master section) is all working! The currrent owner seemed to be under the impression that the previous tech had disconnected parts of it for some reason, doesnt seem to be the case though.
As far as the main mix (programme) output goes, we are still using ports L21 and 22 which has "mix" written on them by the previous owner, and we are currently using M21 and 22 as the main mix insert (again, it has "insert" written on those ports by the previous owner). After inserting some outboard across the main mix I then patched it back into multitracks sends 1 and 2 and the signal appears on the AD converters! just what we wanted, however for some reason the RME hammerfall wasnt seeing it, but thats another issue I'll sort out upon next visit.
I am now going to move onto the send and return section of the console and try to nut it out. I am slowly starting to understand the console, with each session the picture is starting to become clearer, still a while to go though, but man, what a great design!
We are currently patching in every input and output from the outboard pre's to user patchbays, while we were discussing this yesterday a question arised regarding bass guitar recordings. You would normally plug the bass guitar into the high impedance input on the preamp usually found on the front (if the bass player was in the control room), but what if he is in the studio and you just want to record DI, does this mean you have to plug the bass into a DI box (converting it to mic signal), then from the DI into the stage box, then patch it in from the consoles "mic lines" patch bay into desired preamps mic input?
Thanks for all the help guys, between here and the manual im sloooooooooooooooowly getting it, thanks heaps!
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not to be too picky but can we use the term socket (patch bay socket that is) instead of port?
just to call things with their right name 8)
Mattia.
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sure thing, socket it is!
Guys, can someone tell me what the far right hand side of the console is called/referred to? Im talking about the section that has the two XLR sockets and some talkback features, the copy of the manual I have here doesnt mention this part of the console at all, perhaps the copy of the manual is incomplete, i dont know. I want to be able to read about this section to know how it works, etc. Also, a few of the buttons have been replaced by the previous owner but dont have anything written on them, they are just blank buttons so we are lost right now.
I want to find a picture and details on the net but I dont even know where to start because I dont know the correct name of this section.
thanks
Thanks,
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I think it's known as a 621 Producer Panel. Jim would know the variants, in my experience, it's an outmoded feature, and some musician always places a magazine on (or similar) latching the talkback open.
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Hi, as Sintech stated, this is probably an outdated feature these days. The Producers Panel section is all the 19" stuff (if fitted) at the other end of the console to the patch. This could be for mounting client equipment or SSL supplied items like a stopwatch, 693? printer (I'm not kidding) or even the 621 3u Producers Panel. This was meant to be for a Producer to listen to or talk back to various places.
The 621 Producers Panel has a built in mic on the left and switches to talk back to the MD (Musical Director), TB Cues (TalkBack Cues …so the musicians can hear what the Producer is saying) and Omni, which talks to everywhere and can be dangerous (hence in red) as the Producer could end up swearing on tape!!! I can't remember exactly what the Dim switch does but I think it reduces the talkback level from the 621 panel by several decibels.
The listen mic switch allows the Producer to hear what the musicians are saying in the studio via the listen mics in there (which appear on the patchbay) and there are two sets of controls to allow the Producer and others to listen to cue stereo, auxes 1 and 2 and the monitor mix via the headphone jack socket which is linked to the 3 way XLR. I think the jack tip was wired to pin 3 on the XLR judging from a picture I have.
The blank buttons above are user options and are not wired. Later versions of the 621 panel could be remotely wired elsewhere in a Control Room via DL 59, hence the gap in DL numbering if you haven't got this facility. The 24 way Bicc plugs into the console loom or DL 59 if fitted.
One amusing story on this. About 1981, the SSL buyer bought the wrong type of jack socket and quite a few panels were made with type B jacks (GPO versions). Only in final test did this get revealed as the headphones didn't work as 1/4" type A jacks are different to type B meaning some contacts shorted or were open circuit when plugging in a standard type A jack on a headphone lead to the type B fitted on the panel. Type A versions are commonly used in the recording industry of course and type B are really mostly used for older broadcast type patchbays so, beware in future!
Hope this helps from Jim Lassen.
(http://i1070.photobucket.com/albums/u485/jiml6570/SSL%20Other%20items%20for%20sale/IMG_4436.jpg)
(http://i1070.photobucket.com/albums/u485/jiml6570/SSL%20Other%20items%20for%20sale/IMG_4437.jpg)
(http://i1070.photobucket.com/albums/u485/jiml6570/SSL%20Other%20items%20for%20sale/IMG_4440.jpg)
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wow, once again, cool routing! Thanks for the pics too Jim.
So, is this the SAME talkback mic thats activated when you press the "cue stereo", "cue 1" or "cue 2" in the communication section, or is there another inbuilt talkback mic somewhere on the console?
While on the subject of talkback, in the communication section, is it possible to engage the "cue" and "listen mic" buttons together so you can have a discussion with the artist in the studio or do you have to toggle from one button to the other during conversation?
As usual, thanks for the help! Another piece of the puzzle is complete!
:D
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hey guys,
we have been busy patching in all the outboard at my friends place, what a task, taking forever but so worth it having all the outboards in and outs on the patchbays.
We have made a lot of progress as far as understanding and operating the console goes so its all coming together, the more I learn and play with this console the more I love it! Problem is, now I cant see myself using anything else ever! Time to start saving.
I have a question regarding the reverbs units which we are hooking up to the empty user patchbays. Some units have one input only and 2 (L and R) outputs, which I understand, you would send for example a mono snare drum channel to it and have the reverb come back stereo to 2 faders on the console.
Some of the reverb units have a left AND right input, when would you use the right input? IS it for example, if you had a left overhead on one channel (panned left) and a right overhead on another (panned right), you would send the left channel to the reverb units left input and the right overhead to the units right input (and the reverb units returns would be panned accordingly). I would love to hear peoples suggestions from experience as to which is the best way to hook these units up.
Also, we have discovered that one of the cue channels has some serious hum happening, its plasma is always showing the hum present. You cant hear it through the control room monitors ( the consoles monitor and channel paths seem to be ok) but once we plugged the headphone amp in to experiment with cues it was really there and very annoying. Any thoughts?
thanks, hope everyone is doing well!
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Hey guys,
does anyone have or has anyone used this device? Looks like it could make life easier when integrating the console with Nuendo.
Its called Konnektor.
http://www.youtube.com/watch?v=s5M-RzOTcW0
By the way, we are almost done patching the outboard to the consoles patchbay, what a task. Im starting to do research on the best way to sync the 6000e to Nuendo and Apogee clock and converters.
thanks,
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Hi mate,
I'm using The Konnektor here, it works perfectly and I highly recommend it.
Stef.
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Hi mate,
I'm using The Konnektor here, it works perfectly and I highly recommend it.
Stef.
awesome, just what I needed to hear, I will inform the console owner.
thanks,
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Hey all,
hows things? hope everyone is doing good. Its been a while since I last visited, last few months have been crazy busy.
The studio is coming along and we should be ready to roll the console and outboard in there at the end of this month, if all goes to plan.
I have been thinking about headphone mix distribution in relation to the 6000e and have a few questions.
The complex has a control room, large live room studio, and a smaller room studio. From what I understand, the 6000e can only give you 3 separate stereo headphone mix's (cue LR, aux 1LR and aux 2LR). Does this mean that if you were recording a live 6 piece band that 2 of the performers would be receiving one of the 3 mixes via the headphone amp? (assuming we have a headphone amp capable of receiving 3 stereo inputs which can be assigned to any of its 6 headphone channels).
I would love to hear if there are any work arounds or handy tricks in relation to headphone mixes during tracking with this console.
Ideally we would like to create an environment where the artist has their own heaphone mix controller so they can adjust to their liking but im not sure this console really caters for that, or am I wrong? Im used to creating headphone mixes in software land so setting up headphone mixes at the start of a tracking session using the consoles aux's doesnt really bother me either. Any suggestions are greatly appreciated!
thanks guys,
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Hi,
I have recently put in a Behringer P16 system, it works great and definitely worth a try. Try to ignore it's Behringer and try it.
With the console, I use the multitrack group matrix to get the tracks to 1-16 of the cue system. (Press INPUT and OUTPUT and use the short fader to send to the group)
Worth noting... I am 98% of the time running outboard pres and running the console in "MIX" mode. This might a bit fiddly if in RECORD mode but with some thought it will still work.
Stef.
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Hi,
I have recently put in a Behringer P16 system, it works great and definitely worth a try. Try to ignore it's Behringer and try it.
With the console, I use the multitrack group matrix to get the tracks to 1-16 of the cue system. (Press INPUT and OUTPUT and use the short fader to send to the group)
Worth noting... I am 98% of the time running outboard pres and running the console in "MIX" mode. This might a bit fiddly if in RECORD mode but with some thought it will still work.
Stef.
Thanks for the reply Stef. For tracking, We will be using the console in "rec + VCA to fader" mode and we will be using outboard pres almost all of the time, we will also be going "direct" from the small fader to our converters (into DAW). So since we are using the small fader as our "to tape/DAW" level control this means we cant use it as a send to a headphone system yeah?
So in your situation you are patching whichever channels you want from the patchbays Group outputs to the P16's inputs which I assume you've routed into your patchbay?
Stef, can you elaborate on your tracking setup/method. Im curious to know how you are sending to tape/daw and where the "from tape" monitor signal is coming back on the console, will help me see the overall picture, thanks heaps!
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I normally track in Split Mode: "Mix" + "Record" + "VCA to Monitor" this gives the best of both worlds, especially on a large console.
I would set the P16 on Groups 17-32, and pull my off tape/tools mix and feed the P16 from this. (although just normally use a more basic headphone arrangement)
Ch17-32 in "Mix" bring back cross-patched multitrack returns (Tools/Multi 1-16), and feed the headphones, and the Large faders would be my Control Room mix
All outboard mic pre's are patched to the "Insert Returns" on CH1-16 in "Record+VCA" (Hit Ready Group or Tape to go from "mix" mode into this) then routed to tape via "Direct" tapping into Shaping EQ and a little Catching peaks on the way, if needed, Via small faders
Slick'n old skool SSL :-)
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I normally track in Split Mode: "Mix" + "Record" + "VCA to Monitor" this gives the best of both worlds, especially on a large console.
I would set the P16 on Groups 17-32, and pull my off tape/tools mix and feed the P16 from this. (although just normally use a more basic headphone arrangement)
Ch17-32 in "Mix" bring back cross-patched multitrack returns (Tools/Multi 1-16), and feed the headphones, and the Large faders would be my Control Room mix
All outboard mic pre's are patched to the "Insert Returns" on CH1-16 in "Record+VCA" (Hit Ready Group or Tape to go from "mix" mode into this) then routed to tape via "Direct" tapping into Shaping EQ and a little Catching peaks on the way, if needed, Via small faders
Slick'n old skool SSL :-)
thanks for the tips, thats a cool setup too. Question though, how do you use the consoles talkback features/buttons to communicate with the artist, the consoles talkback mic is routed only to "ST cue, aux 1 and aux 2" sockets on the patchbay isnt it?
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thanks for the tips, thats a cool setup too. Question though, how do you use the consoles talkback features/buttons to communicate with the artist, the consoles talkback mic is routed only to "ST cue, aux 1 and aux 2" sockets on the patchbay isnt it?
Hi, the Omni button talks to the matrix also, as well at all cues.
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thanks for the tips, thats a cool setup too. Question though, how do you use the consoles talkback features/buttons to communicate with the artist, the consoles talkback mic is routed only to "ST cue, aux 1 and aux 2" sockets on the patchbay isnt it?
Hi, the Omni button talks to the matrix also, as well at all cues.
ah ha! excellent! That solves that then!
I didnt realize until today that the p16 system isnt just the monitor unit on its own but you also need the p16I rack unit to make it all happen with the console, so the owner is going to purchase all that this week sometime. yay!
Ive been thinking about your setup, is there any reason why your pre's are patched into the insert returns? Will this setup still work if they were patched into the line ins?
Also, where are you patching your 1-16 mutlitrack returns to? Into the line ins of ch 17-32? If so I assume that you then patch from group outputs 17-32 to your p16's inputs (on a user patchbay).
So with this setup, on a 48 channel console, you can record 32 mono sources and have them coming back to 16 channels. Sounds good!
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My signal chain usually (when recording)
Mic -> Mic Pre -> 192 IN -> 192 OUT into console in MIX mode.
I then decide (depending on the session) what goes to what group for the CUE.
With the cue it would typically go GRP 17 -> CUE input 1, GRP 18 -> CUE input 2, etc etc.
Stef.
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My signal chain usually (when recording)
Mic -> Mic Pre -> 192 IN -> 192 OUT into console in MIX mode.
I then decide (depending on the session) what goes to what group for the CUE.
With the cue it would typically go GRP 17 -> CUE input 1, GRP 18 -> CUE input 2, etc etc.
Stef.
Ah ok, so you arent using the console for any part of the "front end" side of things, you are bypassing the console and going straight into your converters and using the console strictly for monitoring and mixing. cool. My small setup at home is like this, but no console involved.
Im thinking that we may set things up similar to sintech's setup, this way we can run the console in "mix + rec" so we can then hit "ready group or tape" to activate the desired channels into "rec" mode. I personally like going direct from outboard pre's to converters bypassing any consoles but the owner of the console wanted everything patched into the consoles patchbay which is fair enough, its his gear!
Stef, do you have all of your outboards ins and outs hooked up to "separate/user" patchbays so you can easily create the signal chain you like before going to converters? Or do you actually get behind the gear/rack and patch it all in from there? The latter is how my small home setup is but in the future im going to have all the ins and outs available on patchbays, for me it becomes a pain in the ass to have to get behind it all the time for different signal chains. I dont have a lot of outboard at home but its still a pain!
thanks for the help guys, really appreciate it, speak soon and keep well,
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Howdy.
I have had a patchbay added in the console for all of my outboard so I come from the mic patch (very top) to the outboard pres.
I will probably use a couple of the SSL pres for a date in February where we're cutting live rhythm tracks for an album.
Stef.
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cool, sounds like fun!
Ive been thinking, there's really nothing stopping us from bypassing the console's channel strips during tracking, we can just patch from the mic lines (top row) to the outboards ins (on our user patchbay) then from the outboards outs (user patchbay) to the converters inputs which come are on another patchbay, then just do the monitoring thing in mix mode with the groups sending to the P16.
Obviously the beauty in using the channels is being able to take advantage of the roll offs, phase switches, eq and dynmanics if you really need it on the way in. I guess who ever is producing/engineering at the time can work whichever way they want.
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thanks for the tips, thats a cool setup too. Question though, how do you use the consoles talkback features/buttons to communicate with the artist, the consoles talkback mic is routed only to "ST cue, aux 1 and aux 2" sockets on the patchbay isnt it?
Hi, the Omni button talks to the matrix also, as well at all cues.
ah ha! excellent! That solves that then!
I didnt realize until today that the p16 system isnt just the monitor unit on its own but you also need the p16I rack unit to make it all happen with the console, so the owner is going to purchase all that this week sometime. yay!
Ive been thinking about your setup, is there any reason why your pre's are patched into the insert returns? Will this setup still work if they were patched into the line ins?
Also, where are you patching your 1-16 mutlitrack returns to? Into the line ins of ch 17-32? If so I assume that you then patch from group outputs 17-32 to your p16's inputs (on a user patchbay).
So with this setup, on a 48 channel console, you can record 32 mono sources and have them coming back to 16 channels. Sounds good!
The reason for patching an outboard Mic Pre to in insert returns, is not to go through the line amp/longer path than needed, so really just utilising the consoles routing, filters an the like. This is the way I saw people doing it in large studios in the 90's.
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Hi Dudes. Yes, you can bypass the SSL mic amp input section by plugging your favourite mic pre into the SSL insert as Sintech mentioned. Heck, why plug it into the SSL input as well if you think you have a better front end instead?
So, plug your mic into your 19" gear (which could appear on the patch) and cross plug the line level output into the SSL insert return. This is a shorter and cleaner feed into the console. Route the signal, use the cues accordingly and take it out to the DAW convertors so you just use the SSL eq, if required, and whatever else you need. I hope I've got this right ... I'm stating this from a distant memory some 15 years or so ago!.
Also, did you know that the mic sources (row A) on the SSL patchbay are fully normalled to the mic inputs (the only patchbay that is wired like this on the console)? This handy bit of wiring means you can also use the mic sources row as a tieline back to the studio for possible cues. If you plug a jack from a cue feed into row A, it breaks the feed into the SSL mic input (row B) below so you can send your signal to the studio wall panel for some sort of cue mix. Not many people know that but, you do now!
Hope this helps from Jim Lassen www.profcon.co.uk and now on FACEBOOK (http://www.facebook.com/pages/ProfCon-wwwprofconcouk-Professional-Connections-Jim-Lassen/371183312969924?ref=hl/)
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Happy new years guys! Once again you have provided handy info and tips! :D
We finally rolled the console and outboard into the studio today, over the next few weeks we will be busy hooking everything up, as well as soldering tie lines, etc.
I have a question regarding the "listen back" mic sockets on the patchbay. I understand that the listen back mics are the microphones you setup in the recording space/studio rooms so that the artist can talk to you. If I plug a microphone into channel 20 of my stage box, do i then simply patch mic line 20 into the "listen mic 1" socket? I noticed there are two listen mic sockets, what would happen if you had 4 separate rooms and needed to talk to all the artists, would you simply plug a mic into the stage box and bring it up on a channel on the desk and mute/solo as needed?
I have also been thinking about mult sockets on a patchbay, is there any difference between a patchbay that has been wired to mult a signal and a simple "y" patchbay cable that would duplicate a signal? just curious.
thanks, hope everyone is well and making plenty of music!
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You can have mults on a patchbay. This may be a German expression but I (and SSL) call them parallels. These are 3 or 4 jacks linked together so that a signal can be plugged into one jack and you can then feed 2 or more inputs from there. It is normally fine with line level signals but, with mics, it may not be a great idea as there will be impedance issues if nothing else. Similarly, for stage use, some people split the stage mic to feed both the Front of House and the on stage monitor mixes without using proper splitter units like an XTA device. Not a good idea really.
You can cross plug your stage box mic sources into the SSL listen mic inputs but, as you say, there are only two available. If you have 4 rooms, you may need some form of distribution box or splitter so that you can press a button to access any one or two of the listen mics at any one time.
If my memory serves me correctly (from about 30 years ago since I last operated an SSL), the listen mics are different in so much that they allow musicians to talk back to the control room but, when the console is in record mode, they automatically shut off to avoid feedback. If you just bought a mic source into a channel input, you would hear the musicians but the mic would not automatically shut off when you then went back into record mode. You would have to remember to mute it as well otherwise, if left on, it could act as an ambience mic and also be recorded to tape plus could be accidentally routed to cues and groups.
In summary (unless I am wrong here), use mults (parallels) for sending a line level signal to various places or use a Y bantam patchcord but avoid passively sending mics to two or more sources by the same method. That is one of the main reasons that rows A and B are fully normalled as opposed to half normalled like all the line level signals are on the console. This means that, if you patchcord mic source 1 into somewhere else, you break the feed from mic source 1 to mic input 1 as they are fully normalled. The console was designed so that you could not feed any mic sources into more than one input due to this different normalling method.
On a slightly different subject, the Omni button on the 651 or 621 (Producers Panel) is coloured red simply because, by pressing it and activating the built in mic nearby, it then sends the signal, via the mic, to all the cues (and maybe other places). This could mean that someone in the control room says "that was a bad take" and it may end up recorded on tape (or ProTools these days).
SSL designed all these functions nearly 40 years ago for a good reason. It is for us to realise and understand that today! They were not the world leader for around 15 years for no reason!
I hope I am accurate with my answer and it helps. I'm not a tech, just a wiring guy and a drummer so, excuse me if I got it wrong!
Jim Lassen (www.profcon.co.uk). Also on FACEBOOK (http://www.facebook.com/pages/ProfCon-wwwprofconcouk-Professional-Connections-Jim-Lassen/371183312969924?ref=hl/). Select the photos, click the albums and view a photo for more information.
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cool, thanks Jim. we have plenty of mic pres so if the listen mic sockets are expecting line level I can just cross patch into a pre, come out of the pre into the listen mic socket, no big deal. I think I need to investigate if there is a switcher or even a small rack mixer that I can plug 3 or 4 mics into, then patch the main out of the mixer to the listen mic 1 input socket, this way I can just mute whatever I dont want to hear on the little mixer/switcher,etc.
Speaking of switchers, we will need one for all our monitors, there will be 4, possibly 5 sets of monitors (2 or 3 pairs of nearfield with sub and 2 sets of mains). I was thinking to plug a switcher into the "mon LS" socket on the patchbay then we can just switch speakers from the switcher. Anyone using anything like this that they can recommend?
Actually, while we are talking about the mon LS socket, I recently read in the manual that the listen back mics appear on the "mini LS" socket, does this mean you always have to have a set of monitors hooked up to the mini LS socket to hear those mics? why wouldnt the listen back mics be routed to both the "mon LS" and "mini LS" sockets? or am I misunderstanding the manual?
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The SSL Listen mic inputs are mic level, not line level I believe so you can't just use a small mixer to accept Listen Mic sources from a studio area and send them to the two available inputs on the console as they would be line level from the mixer. Perhaps you could have some switcher box to allocate the studio Listen Mics and switch them to the console.
Most people use LS switcher boxes and the older consoles at least had a relay output which, when the Alt button is pressed on the 651, switches between 2 sets of mini speakers. This would have to be wired to an external box along with the mini feeds. These two Alt contacts would be wired from a DL, probably one of the centre section ones (31, 32 or 33) on 2 DL pins.
I'm surprised that the Listen Mics just appear on the mini LS and not the main LS although it may be to do with possible feedback within the control room. I'm not sure whether you have to hook up some mini LS to hear the Listen Mics or not.
Again, bear in mind I haven't used an SSL for a long time so my memory may be wrong!
Hope this helps somewhat. Regards from Jim Lassen (www.profcon.co.uk).
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Yeah, I was surprised when I read that about where the listen mic's come back to (the mini LS) oh well, we will have to sort something out.
As far as the mults go, I recently read about radial engineering's chain drive 500 series unit which is a 1 x 4 line level splitter, im thinking that maybe a few of these in our racks would be beneficial because we would be able to use it in different ways for guitar tracking as well as multing signal from the ssl's patchbay.
Also, ive been thinking about utilizing plugin effects as well as the outboard gear we have. Can anyone recommend a way to have the aux sends on the console feed an fx plugin in the DAW. Im a little confused with how to get this happening with the SSL. I'll explain how I think this is done but I think im wrong.
Would you patch the consoles aux send (output) socket to the converters input socket (multitrack send patchrow) then in the DAW, assign that converter input channel to the plugins channel, then route that plugins channel out of the DAW (tape returns patch row) back to a spare channel on the console.
This is the only way I can think of how to do this. Am I on the right track!?
Im also curious to know any work arounds that people have developed if they ever run into the situation of not having enough aux sends on the console, I think once I start using this console this will be a bit of a problem for me because im so used to working in the box which isnt as limited.
thanks guys!
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Hi, regarding limited amounts of Aux:
You have 48 channels, so I'd use the groups as extra FX sends via the small fader. This is very common practise.
Patch the group output to the FX unit, and return it on another set of small faders.
Regarding PT's plugins, yeah, that's exactly it:
Patch from the aux out to multitrack return that corresponds to the Protools input you want to use, make an 'Aux track' in the box, insert a plug in, set the outputs/spare set of outs (It helps if you have say 64 I/O) to a pair of small faders.
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yeah, we will definitely be using the small faders as extra sends, your suggestion of bringing the return to small faders is great, I always think to bring back returns to the large fader in case i need to automate, but I suppose its a good idea to take advantage of the extra 40 inputs you can get with the small faders in mix mode!
I have been going over and over the manual to make sure that things are sinking in but there is one thing I cant get my head around.
I like to use a lot of parallel compression during mixing in my DAW, in software land I simply make a duplicate of the channel then apply different dynamics treatment, so on one fader/channel I have the original, and next to it on another fader/channel the one that has further/different dynamics treatment.
How is this best achieved on the console, where I can have a slightly tweaked original on one fader, then an exact copy of that on the next fader so i can tweak some more and blend to taste with original? I tried the "float" and subgroup way but the large fader on the "original" becomes disconnected from the signal. The only way I can think of is to make a copy of the track in the DAW, send each one to two different channels on the desk and make sure that BOTH channels have all knobs and buttons in exactly the same position, then I can treat one further/differently from there, this will work but im not sure how accurate it will be.
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It's always going to be making the best of the console's real estate.
On a small console (24 channels), and lets say I wanted to use a TG1 limiter as the drum parallel, I'd actually use the stereo cue as the send(to the parallel), to preserve as many small faders, and bring the limiter back on two small faders. The send is post fader, so it will follow the balance whist automating.
I still have 4 Fx sends for verbs, and I always use them mono, and parallel them into anything stereo (via the patchbay parallel)
If I wanted to treat on single track, I'd do a 'cut and split' : Say I have a bass guitar on channel 14, and I wanted to add a thick fat compressor as a parallel blend.
I'd take the insert send from Channel 14, send that to say an LA2A, then return the output to Channel 14 monitor input, bringing the LA2A back on the small fader.
So I'd have the original on the Large fader and the parallel compressed version on the small, with just two patch cables.
If I wanted to add an 1176 on to the large fader insert, I could use a Parallel on the bay to split the feed to the La2a, pre or post the 1176.
Get some extra parallels on your bay's :-)) you can never have enough.
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You can pull feeds from the Tape Returns, without affecting the original signal path, (Tape return 14, to Line input 15) so this costs 1 patch lead, and saves one PT output.
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when you say "you parallel them via the patchbay" do you mean you are using the MULTS on the patchbay?
We are going to invest in some radial 1 X 4 splitters, a recent unit they've done, not only for console/mix multing duties but other studio duties that may come up, this way I can just unplug it and take to any of the live rooms if needed.
your 1176/la2a tip is great! Thanks so much! Thats just opened my eyes to the possibilities! Im loving this place! Thats the thing with manuals, they tell you how to use the console in a conventional way but it doesnt teach you how to exploit the console even further. I guess it also makes a difference when you have used one for a while and you learn to work around a units limitations. I cant wait to dig into this console.
Thanks again all! greatly appreciated! chat soon
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Hey, yeah, we call MULTS, Parallels in the UK :-)
Most studios I used to use would have little blocks of 4 bantams wrapped in heat shrink, that could be used in the ultimate event of running out of console bay parallels/mults.
I know Jim has all the stuff to make this kind of thing up.
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cool, thanks again guys!
I have been trying to find documentation on the correct power up and power down method for the console. I also want to have a look at this paperwork to see what the various faults on the power supply fault indicators mean.
we discovered yesterday that the fault panel was displaying a fault, the "250V plasma" LED's for both power supply A and B were not lit, all the other vertical LED's were lit. However the console seemed to be working ok, even the plasma's. We didnt hear the fault indicator because the "audible indicator" was switched to mute. Is there anywhere I can get a copy of the paper work relating to the consoles power supplies? It seems the previous owner didnt have it or simply forgot to supply it.
We also discovered that the omni communications button doesnt work in the centre section which kind of screws up our plans to use the group outs to feed the headphone matrix amps in mix mode, however the omni button on the producer panel works! Would have been good if it was the other way around! I'll have to sort it out after we have solved the power supply issues.
thanks guys,
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more issues :(
I noticed yesterday that after power up, the cut/mute leds on the monitor/small fader paths stayed on, regardless of whether the cut button was depressed or not. In order to hear audio on that patch i had to press the solo button. Anyone encountered this?
thanks,
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Hi, one channels Logic must be stuck on, forcing all the others to cut.
You'll have to isolate that channel to find it. I'd do this by powering on/off buckets to find the bank of eight, then pull the modules in that bucket until the problem goes, and the channel in my hand is the patient.
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On friday the 13th channel 13 on a 8k I was working on shorted out the solo bus, of course the same day
a big mix client came in to start a big session! I started by pulling the ribbons on the center section and
weeded it out pretty quick then had a scotch with the engineer!
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thanks Sintech,
as soon as Im done making tie lines and other stuff I'll see what I can do. Thanks heaps.
XMAX, I dont EVER want to experience anything like that!
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Tie lines almost done! One more box to do.
I was soldering today and because I was making balanced and unbalanced cables it occurred to me that all this time I have just assumed that all the inputs and outputs on the 6000e patchbay are balanced. I went through the patchbay section of the manual today but it doesnt say. Does anyone know? Is the entire I/O (every patch row) of the console balanced?
thanks,
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quasi balanced for the most part, but yes, hot/cold/shield on all connections.
i would recommend lifting the shields on the non-patchbay side xlrs to avoid ground loops.
(this doesn't work for mic lines and mic pre inputs though, and i would keep the monitor feeds shielded too). just bend them over , so you can always connect them again later.
gets rid of a lot of hum issues.
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Hi Brewery,
Grüsse aus Gießen nach Köln!!
What does "lifting the shields" mean? And "non patch bay XLRs" meaning the DLs from the patchbay that lead to my outboard rack, right?
Cheers,
Marc
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Hi, Jim would be the best person to advise, but.. as I'm installing mine right now.
As standard: The Insert send and return's on Row E and F, only connect hot and cold to the outboard gear, the screen is disconnected at the patch point. From the bus card to the patch point, the screen is present.
The outboard should be screened to the patch point as standard.
General idea i guess, you play with the screens if you have any earth issues.
So far, I've only had to cut the XLR screen on my Neve 33609 inputs.
I've made a pair of small XLR mic leads, with the screen disconnected, patching them inline to see if it helps or not.. this saves a lot of cutting and re-soldering.
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Hi People.
First of all, could I suggest writing a new subject header into posts sometimes as I'm a bit bored by seeing the same old 6048e title (in this case for example)? In nearly every post I do, I change the subject title to match the change of thread. Sorry if I seem out of line here but it is 7:30 am, I'm very tired and last nights dinner is stone cold (plus I'm a grumpy git sometimes!).
Moving swiftly on:
The subject of studio signal grounds could create a whole new menu possibly involving up to 6,439 posts as this subject involves many vastly different views and I believe there is still no known proper answer as the overall studio grounding scheme (if any!) plus the individual equipment AC and signal ground layout wiring has an affect (technical, domestic earthing and equipment circuits, pin 2 or 3 hot balanced gear, unbalanced gear, position in racks, distance apart etc).
As a result, I won't attempt to deal with this now as it is too complicated.
I'll address a few points here based on my memories of SSL wiring plus other issues but intend to start a new post on SSL grounding when I have time.
1 All older (4 and 6K at least) SSL console connections to the studio (at least via DL's) were wired balanced. I have no knowledge of any being unbalanced unlike some cheaper consoles!
2 I am quite sure that all the internal wiring from the patch to any part of the console (channel 6 way Biccs, 651 centre section MRAC's, PCB's and other functions) were wired with the screen NOT connected at the patchbay but all the channel Biccs had their screens connected which were then connected to one of the copper bars running through to the connector panel end OBA (M6) grounding stud.
Only the 96 way DL connectors (which were always wired as balanced circuits) were wired to the patchbay with the hot, cold and screen wires connected at both the DL and patchbay ends whether they were mic's, multi send/returns, centre section echo, cues, monitors, stereo etc or outboard ('user option' in SSL terminology).
3 As the last few posts show, some experimentation is advisable. Use phase reversals on your patchbay (to experiment between 2 or 3 hot balanced XLR connections) plus a made up lead with the screen 'lifted' (not connected) to address any signal grounding issues.
4 This is a complicated issue and the whole studio and building earthing needs to be known before a proper answer can be given. As Sintech states, 'play with the screens if you have any issues' Brewery states 'lifting the shields on non patchbay side XLR's'. This is probably okay as XLR's are balanced connections but, if that piece of balanced gear is pin 3 or 2 hot and is then plugged into a device which is unbalanced, you may have lost your signal ground (big possible future topic).
5 As a general rule for now, I would say, keep all the patchbay, panels and other connections in a studio fully wired and only disconnect screens (shields or drain wires) at the studio equipment ends (outboard, ProTools etc) if really required if you have any problems. Don't cut them off as you may need them if or when you change your equipment in future. I will post some more findings on this in due course as I am about to complete a studio install shortly and will investigate the whole grounding issue in more detail. I intend to look at an SSL console, document the signal grounds and AC earthing, measure impedance differences within various equipment (signal screen verses metal case and AC earth) and come up with some sort of cunning plan!
Decades ago, I remember that there were two rules here in the UK. Either ground the signal at the source and lift at the destination or the other way round. I don't think that either of these applies especially now when everything has got much more complicated and involves far more signal formats than way back then when it was all analog.
6 Never lift a signal screen on a mic feed throughout the circuit otherwise the vocalist may die and you may be in court.
7 Never lift an AC earth on a piece of equipment as it could also result in death. I've seen this done in some big studios in the past and it is crazy. The earth wire was sometimes folded back and sticking out of the AC plug so someone else (probably the Insurance investigator or the Police at that point) could tell that the gear wasn't earthed …!
I reckon that some of the signal noise problems could result from balanced gear (either 2 or 3 hot) connected to unbalanced gear in the signal chain at the same time. I guess, in theory, if everything was routed through the SSL patchbay, one could disconnect the signal grounds at the other ends (except for the mics) for balanced gear if there were any grounding (not earthing) issues assuming everything was on the same electrical circuit and not shared with items such as fridges for instance.
Also, especially with more digital stuff being used within a studio setup, keep the analog cables away from all the others within the trunking areas if it is not too late already. I remember a Sony DAT machine frequency affecting audio cables back in 1987 at Steve Winwood's studio in Oxon, UK so, hum and noise problems may not just be due to lifting screens.
I am not a tech person and have only expressed my views after wiring numerous different studio types plus dealing with and observing these issues over some 30 years. I have constantly researched this subject on the internet and still don't have a definitive answer (there is also a lot of bullsxxt out there) even now especially as more digital, optical and Cat type gear is involved in studios which adds to the confusion as some of these at least, could affect the audio signal.
I hope this helps but this a huge and somewhat unknown subject even now as you can probably tell from what I have written unless, I am talking rubbish (trash) as well?
Regards for now from Jim Lassen (www.profcon.co.uk). Also on FACEBOOK (http://www.facebook.com/pages/ProfCon-wwwprofconcouk-Professional-Connections-Jim-Lassen/371183312969924?ref=hl/)
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all of what jim said is correct, and there are lots of different approaches when it comes to grounding.
i came to find it easiest and most effective (i do around 3 major installs a year) to star ground everything from the patchbay.
you need all your mic lines and mic pre inputs shielded. both to ensure phantom power is working properly, and (like jim said) not to accidentally kill people.
i do however lift the shields on all other outboard and tape send/returns. the benefit is that it safes time (only 16 wires per d-sub, and no shrink tubing on the shields!) and, while it in most cases doesn't make a difference, it avoids ground loops. it can really only make the whole system quieter.
the only times i had to reconnect the shield is with a few really old american pieces (UA175b for example). if there is a hum, it will not at all be subtle, so you know instantly what to reconnect. the other good reason to only bent the shield back, and not snip it off, is that you might want to use some of them for mic pre ins in the future.
on converter d-subs, i snip the shield off right away. never had any issues there, and it makes everything easier.
like i said, there are different approaches to this, but i'm very happy with the noise floor this one achieves.
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oh, one more thing - on unbalanced connections you can lift the shield too, but make sure cold is connected to ground. (which you should do either way, to not lose 6 dB from losing a leg :-)
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Thanks for backing me up Brewery (I am relieved that I wasn't talking rubbish!). A star system is my preferred choice as well really as it simplifies the wiring as you mentioned and references all the signal grounds back to the central patchbay on an SSL rather than some unknown grounding wiring route at the other ends. Also, some equipment may have a separate power supply which could mean it has no direct relation to a proper earth as it may be a low voltage feed on 2 wires (+ and 0V).
By using a proper console like an SSL and all analog signals going through the patch, the star grounding method is a sound choice (excuse the pun) assuming outboard and multitrack racks (everything really that goes back and forth to and from the SSL patch) are referenced back to the central star point perhaps even with the same length of cable meaning that, if the shortest length is 2m and the longest is 10m, make them all 10m. Then you know that they are all the same going back to the central grounding point. Also, use big wire, if you can afford it, as there is less resistance.
If you are not using a proper console, which is becoming more common these days, you may just have a powered 8 or 16 channel bucket, some convertors, a computer and no patchbay. That's when it gets a bit more difficult to work out as there may not be a central ground reference point and various items of equipment are interconnected between themselves.
Brewery's point about lifting the screen and connecting the cold to the ground contact on an unbalanced jack reminded me of an installation I was involved with over 20 years ago. By the way, don't do this as you will find out later. The tech at the time said he was going to wire all jack unbalanced connections to equipment with 'stereo jacks', which is the wrong expression in Pro-Audio installation terms as they are really balanced single circuit jacks even though they are sometimes used for stereo, unbalanced, circuits on a connector or even mono insert send and returns on smaller desks (notice I didn't say consoles!). I think headphones are the only professional use of balanced jacks being used for stereo purposes but may be wrong!
Anyway, back to the story. All unbalanced outboard with 1/4" type A jacks got wired with balanced jacks and plugged in (I did have my doubts at the time). By the way, don't get confused with the GPO type B jacks which do not make a proper contact ... SSL bought these once by mistake for the 621 Producers Panel and found out the hard way during testing but that's another story.
Again, back to the plot and finally the punchline. When it came to testing, some of the unbalanced outboard didn't work properly. The reason was that the units were fitted with unbalanced jacks so, with a fully wired balanced jack plugged in, the ring (cold) had nowhere to go (or be referenced to) as there was no ring cold contact wired within the unit so it floated until the problem was discovered and fixed, in this case by moving the cold ring wire to the screen connection by the embarrassed tech.
People, perhaps research your equipment a bit before wiring. Open the top, see the service manual and understand the wiring arrangement especially the relationship between the audio ground and AC earth. In 1999, at a Mutt Lange studio in Switzerland, we took apart about 100 pieces of outboard to fit new transformers or change the links as the stuff had to be converted to European AC voltages. At the same time, all the signal to AC grounding/earthing was looked at and documented. There were quite a few differences between the units and it was an eye opener for me.
Hope this proves useful and I have got my facts right ... gulp. Regards from Jim Lassen (www.profcon.co.uk).
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Quick question as I'm just about to start the DLs for my converters. I have already done the D-Subs for the converters and they're fully connected +, - and Ground at the D-Subs.
Can I just connected the + and - at the DLs (and not the - I guess it's called - "audio ground").
FYI, we're also just figuring out the electric installation in my room and we will do a star grounding system.
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oh, one more thing - on unbalanced connections you can lift the shield too, but make sure cold is connected to ground. (which you should do either way, to not lose 6 dB from losing a leg :-)
Are you talking ins or outs? The subject of connecting unbalanced signals to balanced is huge.
Short answer, if in doubt, use a transformer!
Long answer, definitely new topic!
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Edited quote from Marc. "As I'm just about to start the DLs for my converters, I have already done the D-Subs and they're fully connected +, - and Ground at the D-Subs. Can I just connect the + and - at the DLs and not the audio ground? We're also just figuring out the electric installation in my room and we will do a star grounding system".
Marc (and others). You studio chaps need to address your technical ground ideas first of all before thinking about lifting individual grounds at various ends later on. Your D subs are already wired with ground wires but you are just figuring out the electrical installation, and will do a star grounding system. Surely, this is the wrong way round. Sorry Marc (I love your SSL refurb but call me or Eric ... Brewery ... on this so one of us can help).
Some SSL's (or other consoles) in professional studios may have a proper plan for technical grounding worked out beforehand but, these days, more smaller SSL studios may be located within a house. Therefore, you have to deal with the normal house electrics as well.
Balancing transformers is an expensive option and not necessarily required (although I have just posted an advert in the buy/sell menu for Sowter 5069 1:1 transformers). The whole technical grounding issue is indeed a complex subject but, to answer part of your question in a sentence, wire the DL as a balanced circuit including the screen (audio ground) drain wire.
The passive wiring in a studio (patchbays and panels as an example) should always be fully wired. Only the other ends (inputs and/or outputs) may need to be addressed in terms of lifting audio grounds but, as Brewery stated, a star earth system is preferable.
Perhaps some of us can swop ideas regarding grounding and present some proposals on this site. Eric (Brewery) plus Andy (Sintech), Mata and others could help. I haven't got time to draw some diagrams on this for now but it should be fairly simple.
More details from me (and hopefully others) later. One of the reasons I haven't posted something on this subject yet is that, when I draw something on paper (not computer) and try to scan it via my IMac, the pdf comes out as 12mb or something like that rather than 70kb or so. If I can master that problem, I will post many more topics on the forum (please advise and contact me directly if possible to save cluttering up the site).
Regards from Jim Lassen (www.profcon.co.uk). Also on Facebook FACEBOOK (http://www.facebook.com/pages/ProfCon-wwwprofconcouk-Professional-Connections-Jim-Lassen/371183312969924?ref=hl/)
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Offtopic.
Jim, check what dpi resolution your scanning software is using. 300dpi and text/line art (black and white) settings is good enough for diagram pdf's. Hope this helps.
-Paavo
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I agree with Jim. I did not look at this thread for a while because 6K's were always yawnsville for me. Nuff respect because they are quite relevant today.
I am not familiar with the electrical codes in the UK or Europe. In the US I have not needed a "star ground" system since the early 90's. All that baloney with a third wiring infrastructure terminating at a bus bar under the console and alligator clip-leads going to floater outboard or synths in the CR became irrelevant when I started to specify "hospital grounding" for the electrical. That has a separate third wire to every outlet box for third pin "ground" running back to the circuit box to it's own isolated bus bar which connects to the neutral bus bar at the box and then to the ground bond.
This style of electrical wiring creates it's own "star ground" eliminating any extra wire connections between chassis or 0000 welding cables run to mother earth. All chassis are referenced to the electrical box bond to mother earth. With that electrical foundation, everything plugs three-wire to the electrics.
Then on to the audio wiring where I connect all shields at the patch bay connections, DL's here, and lift at the opposite end. As written previously, all microphone lines and utility tie lines get shields tied at both ends- more for P48 considerations than "death prevention". In the 80's I was a purist and would lift all shields at the "load side" of a run, but that would make a mess at the DL's under the bay and make outboard changes more difficult that necessary- sometimes KEY inputs were on the top row or reverb AUX outs on the bottom. Very 20th century and not much of an issue these days! There are not many studios left with 32 I/O of digital reverb or a couple of Kepex racks.
As mentioned sometimes amplifier runs or vintage outboard would want to be connected at both ends, but everything else multitrack tape, DAW, outboard (except mic pre inputs!), externals, etc. has the audio shield lifted at the equipment end not the console end.
Not mentioned yet is what to do once you have everything wired. You want to connect the room one system at a time. Start with the monitors DL and test all monitors for noise. Add multitrack outputs and listen (carefully!) to the noise floor. Continue step by step with the multi inputs, aux i/o, outboard, etc. one DL at a time listening to the noise floor at each step. It is easy to identify any problem equipment this way.
I think that anyone integrating an SSL into a studio is already starting with a HUGE advantage in regards to noise floor and "grounding". One thing I had to deal with during the transition from tape and consoles to "boxes" was the lack of heavy metal in the room. Besides the SSL overall quality, grounding is much easier with at least one huge chassis relative to everything else in the room. It creates a greater potential for the flow of noise to ground. The dynamics of grounding and system noise elimination are completely different when all of the chassis are relatively the same weight except maybe a large power amp.
Mike
PS: It's been years since I had a 4K bay out to tighten patchpoints, but I swear that all internal wiring shields are connected to bay sleeves in both integral and remote patchbay consoles. Maybe a lift at the inserts only if any. I always respected the tight wiring work and all that black shield sleeving. Big sleeve fan here.
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Thanks for the replies and great info and advice guys! This is awesome.
I have finally begun testing the outboard via send and returns and insert points, as well as using small faders as fx sends. Every XLR to DL loom (for the outboard) is all balanced and so far there havent been any hum or interference issues and the system seems to be very very quiet, so we are happy about that. Obviously I havent had the chance to test out many different "real world" scenarios as there are a million possible combinations but as we use the studio more im sure some "grounding" teething issues will arise, at that point I will address them.
I do have a question though regarding the group outs on the console. I had the console in mix mode today and was using the small fader on a channel as an fx send, (i pressed the output button next to the small fader and in the routing matrix i pressed number 4) I patched group out number 4 to the input of a reverb unit, then patched the reverb units returns to a spare channel.
I happened to notice that channel 4 on our AD converters was also lighting up, so this meant that not only was the signal coming out of the group out patch row but was also simultaneously feeding the multitrack sends row beneath it. Is this normal? I thought the group outs only feed the multitrack send patch row when the "direct" button is pressed.
thanks guys, and hope you are all well!
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Group Out=Multitrack Send. You can either deadpatch the multitrack in, set DAW meters to output or ignore.
Mike
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Group Out=Multitrack Send. You can either deadpatch the multitrack in, set DAW meters to output or ignore.
Mike
cool, thanks Mike. So, hypotheticaly speaking, what happens if you were using all of your group outs during a mix and wanted to patch the consoles programme/main out back into your DAW to record the 2 trk? Im assuming that patching the console's mix output into multitrack sends 1 and 2 will remove the group out signal thats already present there? Now that I think of it, same goes for using DAW effects during a mix right?
thanks
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It"s always good to have lots of converter channels. Few examples:
- Feed individual tracks into console (obvious!!)
- send to Reverbs/Delays in the DAW from Echo Send/Cue Send/Routing Matrix
(Think of the great Impulse Response Reverbs and how many o these units can be run on a 8-Core Mac)
- bring DAW FX-Returns up on the Small Faders (and/or Echo Returns in the Centre)
Reminds me that I'd love to have the mod on my desk that allows Echo Send 4 to be sent to the Routing Matrix. I think that was a feature of the very last G+ Classic Consoles. With this feature, using the routing matrix as FX Send doesnt use up the Small Fader.
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Group Out=Multitrack Send. You can either deadpatch the multitrack in, set DAW meters to output or ignore.
Mike
cool, thanks Mike. So, hypotheticaly speaking, what happens if you were using all of your group outs during a mix and wanted to patch the consoles programme/main out back into your DAW to record the 2 trk? Im assuming that patching the console's mix output into multitrack sends 1 and 2 will remove the group out signal thats already present there? Now that I think of it, same goes for using DAW effects during a mix right?
thanks
Yes. That is the magic of a patch bay.
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Hey guys,
Is it ok to send a signal from one channel to more than one group out via small fader? Is the routing matrix designed for this or is it only designed to send the signal to only one group out? yesterday I was fiddling around with the console and out of curiosity I wanted to see if the signal from a channel would come out of more than one group out, so I pressed output button on the small fader then pressed the first 5 buttons that channels routing matrix and the signal appeared on the first 5 group outs at the patchbay. Im not sure that i'd be routing this like this in a real life scenario and maybe there is a purpose for it that im not aware of but I just wanted to know if its a bad thing to do.
thanks,
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Go for it.
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Yes, very popular mix technique. Doing this all the time, like bringing up a Kick on 3 or 4 channels, compressing and EQing them differently and then finding a good balance.
In Mix Status, you bring the signals up on the VCA Fader by choosing Subgroup as input at the top of the channel module.
Hey guys,
Is it ok to send a signal from one channel to more than one group out via small fader? Is the routing matrix designed for this or is it only designed to send the signal to only one group out? yesterday I was fiddling around with the console and out of curiosity I wanted to see if the signal from a channel would come out of more than one group out, so I pressed output button on the small fader then pressed the first 5 buttons that channels routing matrix and the signal appeared on the first 5 group outs at the patchbay. Im not sure that i'd be routing this like this in a real life scenario and maybe there is a purpose for it that im not aware of but I just wanted to know if its a bad thing to do.
thanks,
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Hey guys,
Last night I was testing some of the outboard reverb units to make sure everything was hooked up correctly.
I had a mono drum track playing on one of the consoles channels and via aux send 1 I sent it to a lexicon 480L, (sent to the 480L's Left input only which drives its internal machine 1 and the first set of its stereo outs). I bought the 480L Left and Right returns up on 2 channels on the console. So I had the original on one fader and the effected signal on the other 2 faders.
I noticed that when I bought up the 480L return faders there was noticeable smearing/phasing happening, particularly with the hi hats and within a certain volume range, presumably towards matched level with the original. I set the 480L to DRY and did the following. I flipped the polarity on the fx return channels to see if the loom feeding the 480L was wired ass about but when I flipped it and got the levels matched the bottom end dropped a little which made me think that its not wired wrong. However, the severe "trough" you'd expect to hear with 2 almost identical signals reversed from one another didnt happen, it does happen but not like some of the other units in the same rack.
At this stage I thought it may be the arrival time difference from the cables going to and from the 480L so i decided to test another unit in the same rack (same loom, same distance which is around 10 meters total). I patched in a tube tech eq, set it to bypass and flipped its returns on the console, matched the levels, and almost all the signal was gone. Flipped back to normal it was reinforced, louder, etc. So the phase relationship between the original and the tube techs was much better than the phase relationship between the original and the 480L, even though its on the same loom and traveling the same distance.
You dont seem to notice it as much with a wetter reverb setting on the 480L, its a lot more obvious with shorter, drier, more room type settings but its still there and its not a good thing. even with its medium hall setting you could hear it and is unusable.
Next time I visit I will unplug the 480L's inputs and outputs and connect them together essentially just making the signal travel that loom path and see what happens.
Has anyone encountered this sort of thing before? Anyone got any suggestions on what may be happening?
thanks,
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Yes, very popular mix technique. Doing this all the time, like bringing up a Kick on 3 or 4 channels, compressing and EQing them differently and then finding a good balance.
In Mix Status, you bring the signals up on the VCA Fader by choosing Subgroup as input at the top of the channel module.
Hey guys,
Is it ok to send a signal from one channel to more than one group out via small fader? Is the routing matrix designed for this or is it only designed to send the signal to only one group out? yesterday I was fiddling around with the console and out of curiosity I wanted to see if the signal from a channel would come out of more than one group out, so I pressed output button on the small fader then pressed the first 5 buttons that channels routing matrix and the signal appeared on the first 5 group outs at the patchbay. Im not sure that i'd be routing this like this in a real life scenario and maybe there is a purpose for it that im not aware of but I just wanted to know if its a bad thing to do.
thanks,
Hey guys,
been away for a while, hope you are all well. Been busy as hell, its crazy right now.
I think I get what you mean here. So, I have a kick signal returning from DAW on channel 1 on console, I assign it to matrix number 1, 2 and 3, I patch in my compressors of choice to the 3 groups outs on patchbay, return the individual compressor outs to line inputs on channels 2, 3 and 4. Find a balance between all four VCA faders, then float all channels to one VCA group fader by hitting "subroup" on that channel and pressing the float button on the other four (as well as selecting the subgroup number in the matrix of each channel). My question is, if this is correct, the original kick channel number 1 is already utilizing its matrix and group out function, can you still float this channel to the VCA subgroup fader at the same time? Hope im making sense! Its late here right now!
thanks,
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always something I remember just after hitting send! 2 things actually!
1) In mix mode, do the solo and mutes button affect the signal being sent out of those same channels group outs via small fader?
Im thinking no because there already are a set of solo and cut buttons next to the small faders. But, the other day we were running a test session and we had a drummer playing going straight to DAW from outboard pres, returning to desk VCA channels and were sending him his monitor/headphone mix via the group outs of each channel, small faders were up and only the input buttons were engaged. We solo'd the kick while he was playing and he told us it also solo'd in his headphones, I was under the impression that you should be able to solo and mute the control room mix while not disturbing the artist during a performance, I like to do this a lot during tracking, especially drums and doubling guitars. Something we are doing wrong?
2) When using the small fader as a send fx fader, to send "pre" large fader do you simply press only the input button, and if you want it "post" large fader press only the output button. Im confused as to which scenario requires them to ever be both pressed an why?
well, thanks again guys, going to bed now, been a long week, sick as a dog and spent most of today in a hospital! booooo!!!!!
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Great questions that I've been asking myself many times as well... hopefully somebody can enlighten us!
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Yep, SOLOs and MUTEs will affect the signal going to the groups. Learnt this the hard way!!
Have both the INPUT and OUTPUT buttons pressed if you want a pre fader signal sent with signal processing ie EQ and DYNs.
Stef.
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Yep, SOLOs and MUTEs will affect the signal going to the groups. Learnt this the hard way!!
Have both the INPUT and OUTPUT buttons pressed if you want a pre fader signal sent with signal processing ie EQ and DYNs.
Stef.
oh that sucks! I guess I can just pull down the large faders but thats gonna get annoying real fast, I guess there are gonna be some compromises when using the console in a hybrid sort of mode/setup, is to be expected.
RE small faders: Ok, cool, so it doesnt matter what combination of "input/output" buttons are selected for the small fader, the signal will always be pre fader but the inclusion of eq and dynamics is altered, got it. Thanks
Stef, did you have any insight on the question in the previous post about using a channels routing matrix to send its signal to the group out AS WELL AS assign it to a created VCA subgroup?
thanks again!
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I'm a little confused by all of that..
"VCA Subgroup" - are you referring to a VCA group or a physical/audio subgroup/buss?
Stef.
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I'm a little confused by all of that..
"VCA Subgroup" - are you referring to a VCA group or a physical/audio subgroup/buss?
Stef.
Yeah, sorry, im talking about an audio channel that has its "subroup" button pressed (not the groups near the centre section)