Interconnect brainfart...

[Brad1]

OK...I should KNOW this, but I'm setting things up differently, and can't wrap my head around this. I've used this mixer in a dozen different ways, but never tried this.

Hooking up a Mackie 1604 (NOT VLZ) "Channel Access" to a TRS patchbay, and out of that into an M-Audio Delta 1010 interface.

They are TRS jacks, but they are not balanced, as I had assumed until I reviewed the manual. It says to "Insert TS plug to first click for channel direct out, not interrupting main". The manual also states that inserting a TS plug in completely will provide a direct out, which does interrupt the main. That's OK, and that's what I want. (A TRS cable would be used for a send/return from effects. Understandable...no problem...not using that for this.)

Here's my confusion. It's an unbalanced signal going out to a TRS patchbay. The patchbay outs are TRS cables going into balanced TRS Delta 1010 inputs, set to +4. (The Delta will automatically accept unbalanced TS).

Can I just leave it as is, or do I need to crawl in back and change all the TRS cables, from the patchbay to the Delta, to TS?

In other words, is the entire chain automatically unbalanced just be using TS from mixer to patchbay, or will the Delta be seeing a 1/2 balanced, or otherwise odd, signal?

Also, might I need to change the Delta input levels to -10, from +4?

I've been mulling this over all day, and can't think it out completely. The equipment is over at our rehearsal space, so I can't experiment until Saturday. That's actually the last step of connections before setting mixer, then software, levels and hitting record. Hopefully, someone can save me some time, and say "Use it as it is", or at least make it just a 5 minute cable swap.

Thanks,

Brad1

[Steve Conner]

Hey Brad1

I used to use a similar setup, with a Soundcraft Spirit M12 hooked up to a Delta 1010. The Spirit had balanced direct outs, so I just used TRS leads.

It sounds like those "Channel Access" sockets are just inserts. So they're unbalanced, the tip is the send and the ring is the return. So ideally, you want a bunch of special leads, with 2-conductor screened cable, wired as follows:

Tip at mixer end to tip at Delta end
Sleeve at mixer end to ring at Delta end
(Nothing at mixer end) to sleeve at Delta end, but cable screen connects to it

If you just swap TRS for TS at some stage, it'll short the sleeve to the ring, and that can potentially create a ground loop. The hookup I suggested above uses the Delta's balanced inputs to break the ground loop.

If you use TRS all the way, you'll have one leg of the Delta's balanced input connected to the mixer's return. So, two inputs connected together. That might not do anything bad, or it might be noisy.

Your mixer manual should tell you if the Channel Access ports are +4 or -10. I'd guess +4.

[Brad1]

Steve,

Thanks. The manual doesn't state -10/+4 from those, but that's no problem. I'll just set it for the best level/least noise.

http://www.mackie.com/pdf/archive/cr1604_om.pdf

pp. 13, 15-16

I think the Mackie diagram (page 16) is confusing me. On the "Insert mono plug to first click" example, the switched TRS jack is closed, so the tip of the plug is getting signal from the ring lug, because it's shorted to the tip lug...and the ground (sleeve) is solely from the TRS jack sleeve connection.

On the "insert mono plug completely" example, the switched TRS jack T and S connections are now opened. The tip of the plug is getting signal from the T of the jack, and the the sleeve of the plug is connected to both the R and the S of the jack.

Now, I'm trying to imagine this as the inputs to the Delta. I'm...confused. I'm thinking it may work better to go ahead and swap out all the TRS plugs from the patchbay to the Delta with TS? That should pretty much render the TRS patchbay a TS patchbay, since all ring and sleeve connections are now connected by all the plugs sleeves? The Mackie will be outputting an unbalanced TS signal, the patchbay will be passing an unbalanced TS signal (with all Sleeves and Rings shorted), and the Delta will be seeing a TS signal with a simple TS plug. Basically like just removing the ring connectors from the equation?

That's going to suck, because I had 4 other channels of better quality outboard preamps to use after, and I wanted to simply remove channels 1-4 inputs of the patchbay, and run balanced XLR-TRS cables from the preamps to the Delta. Guess I'll be swapping four cables again when that time comes.

I think I have this straight. Any other thoughts, suggestions, etc?

Thanks,

Brad1

[Steve Conner]

That's going to suck, because I had 4 other channels of better quality outboard preamps to use after, and I wanted to simply remove channels 1-4 inputs of the patchbay, and run balanced XLR-TRS cables from the preamps to the Delta. Guess I'll be swapping four cables again when that time comes.

Like I said, make yourself some custom TS-to-TRS cables that go from the desk to the patchbay. Then your patchbay will still be balanced and you can patch balanced things into it.

It should also work if you just replace the cables from desk to patchbay for TS-TS ones, but you may get some hum from ground loops. In either case the cables from patchbay to Delta should stay TRS so that you can patch balanced sources in.

[Steve A.]

That's going to suck, because I had 4 other channels of better quality outboard preamps to use after, and I wanted to simply remove channels 1-4 inputs of the patchbay, and run balanced XLR-TRS cables from the preamps to the Delta. Guess I'll be swapping four cables again when that time comes.

Brad:

I have the original 1202 and they call these TRS jacks "Channel Insert" which I guess is less confusing. Although the 1604 manual labels them "Channel Access Tip=Out Ring=In".

Regardless of the name the block diagram for the mixers (see your page 24) shows that the Channel Access jacks come after the mic preamp w/ gain trim control, 3 band equalizer and fader so I'm sure you weren't planning to run the output from the Channel Access jack to the higher quality external microphone preamp.

I believe that you will get the best results plugging your balanced microphone directly into the external preamp. I have a half dozen jack bays but I have only used them for line level signals. If you wanted to keep the mics plugged into the patchbay you might want to look into the patchbay manual; especially on the newer ones there are a lot of tricks you can do with them depending on how you configure the jack normalling (not to be confused with Jack Normalling, Chicago Cubs utility infielder, 1952-1961, .261 LBA).

On page 29 of your manual it shows you how to modify the channel access points to pre-fader pre-eq or just pre-fader. But they are always after the mic preamp.

These mixers were very innovative when they came out. Reasonably priced for the home studio, with decent (if not exceptional mic preamps), a 3 band EQ that sounded musical not sterile, and perhaps the easiest user interface in terms of setting levels. Plus the gain controls/faders which had 0dB (Unity) right in the middle with oo at the lowest setting and +20dB at the highest. Perfect for home studios.

Always the mad scientist I added channel inserts on the two output channels. While I have a drill and some jacks- isn't that what they are for?

Steve Ahola

P.S. Be sure to listen to Steve Conner- he knows more about this stuff than the rest of us combined! As he surmised the output level from the channel access output is +4dB.

[Brad1]

Regardless of the name the block diagram for the mixers (see your page 24) shows that the Channel Access jacks come after the mic preamp w/ gain trim control, 3 band equalizer and fader so I'm sure you weren't planning to run the output from the Channel Access jack to the higher quality external microphone preamp.

I believe that you will get the best results plugging your balanced microphone directly into the external preamp. On page 29 of your manual it shows you how to modify the channel access points to pre-fader pre-eq or just pre-fader. But they are always after the mic preamp.

Steve Ahola

Steve,

Thanks. Yeah, I realize they post EQ/Fader. That's OK. We'll be recording 3 mic'ed guitar cabs, a direct bass, and 4 drum mics. Then, we'll record vocals later. Post EQ-fader will be ok, because though we are RECORDING bass guitar direct, the bass amp will also be used. That means the bass is likely to get into, especially, drum mics. I'm going to TRY to EQ out anything below what any of the instruments needs to lessen the bass picked up through other stuff. I'm sure it may be less of an issue with guitars, since they'll be so close to the guitar amp speakers.

We're basically recording a live performance, and then adding vocals. So, the direct-recorded bass track, itself, will be the cleanest signal of all them, ironically, as all the other tracks will have bleed...especially bass bleed. I've already mentioned that we may need to turn ALL our amps down a bit, but I'm fairly sure I'll have to actually record the first tune, then PROVE to them why I say that. We need to hear the bass while playing. We'll see what happens. If nothing else, and it get's TOO much in the drum mics to make mixing the drums adequately impossible, I've already warned the drummer that we MAY need him to redo the drums...alone...while listening to the recording on headphones.

Anyway, I sat down and drew out my confusion of balanced-to-unbalanced, which was my main concern. I'm sure I can just leave the TRS cables.

Since the Delta1010 will accept unbalanced TS, and the Mackie outputs unbalanced, as long as I have the Mackie out to the patchbay TS, everything downstream into the Delta will be acting as a TS. It'll be just like going straight to the Delta from the Mackie with TS plugs.

The input to the patchbay has shorted the ring and sleeve, with the TS plug. So, the TRS plug out of that has been turned into a TS, and on through. All the rings downstream of the patchbay input are shorted just because the first input is. It just looks like a regular 1/4" mono plug to the Delta. Everything's good...I think.

Thanks,

Brad1

[Steve A.]

Steve,

Thanks. Yeah, I realize they post EQ/Fader. That's OK. We'll be recording 3 mic'ed guitar cabs, a direct bass, and 4 drum mics. Then, we'll record vocals later. Post EQ-fader will be ok, because though we are RECORDING bass guitar direct, the bass amp will also be used. That means the bass is likely to get into, especially, drum mics. I'm going to TRY to EQ out anything below what any of the instruments needs to lessen the bass picked up through other stuff. I'm sure it may be less of an issue with guitars, since they'll be so close to the guitar amp speakers...

Brad:

There are digital tricks you can do that I believe would be better than trying to EQ out the bass bleedthrough. Because the bass is recorded direct you can subtract a portion of that signal from the instruments that have the most bleedthrough.

Back in the 70's there was an interview on KPFA with the soundman from the Grateful Dead (who pretty much invented the idea of touring with your own top notch PA). The soundman was saying that the crew would spend many hours setting up the microphones so that a minimum amount of EQ was needed. He explained that EQ basically puts some of the frequencies out of phase with the others. BTW phasing is not just "in" or "out but covers a full 180 degrees; some of the new software takes this into account (the old mixers would often have a phase toggle switch which was better than nothing).

You may need to make a choice between bass bleedthrough and a loss of some fidelity due to EQ and other digital tricks. Complete separation of the tracks allows for maximum flexibility in mixdown but I believe that there are other factors to consider.

Will there be someone there operating the mixer, or are you just going to hit "Record" and hope for the best figuring that you can fix it in post-production.

Live multitrack recordings are cool but a lot has been done just mixing them down to two channels for the PA mains or a radio broadcast. Or have the person running the mixer go for the best sound from the stereo output; that would help determine how well the 8 individual tracks are working together and leave less to deal with in post-production.

Just some ideas and I'm sure that it is nothing new to you.

Good luck!

Steve Ahola

[Brad1]

Steve,

Everything worked well. I did have to switch the inputs on the Delta to -10 to get a healthy level to the DAW. What I did was set the Mackie channel and main faders to "U". All EQ knobs centered. Then I set levels on each instrument channel input trim to just hit the yellow "pre-clip' LED on the Mackie, when, say, a drum is pummeled as hard as possible.

Then, I set all Trim for each channel in Sonar to "0", then I adjusted the Delta control panel channel input faders while instruments were being pounded and blasted at the volumes were used.

All channels, except the kick drum, were left with EQ flat on Mackie. I tweaked that a bit to get a bit more low, and less mid, and get the beater "snap".

After recording a couple, while doing a quick levels mix upon playback...sounded pretty good through the PA, and headphones. I think I can work with it after I bring it home to the KRK monitors.

The MOST surprising thing was...NONE of the other tracks picked up hardly ANY bass. I would have sworn that the two drum overheads, at least, would have been picked up a bit. The drums are in a corner, the bass cab about 5' along the wall to the right, pointing out straight, and the three guitar amps were to the left of the drummer, and about 4' separated from each other, kind of in a mild curve to where the farthest one was near the other wall, and slightly pointing back across. All open back cabs. We weren't playing real loud, but we weren't playing real LOW, either.

I was amazed at the minimal bleed of each track, as we solo'ed them. I must have set something up right! I can work with these. Denoise each track, some mild compression and EQ. I had everyone turn down their amp reverbs, and we used no delays (which was difficult for me, because as the "texture guy", I use delays often, while the main lead guy doesn't, and the main rhythm never does). Figure I can add a bit of 'verb and any delays later, and it'll make it easier for me to clean up the tracks, first, before I do anything else to them.

Next up...vocals. We got through the instrument recordings of 10 tunes in 4 hours. Three, we had to do twice. None had to be done more than twice, though there may be a couple things some of us might want to redo, individually, if I can't make it disappear, or minimize the clam. Probably could have done it in less time, had I not insisted on backing up the BUN. files to an external drive.

Anyway, thanks for the help, guys. My main confusion was the unbalanced-to-balanced thing, and that worked out fine once I sat down and drew it out. The Mackie to patchbay is an unbalanced TS 8 channel snake, 4' long. The other TRS cables from PB to Delta are buried in the back of a deep case, against a wall, that also houses the rack-mount computer, and Delta interface. I was just hoping to avoid digging around inside that. And, I did.

May get a nice little demo out of this! Much better than the previous attempts of mixing straight down to stereo HiFi VCR while rehearsing, and someone inevitably imbalancing the whole thing by turning up or down. Not like I could grab a fader. My hands are busy with the guitar. Those previous attempts DID have some use, though. I was able to convince them of levels management, the wisdom of tuning up properly...and why noisy-ass, hissing, buzzing Little Big Muffs suck for recording.

Brad1

[Steve A.]

Everything worked well. I did have to switch the inputs on the Delta to -10 to get a healthy level to the DAW. What I did was set the Mackie channel and main faders to "U". All EQ knobs centered. Then I set levels on each instrument channel input trim to just hit the yellow "pre-clip' LED on the Mackie, when, say, a drum is pummeled as hard as possible.

Great to hear that your recording turned out well- it sounds like you certainly know what you are doing!

I think that the last thing I would want to hear in a studio is "don't worry, we can fix that in post-production... with EQ!"

A little bit on the drums is fine, but some people will tweak the EQ on every single track and by the time they are done they have lost some of the phase coherence resulting in a muddy sound. Each channel might sound great by itself but mixed together it just doesn't blend right... Of course I am preaching to the choir when I tell you that- I mention it only for the benefit of those who think that equalization is next to godliness...

Steve Ahola

P.S. I don't know what you are using to denoise the tracks, but with Adobe Audition 3.0 you have control over all sorts of parameters (like the FFT size for the noise sample). The control I use the most is like a blend control going from 0% to 100% NR. I use it mainly for 24 bit vinyl rips and I usually set it somewhere between 40 and 75. Too high and you start getting a hollow unnatural sound and setting it too low will leave a lot of the noise in. So I usually come up with a compromise between those two extremes that sounds best to my ears. In other words leaving a little bit of noise in there isn't all bad.

Another point: when the band is playing at the higher volumes it will usually mask the noise, which you usually only hear in the quieter interludes. So I might denoise just the quiet stuff and leave it off when the band cranks it up a bit. And sometimes I will even fade the NR in and out, using an intermediate value in the middle so that there is not a sudden jump or drop in noise level. But like I said I do all of that mainly for vinyl rips and also for doing digital remasters of 30-40 year old recordings being exchanged among the traders.

I attached a screen dump of the Audition Noise Reduction window. I'm sure that other companies have come up with better algorithms but I point this out because I'd expect any NR program to have these controls or even more. "Low" "Med" "H" just doesn't work for me.