Using a hardware compressor during mixing

Xcaliber

Xcaliber

New member
Ok, let me say I'm a total newb when it comes to using hardware processors of any kind so I might be totally off base with this, but I need help. I recently bought 2 channel hardware compressor and I want to use it to add compression to tracks while mixing (not while recording)...mostly because the tracks I'm mixing were not recorded by me. I know some people are going to say "just use a software compressor in the box and be done with it", but I really want to try the hardware compressor even if for no other reason than do learn how to do it. I'm also planning to use a patchbay (again, because I can and want to learn) so that is in the "mix" here as well.

So here's what I'm thinking (based on something I read on the Internet):

Run two of the balanced outputs of my TASCAM US-1800 to two inputs on the patchbay and set those inputs to "thru". Run the same two outputs from the patchbay to the inputs of channel 1 on the compressor. Run the outputs from the compressor to another two patchbay inputs and then run the same outputs from the patchbay to two inputs on my TASCAM. Then I can route the track in Studio One to the outputs that go to the compressor and create a new track that uses the inputs from the compressor so I can have a compressed version and the original uncompressed version in the DAW.

I'm probably making this more complicated than I need to and I appreciate any advice/help you can give me.

Now that I read that, will it even work? I'm so confused.
 
That should work, though unless you have a bunch of compressors and need to reassign them to various outputs and inputs on the Tascam I would probably just skip the patch bay for simplicity's sake. There's a whole range of different ways to integrate the patch bay, but none of them really help you much in this case.

The way I'm using patch bays right now is to have one that's just for the compressors and other outboard, one that brings the console inserts to the rack and a third that brings the interface inputs to the rack. I record from the console's insert sends and I can split the signal off to the interfaces before or after the compressor. I can patch any compressor to any channel and route any channel on the board to any input on the interface (though usually Ch 1 of the board just goes to Ch 1 of the interface etc.).
 
Be aware of shifts in alignment. Some apps have features that presumably compensate for the round trip delay.
 
mixsit said:
Be aware of shifts in alignment. Some apps have features that presumably compensate for the round trip delay.
Click to expand...

I'm pretty sure if it records live sources in sync the round trip will be just as in sync. Unless the outboard has its own latency.
 
I do this with my own compressor. I have a tiny bit of latency but it's ever so slight. Just a small nudge back to the original spot and I'm good to go. Nothing to really worry about.
 
bouldersoundguy said:
I'm pretty sure if it records live sources in sync the round trip will be just as in sync. Unless the outboard has its own latency.
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I only used it outboard FX (Sonar) and it definitely got a bit more delay in the recorded FX track than what I heard in the live monitor mix - tracks/mix and the FX going through a mixer. When it was recorded and played back, it was like a bit of pre-delay being added.
I was not however doing Sonar's 'hardware insert compensation' - 'ping and set' routine.
 
So looking at things this evening I realized that my compressor only has one 1/4" input and one XLR input as well as one 1/4" output and one XLR output. Can I achieve the setup I described above by with just one output and one input on the TASCAM?
 
Xcaliber said:
So looking at things this evening I realized that my compressor only has one 1/4" input and one XLR input as well as one 1/4" output and one XLR output. Can I achieve the setup I described above by with just one output and one input on the TASCAM?
Click to expand...

You won't want to compress anything that's stereo, but you can compress mono tracks.
 
mixsit said:
I only used it outboard FX (Sonar) and it definitely got a bit more delay in the recorded FX track than what I heard in the live monitor mix - tracks/mix and the FX going through a mixer. When it was recorded and played back, it was like a bit of pre-delay being added.
I was not however doing Sonar's 'hardware insert compensation' - 'ping and set' routine.
Click to expand...

Why would there be a difference between a live source and looping the signal back? Actually, how could there possibly be a difference in timing between a person keeping steady time with playback and the playback being in time with itself? And if one is off the other is inherently off by the same amount for the same reason.
 
bouldersoundguy said:
Why would there be a difference between a live source and looping the signal back? Actually, how could there possibly be a difference in timing between a person keeping steady time with playback and the playback being in time with itself? And if one is off the other is inherently off by the same amount for the same reason.
Click to expand...
You got me there, I can't explain it (maybe I'm missing something or have it wrong.. hmm.
Why do they have (or need) specific 'external hardware insert routines in the app, but don't need it for over dubs?
 
mixsit said:
You got me there, I can't explain it (maybe I'm missing something or have it wrong.. hmm.
Why do they have (or need) specific 'external hardware insert routines in the app, but don't need it for over dubs?
Click to expand...

Other than when the outboard processor itself introduces delay, I don't know. Maybe it's just more obvious when you loop something back so that's when people make the effort to align it better.
 
So, I heard the feedback on using the patchbay, but I have some other equipment I want to introduce as well (a rack mixer for one) and I also just want to learn how to use the patchbay for my own knowledge (I love playing with and learning how to hook up/use gear). I understand it's not necessary for me at this point though.

Also I have an (un)natural love for wires and hooking things up. :)

That being said is "thru" the right configuration for what I described in my original post? It seems like when the connection is not "patched" I would want the signal to be "broken" which is my understanding of when I would want to use "thru".
 
I don't know what the maker of your patch bay means by thru. The standard terms are normaled or non-normaled. You don't need it to be normaled, but it doesn't really matter for what you're doing. With the normaled configuration it may be possible to have a feedback loop, but you'd have to close the loop in the DAW and that's not likely.
 
In your case, you would set the whole patchbay up as non-normalled. This means that the jack in the back connects to the jack in front if it and nothing else. A normalled patchbay is used when you are hooking inserts up to it, and need the patchbay to loop back when you aren't actually patching something into the channel. Since your tascam doesnt have inserts, a normalled patchbay wont do anything useful.

Simply connect all the inputs and outputs of everything relevant to the back of the non normalled patchbay. Outputs on top, inputs on bottom. (or the other way around, just be consistent). Now you can use a patch cable in the front to patch any output to any input.

You obviously have a mono compressor. Just use either the 1/4 or the XLR ins and outs, not both. It shouldn't matter which you choose.
 
Thanks! I will use non-normalized on the patchbay, that makes sense to me. I had the "thru" setting wrong, it's for hooking up CD players (etc.) and not used for the application I am talking about.
 
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