effects loop - really?

deadfoot

New member
I have a Tascam M224 mixer and have 2 questions about effects loops:

1-I think I know how to hook up my compressor or echo unit into the effects loop, but I want to make sure:

send of effects loop to input of compressor
output of compressor to return of effects loop

Is it really that easy? Are there any drawbacks doing this when recording? Can I chain my echo unit in there too?

2-The M224 has an FRL send and return, which looks like another effects loop (maybe the "L" in FRL stands for loop?) I don't have a manual for the board yet. Is it simply another effects loop?

Thanks!
 
Compressors are not ususally hooked up to sends. They are connected to channel inserts. Effects units are used with the sends though and hooked up as you described.
 
Thanks HawgDawg - Do you know the technical reasons why I couldn't use the compressor through the effects loop? It seems to work okay and is certainly a lot cheaper than buying a compressor for every insert.

By the way, for anyone interested, the FRL send/return seems to be another effects loop that additionally can be switched between intrument and tape input (for the first 16 tracks anyway).
 
You could use it on an Aux, but I wouldn't suggest using it on more than one channel of the mixer and I would not use the return. You would use just the output of the compressor to send to the soundcard. An Aux channel effects loop basically just blends the effected sound in with the dry sound, unless your mixer offers control over wet/dry signals.
Why not use it on multiple channels via an Aux?
Because an effect such as a compressor/limiter acts on a single signal in regard to the attack, release and threshold you set. If you are sending a signal from one channel, the compressor will be acting on that signal alone within the parameters you set. Now all of the sudden you've got a signal coming from one of the other channels. The compressor may not even be done with the first signal (attack, release and threshold), so the second signal gets no compression.
If you require compression/limiting on all channels ... you need to either ...
A: have a compressor for each channel .... or
B: have a compressor on the master channel and record the sub-mix.

Use your Aux send loops for color effects such as delay and reverb.

HTH
 
In addition to the reasons above, you need the entire signal to go through the compressor.

A channel insert sends the entire signal to a device, the signal is processed, and then sent back to the board via the same channel. An aux send sends however much of the signal you wish to the device and then sends it back to the board usually through an aux return channel; so in the case of a compressor, you would essentially need to set the aux send to 100%, so all of the signal is compressed, and set the aux return to a suitable level. You also now have the dry signal to deal with (the signal without compression), that you would need to cut out of the mix (which you most likely will want to).

In short: use an insert. :D


PS: You'll need an insert cable. This is a Y-cable that is TRS ("stereo") on the single end with two TS ("mono") ends. Looks like this: http://www.hosatech.com/hosa/images/STP-200.gif
 
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Block diagram wanted

Hmmm does it really say "FRL"? Mine says FLB! And as far as I know it stands for Foldback. I think it was designed (but I also don't have the manual
and/ or the signalflow diagram) for a feed to the headphones in a studio situation and a monitormix on stage in a live situation. The Aux was made for auxilaries. Nevertheless the double function of this FLB is also in the deal. The FLB can be used as a prefader (I assume) FXsend. The tapebutton I think is to let the musician to the mix that is recorded (on "tape") or to his own voice or instrument he is playing directly. The purpose of course is to make multitrack recordings. For live as well as studio a premix is nice, since in live situations you would like to have a different mix in your stage monitormix from that one of the F.O.H. speakers. For studio it is nice because you can choose where the musician listens to, the music on tape and/or her or his instrument of voice that is being recorded. By giving the musician his/her own mix, you can still work with those items after the pre-Aux.

And about the compressor I would suggest: buy one or two patchbays. That way, you can connect freely in any insert (in turns of course). I would not use a processor on an aux. You can connect it for the final compression on the stereobuss, to any main or subgroup output and route it back into a channel or a recording device. Again, Patchbays will make this all switchable, so more flexible.

MOM

PS. Anyone knows someone with a signalflow/block-diagram of this mixer?
 
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