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Old 8th October 2008   #1
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Using external effects and Logic... HELP!

Hi there!

So, I've just got my first external stereo compressor, the FMR RNC, which I got really cheap on eBay. Now, I've connected it to my Ensemble unit and I've successfully sent Superior Drummer to the outputs on the Ensemble, over to the RNC, and then back again to the inputs on the Ensemble, and the compressor sounds great. Yay!

But! If I raise the latency, just for testing, to let's say 1024 samples, it doesn't sync at all. It's like having loads of UAD plugins but without the latency compensation.

The latency compensation in Logic is set to "All", which means both audio and plugins.

What am I missing here? Anyone?
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Old 8th October 2008   #2
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Unfortunately Logic doesn't compensate for hardware inserts automatically.

If your mixing entirely in the computer I'd suggest you try it as an insert on the master buss ( out 1 and 2 ) that way any delay induced is on every track and you benefit from the analogue goodness on the whole mix.

You can use it on the drum buss as you intend but it's a lot more work, you'll need a plug-in like

Expert Sleepers - Latency Fixer

And then you'll have to spend some time setting it up. There's a good discussion on it here

http://www.gearslutz.com/board/high-...-ensemble.html

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Old 8th October 2008   #3
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With the I/O, there is this bundle Artificial Audio Pluginmenu
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Old 8th October 2008   #4
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Inserting an analogue hardware device like a compressor is exactly the same as inserting a singer. That is, it is just as possible to get a singer's singing to be properly lined up with the cue mix, as it is to get a compressor's output lined up with the audio that you're sending it to it.

Make sure that software monitoring is turned off, and that latency compensation is set to All. With SW monitoring off, the latency (buffer size) is completely immaterial -- it is inaudible.

This is very often presented as a problem of automatic latency compensation. It is not that at all. It is a simple matter of aligning the audio coming out of the audio interface with the audio going into the audio interface. And this should be possible, down to the sample, regardless of any buffer settings, unless you are trying to monitor the inbound audio through the DAW as well.

There is much confusion about this topic on the posts in this board ...

-synthoid
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