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gain setting in ableton's external audio effect plugin


d-a-m-o

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sorry this is a total noob question :

 

I want to set up an hardware delay on a return track in ableton.

everything is set but I got a question regarding the input and output gain of the plugin and peak level :

 

what is an acceptable peak level for the in and for the out ?

 

 

 

 

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I tend to aim for 'keep in the greens and occasionally hit the yellows' - That way you have enough headroom to play with stuff later down the chain. Think that's generally around the -12db mark....

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No. The manual addresses your question only by saying to avoid clipping. There's too many factors related to each individual's hardware and software involved to have a 'standard peak level' beyond don't clip: mcbpete's response is your best guide imo.

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Go 24bit, get 144dB of dynamic range and try to have all your peaks under -16dB FS. I’ve read countless times from quite reliable source that it’s most AD/DAs analogue path sweetspots : the more headroom the better, and S/N ratio is a non-issue when recording and mixing in 24bit. Use a gain plugin afterwards if needed.

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Absolutely, though (when thinking about this last week) there's obviously a s/n concern if using an external hardware delay unit - If it's a new fangled digital delay you'll probably be fine at this level but if it's like an old 1st gen Roland RE-201 you'll probably want to be feeding a considerably louder signal than -16db into it to minimise the impact of the noise floor.

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It's an OTO machine BIM, I set the external audio effect plugin on a return channel, dry/wet of the plugin is full wet, dry/wet on the BIM is full wet and I use the send of the instrument  channel to blend dry and wet signal. -12db was ok, no noise in the background but I think my question was more like : what is the correct ratio between out volume and in volume for a hardware delay ? Also I read that when boosting the input volume it's better to do it in the audio interface soft rather than in the external audio effect plugin, is that true, I'm a bit lost

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You can try to run a sine wave in and out your soundcard (a DA/DA loop) and check what’s closer to the original signal, either running phase cancellation tests or using something like Voxengo Span. I’d happily give you a more informative / useful answer if I could.

Oh and try to add input gain directly on the OTO, depending in the signal path it might add a quite nice little something to the source Is it possible to bypass the delay (but not the conveters and analogue stage)? I believe it’s quite a tasty piece of gear you have here

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Ps: I meant DA/AD. I haven’t run these sorts of tests for years as am 100% ITB now, but they can be quite insightful!

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You mean bypass the delay and keep the 12 bit character ? I've read somewhere that it is possible and Denis is working on a new update with a dedicated mode for this purpose ! Lovely piece of gear for sure ! Thanks for the tips Nil, will try it ASAP !

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