Something i find curious about LIVE...?

djritchie · Post by **djritchie** » Mon Jan 07, 2008 2:12 pm

Say for example, i have 1 audio channel with my kick drum on it. And then i put a Waves C1 compressor (or the like) on it.

There seems to be no way i can reduce the amount of input going INTO the compressor? If i pull the channel fader down the LED going into the compressor remains the same, so i can change the audible level but not the actual input signal?

It means that in some cases i get to much input into my comprssor meaning its already in the red. In FL Studio i cud just pull down the channel fader and it would correspond to the input level going to the compressor - You get me?

elektrovert · Post by **elektrovert** » Mon Jan 07, 2008 2:18 pm

put utility in the signal path first maybe?

djritchie · Post by **djritchie** » Mon Jan 07, 2008 2:23 pm

yes can do that but then the input into the utility plugin is still proportionate to the original channel volume (not adjustable into the utility plugin via channel fader) - so if it was peaking it wud still peak into the utility plugin but yes it cud be turned down. still not right tho!?

andydes · Post by **andydes** » Mon Jan 07, 2008 2:38 pm

Turn down the gain slightly in the clip properties box, perhaps?

djritchie · Post by **djritchie** » Mon Jan 07, 2008 2:40 pm

yeh i can do that too , but then i have to alter every sample etc - all work arounds - just wondered if there was an actual reason or way to do this proper - its annoying as i do my mixing as i go.

zsazsa · Post by **zsazsa** » Mon Jan 07, 2008 2:49 pm

isn't that why sends are invented?

Angstrom · Post by **Angstrom** » Mon Jan 07, 2008 2:51 pm

djritchie wrote:yes can do that but then the input into the utility plugin is still proportionate to the original channel volume (not adjustable into the utility plugin via channel fader) - so if it was peaking it wud still peak into the utility plugin but yes it cud be turned down. still not right tho!?

'peaking' in the realm of 32 bit, or 64 bit audio does not mean distortion. you could send that signal waaaay out of the top of the input meter as long as you then brought it back down again prior to the master stage and file writing.

you can test this out by getting 5 utilities in a row and turn them all up to maximum, then add a few more utilities and bring the signal back down - you will notice that the signal is not distorted. This is because the floating point audio engine has a huge headroom.

don't worry about clipping inside the app, just make sure you bring it all down below 0db before the output stage.

djritchie · Post by **djritchie** » Mon Jan 07, 2008 3:04 pm

thanks for the advice mate!