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ghast

Posted on Saturday, August 06, 2005 - 02:07 am:

Hi

Does anyone know a way to bring up quiet portions of a WAV file automatically, without selecting individual portions of the file graphically and then transforming them? I need to do it because I'm using homemade cutoff filters for high and low frequencies. Naturally the volume drops off sharply whenever I turn a filter on. I'm using Audacity to manually normalise volume levels in the music, but it's not practical to do it this way because of the short bursts of quiet sound. So what I'm after is either a really clever look-ahead compressor which has the muscle and the brains to maximise the use of bandwidth in short bursts of quiet sound (most limiters can't do it assertively enough), or else a computer application that will run through a WAV file and make calculations at intervals to normalise volume levels. Any ideas? Thanks

Neil C

Posted on Saturday, August 06, 2005 - 05:17 pm:

Well, in Cubase you could just select a section of the finished track (as big or small as you like) and gain adjust it. Or do it with automation.

Why do you need to be turning the filters on and off (and what are you using the filters for)?

Any look-ahead compressor could squash the louder bits to bring them in line with the quiet bits but you shouldn't be using a compressor for corrective surgery when you have control over the original mix.
Or is this a live thing?

ghast

Posted on Sunday, August 07, 2005 - 09:18 pm:

Hi Neil. Yes, that's how I've been doing it "manually" (except with Audacity not Cubase). It works perfectly, only it takes a long time because you need to really zoom in to select the right portion of the file, otherwise it will sound chopply.

A limiter would be able to do this live and in real time while the piece is playing - obviously using Audacity the way I described is not a live method. What I have been doing is using a real-time studio application similar to Cubase on the computer to shape my instrumental tracks and to mix them down. I record one instrument, put it through a dozen or so DSP effects (including compression) on the computer, record another instrument tracks, "bounce" that down (again, digitally) to one with more effects and adjustments, add another instrument track, repeat the process again, etc.

The filters are not a part of the computer - they are a physical modification I made myself of a sound generation device. I have been using them to effectize sequenced beats and to make other instruments sound cool by switching the cutoffs (actually crossovers) on and off.

Reflecting, I'm using a lot of compression to try to get the sound I want. In the creative process I've been thinking like a computer and looking for more and more VST effects. I already have probably the best VST limiter there is so if it's not doing what I expected I need to adjust my way of thinking. But learning a bit about psychoacoustics (not my field - just an amateur) I'm getting turned off from producing music this way. I'm going back to basics with analog tape and no-computer for a while to restore my music brain, so thanks for your help but on reflection I need to change my approach and not an answer to my super-techno question on this thread.

waxx

Posted on Tuesday, August 23, 2005 - 10:30 pm:

vst's like psp vintagewarmer and e-phonic xpressor can do that, if i'm right. Both got a different effect (psp vintagewarmer is a limiter/tape simulator and e-phonic xpressor is a kneeless opto compressor).

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