Speech recording - What Compressor parameters should I use?
Posted: Tue Aug 09, 2011 6:03 am
I'm doing speech recording from a very basic setup - a single Shure SM58 clone mic, Kustom KMP4080 amp, and a couple of Kustom KSC10 speakers. The amp has two outs, a pair of RCA jacks and a 1/4" mono jack. I run the mono jack into the mic input jack on a old rebuilt HP/Compaq notebook PC running Windows XP, and record the audio using Audacity 1.2. (I know, I should install the 1.3.13 beta, and will, as soon as I gain enough familiarity using it on my desktop.)
For months, I've been taking the recorded file to my desktop to do manual editing of the file, typically consisting of: Trim the start; trim and fadeout the end; run minimal noise reduction; run the high pass filter with a cutoff of 300. Then I spend what can turn out to be hours going thru and manually selecting the spots where clipping occurs, as well as areas that get close to clipping, and dropping the amplitude down on them, in order to run the amplify effect on the whole file to maximize the sound levels.
Then, it finally dawns on me that most of the above effort can be done a lot faster and probably a lot better, by running the Compressor effect. Great! But can anyone suggest what I should use for the parameter settings when recording speech? Threshold? Noise Floor? Ratio? Attack & Delay times? Despite pouring over the description of the effect in the documentation, I have no clue.
Also, one of the basic difficulties I'm having is understanding the graph used in the Compressor. Forgive a ignorant newbie's remarks, but it seems backwards to me. Why are all the decibel values negative? Why do the axes cross at -60dB and the zero point appear in the upper right corner?
Any and all comments or suggestions appreciated!
For months, I've been taking the recorded file to my desktop to do manual editing of the file, typically consisting of: Trim the start; trim and fadeout the end; run minimal noise reduction; run the high pass filter with a cutoff of 300. Then I spend what can turn out to be hours going thru and manually selecting the spots where clipping occurs, as well as areas that get close to clipping, and dropping the amplitude down on them, in order to run the amplify effect on the whole file to maximize the sound levels.
Then, it finally dawns on me that most of the above effort can be done a lot faster and probably a lot better, by running the Compressor effect. Great! But can anyone suggest what I should use for the parameter settings when recording speech? Threshold? Noise Floor? Ratio? Attack & Delay times? Despite pouring over the description of the effect in the documentation, I have no clue.
Also, one of the basic difficulties I'm having is understanding the graph used in the Compressor. Forgive a ignorant newbie's remarks, but it seems backwards to me. Why are all the decibel values negative? Why do the axes cross at -60dB and the zero point appear in the upper right corner?
Any and all comments or suggestions appreciated!