Fons Adriaensen wrote on Thu, Jul 25, 2013 at 07:40:53PM +0000:
On Thu, Jul 25, 2013 at 02:21:50PM -0400, Martin
Cracauer wrote:
Does anybody know why wav files that do not have
clipping would clip
when encoding them with lame?
Because the peak sample value says nothing about the real
level. Take a square wave with peak values +/-1. It is the
sum of a number of sine waves, with frequencies 1,3,5,7,...
times the frequency of the square wave. The first one has
a peak value of about +/-1.27, that is +2 dB. So any encoding
that looks at the spectrum (and mp3 does) will see a level
that is +2 dB.
Ah. Makes sense. Thanks so much!
Is there a rule of thumb how many db less I should give music to avoid
this? What would be the value for pink noise starting at 40 Hz?
If you really want to normalise on the peak level, use
a
lower one. If you want all your samples to have the same
loudness, use RMS instead of peak, or a real loudness
measurement such as provided by ebumeter.
Is there a way to hook up ebumeter to just an audio file or a stream
not associated with real time? It seems to come in a jack package only.
Thanks again
Martin
--
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Martin Cracauer <cracauer(a)cons.org>
http://www.cons.org/cracauer/