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 -- %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% Martin Cracauer <cracauer@xxxxxxxx> http://www.cons.org/cracauer/ _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
