[linux-audio-user] Peak Level Monitor/Limiting

[Arnold Krille]

2007/3/29, Dragan Noveski <perodog at gmx.net>:
> Arnold Krille wrote:
> > 2007/3/29, Folderol <folderol at ukfsn.org>:
> >> As many of you will know I principally use ZynAddSubFX to generate most
> >> of my sounds, so I can't really be sure if this is a Zyn problem or a
> >> general one with software synths and mixing.
> >> Very occasionally I get high amplitude spikes typically 6dB above the
> >> 'normal' peaks. These give a false idea of the signal level so that if
> >> I'm not watchful I end up recording at a much lower level that I need
> >> to.
> > Whats the problem with recording at lower volume? As long as there is
> > no da/ad involved you won't have any quality-loss.
> > But I prefer to work with hot-signals too. ;-)
> i think, recording at low level, you are kind of "loosing disc space".
> lets say, if you are recording at 24bit, on low level the signal
> information will take only 12 bit (example) and the rest of 12 bit will
> be only empty data information.
> don't think that one can repair that with some gain plugin or
> normalization...

Well, I was assuming you are recording with ardour, timemachine or
some other jack-based app. These use float internally and at least the
first two also save to disk in float-format. Which means that 32bit
are 32bit regardless if you are near to 0 or using the whole -1..1
range.
And as float-format means x.xxxEyy you won't loose any quality if the
. moves forth and back...

But as I said, I prefer hot levels too. And float-format also helps
for this as you don't get overflows in the normal audio-range which
means you can save files where amplitude goes beyond -1..1-range. You
should limit/scale that on playback though...

Arnold

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