On Nov 1, 2018, at 10:56, Rob Landry <41001...@interpring.com> wrote:
> My understanding of "normalize" is that it means multiplying every sample in
> a file by some coefficient calculated to set the largest sample in the file,
> i.e. the peak amplitude, to a specific point, such as -13 dBfs.
>
On Thu, 1 Nov 2018 11:06:31 +1300
Robert Jeffares wrote:
> In digital audio, overshoots cause lots of instant distortion.
Almost right.
"Overshoot" has a specific technical meaning, ( involving bass ) and
isn't as intuitive as one might think, but I think I know what you mean.
In analog
On Thu, 1 Nov 2018 07:45:26 -0400
Fred Gleason wrote:
> and rewriting audio can cause quality degradation (especially if MPEG Layer
> II format is
> involved).
Can cause ?
I would argue that rewriting audio in any format other than what it already is,
absolutely *will* cause some degree of
My understanding of "normalize" is that it means multiplying every sample
in a file by some coefficient calculated to set the largest sample in the
file, i.e. the peak amplitude, to a specific point, such as -13 dBfs.
However, lately I've seen "normalize" used to mean adjusting the *average*
On Oct 31, 2018, at 15:43, Gregory Avedissian wrote:
> Is it safe to do something as simple as having the script go to /var/snd
> and run:
> for i in *.wav ; do sox --norm=-5 "$i" /tmp/"$i" && cp /tmp/"$i" "$i" ; done
>
> Oh yeah, there will be rm /tmp/"$i" in there, too. Just noticed that.
>
