Re: batch processing -> loudness matching

On Aug 5, 10:05 am, RD Jones <ann...@xxxxxxxx> wrote:
On Aug 4, 12:51 pm, Kompan <rok.kom...@xxxxxxxxx> wrote:



On Aug 4, 7:22 pm, Laurence Payne <l...@xxxxxxxxxxxxxxxxxxx> wrote:

On Tue, 4 Aug 2009 09:51:00 -0700, "Richard Crowley"

<rcrow...@xxxxxxxxx> wrote:
1. find out the RMS or loudness of the file aaa.wav
2. change volume of bbb.wav so it matches the loudness of aaa.wav

I'm most comfortable in Wavelab, is there maybe a plug-in that this
could be done with?

I have over 2k files to process, any help or direction would be
greatly appreciated.

Adobe Audition has a "Group Waveform Normalize" feature.

Why not just normalize them all to some fixed point? That way
you could spread the job across multiple machines and run
them in parallel to speed the process.

Because unless all the wav files were very similar in musical content
and  recording technique, and had similar peak levels (even if only a
single hit in each track) normalising wouldn't get anywhere near
giving equal perceived volume.

Hey.

Thank you guys for fast answers.
Normalizing is, as said before, not an option.
Even having the whole package in the same perceived volume is not,
since files have different volumes for a reason.

I have a guy who is making a simple tool at the moment.
He will be calculating RMS
(http://en.wikipedia.org/wiki/Root_mean_square)
and then adjusting volume to other file so that their calculated RMS
values are the same.
I'll let you know how it works.

In the mean time, since (as said before) loudness is a complicated
thing, are there any other simple "checks" that we could put in to the
RMS calculation, so that perceived volume would match as much as
possible?
(currently we're ignoring everything under 0.3%, to ignore the
silence)

Of course, suggestions on a proven piece of software or plug-in is
even better.

This is a recipe for clipping, massive hard digital clipping.
Without calculating the peak levels, an indiscriminate application
of digital gain (it's almost always done to RAISE levels, isn't it ?)
will certainly result in peaks being pushed past 0dB.

Let me say it in no uncertain terms...
HARD DIGITAL CLIPPING IS BAD.
It is bad form, frowned upon by professionals, and to be avoided.

Sound Forge will do what you want in batch mode, but you have
to establish the rms level of "aaa.wav" first manually, then run the
"normalise using: average rms power" and select "if clipping occurs:
apply dynamic compression". You could also chain together an
rms leveling compressor and a peak limiter, but you'd still need to
supply the rms reference.

Just so we're clear here, raising rms levels without also limiting
peaks will most likely result in clipping.

rd

Hey.

Thanks again for suggestions.

The tool works better than expected. Loudnesses really do match. Can't
believe how simple this was.

Unfortunately we do experience some clipping. Most of the time there
are only few samples that got a few % over every 10 seconds or so.
They're clamped to 100%. The dynamics of audio files is fortunately
pretty high.
We could probably do some sort of soft clipping, shouldn't be to hard
to calculate:
http://upload.wikimedia.org/wikipedia/commons/2/2a/Clipping_compared_to_limiting.svg
and see how it works.
Or we could just log in to a .txt file where clipping occurs and limit
peaks before running the tool.

Will check sound forge's batch processing.

Thank you all and feel free to throw in a suggestion, if you think of
one.
.

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