Re: batch processing -> loudness matching

RD Jones wrote:

[someone asked]

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

[Richard Crowley suggested]

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.

I still concur that normalizing to a target average is the way to go, but
I'm not sure which tool that will be the optimum to choose.

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.

Which is why average values by some, suitable, definition is the answer.
There will still be some freak files, something gentle with stray percussive
peaks is likely to end up too loud by that procedure.

Thank you guys for fast answers.
Normalizing is, as said before, not an option.

Allow me to humbly suggest using the wording "aligning" for this, Audition
calls it "group level normalize".

Even having the whole package in the same perceived volume is not,
since files have different volumes for a reason.

This is why I always do it manually even when aligning to a target AVG.

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.

It may - or may not - work better if he highpasses the data to calculate on
at 120 Hz.

[ la snippage molto]

This is a recipe for clipping, massive hard digital clipping.

No. Do it in 32 bit and include a limiter or just include a peak check.

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.

Yes, try working with 32 bit files rather than 24 bit files, 8 bit float
above 0 dB makes life comfortable.

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.

Get real, digital clipping is clean and therefore only moderate audible and
it is not audble if shorter than a couple of milliseconds. Anybody can and
should test how such things work and test your favorite softwares unclip
tool and what it can do and can not do, so that you know when to worry.

Also it can't be bad and frowned upon with all those gravely clipped cd's
that have overshot the muliband-processor, looks more like an industry
standard to me.

What is really really really bad is opamps that reverse polarity or do other
silly things when overdriven, included any such opamps in the input of the
converter chip. Test your equipment and find out what it does.

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.

Ah, that's great, it is easy to come up with a target value, it is just a
statistics chore on the selected material.

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

Yerp, the way to go is to convert to 32 bit file, process, convert back,
doing this on 16 bit files is not in my opinion the preferable choice. Stuff
that is to end up as mp3 should stay well out of the upper dB or dB's. Also
something that any sound engineer should test ....

rd

Kind regards

Peter Larsen



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