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PianoFreek 28 August 2015

Hello there audiophiles.

I am looking to get your take on a little problem I am having.

The issue is to find a loudness/RMS based batch normalization tool (just for batch volume normalization) that is not WaveLab. I heard that Sequoia/Samplitude would have this feature. But looking over the PDF-manual it seemed to have it, and yet not have it!

Let me be more specific regarding my issue:

1. I have multiple recorded files, of a sampled key-instrument
2. they all need to be in the same level (perceptual level)
3. they can thus not all just be normalized from a peak level standard (i.e. at -3 db or similar)
4. they should not be limited (dynamically limited or clipped)
5. they should not be altered dynamically in any way what so ever
6. these are stereo audio files, and the stereo-relationship is not to be changed either

7. they need to be level normalized based on RMS and short-term loudness levels, not according to some broadcast standard, as this is determined by longer audio parts (over a program or so for television).
My files are all under 1 minute in length and need to be processed by a tool that can look at about 1,5 second of the peak audio material and from this intervall process audio files across a batch out of these values.

Wavelab is my go-to for this kind of thing, but last I tried with that program, I ended up with stereo phase errors. Wavelab had changed my Left and Right channel individually, based on my settings. There was no setting for dual mono or stereo-file relationship, and this is where I am right now.

Can anyone help me in finding the right tool?

Thanks a lot.

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Comments

pcrecord Mon, 08/31/2015 - 03:03

PianoFreek, post: 431908, member: 49413 wrote: 1. I have multiple recorded files, of a sampled key-instrument
2. they all need to be in the same level (perceptual level)
3. they can thus not all just be normalized from a peak level standard (i.e. at -3 db or similar)
4. they should not be limited (dynamically limited or clipped)
5. they should not be altered dynamically in any way what so ever
6. these are stereo audio files, and the stereo-relationship is not to be changed either
7. they need to be level normalized based on RMS and short-term loudness levels, not according to some broadcast standard, as this is determined by longer audio parts (over a program or so for television).

Unless I'm missing something, your point 2 and 7 are not compatible. Perceived levels and RMS levels aren't gonna give the same results.
If you were looking only looking for peek levels, I'd recommand AVS audio converter. It's very fast, process multiple files at once and it's not expensive.
Also processing 1.5 second ain't gonna give consistent levels unless your audio content is always at the same level from start to end.
I'm currious to know what those audio files are...

DonnyThompson Mon, 08/31/2015 - 04:23

pcrecord, post: 431952, member: 46460 wrote: I'm currious to know what those audio files are...

As am I - because this will largely determine the level(s) of the final-out ( final product)... And as Marco cites, your goals with #'s 2 and 7 on your list are conflicting.
If you are normalizing one file - which may be dramatically different from another in transients, dynamics, and RMS short-term, your common average between the two could differ greatly, and unless your levels are always the same, I fail to see how 1.5 seconds will serve in giving you what you want.

I have to be missing something here...

paulears Mon, 08/31/2015 - 07:03

To maintain stereo integrity, you would have to convert the mono tracks to an interleaved stereo, which should then process properly, but like the others, I'm not really sure your parameters would be considered normalisation? You seem to need processing, based on content and time, and that's not the same as just doing a little maths?

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