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December 13, 2007, 01:06:22 AM
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Topic: Adobe Audition and the Normalize Function  (Read 690 times)
« on: January 30, 2006, 08:55:36 PM »
Liebo Offline
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I am a newbie to these forums, but I have been using Adobe Audition for some time and Cool Edit before that. I have extensive experience in a very limited number of uses. (Primarily I use it to do simple edits on sound clips and music, with some occasional mixing involved.) But what I am having trouble with is the normalize function. I imagine that once I explain my issue and someone leads me in the right direction that I'll feel pretty silly. So here goes...

Some of the music that I am processing, for any number of reasons, extends beyond the 0 dB level. During playback, it reaches into the red and the playback is understandably substandard. I do not want to use the hard limiting feature because, by my understanding, it will more or less clip the waveform, resulting in a wave that is not entirely similar to the original. What I would prefer to do is normalize the entire song, adjusting the entire wave so it more closely resembles the original.

So I normalize the song, play it back to my satisfaction, then save it. But when I re-open the song in Audition, voila, it has returned to its original version before it was normalized, with its original levels. I have also run the batch processing feature on a group of files and gotten the same result.

So, what am I missing? Is there a check box unchecked? A button unpressed? An option unclicked? All I want is to quickly normalize one file first, then be able to batch process the others when I am certain the function will save properly. I imagine the fix is pretty easy and I have just missed it. So I am asking for your assistance.

And I am prepared to look silly--I'm even looking forward to it.

Thank you in advance for your help.
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Reply #1
« on: January 30, 2006, 09:55:16 PM »
Graeme Offline
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Quote from: Liebo
I do not want to use the hard limiting feature because, by my understanding, it will more or less clip the waveform, resulting in a wave that is not entirely similar to the original. What I would prefer to do is normalize the entire song, adjusting the entire wave so it more closely resembles the original.


I think you have a misunderstanding of these two functions.  If you have material which is clipping, then using the normalising function will not do anythig to reduce the clipping (or the sounds made by that clipping).  By the same token, the hard limiting function is intended for use on files which do not clip, but on you wish the raise the RMS level (to increase its perceived 'loudness').

In your case, you need neither of these functions.  What you should be doing is using the clip restoration function, as this does make an attempt to resotr the original waveform.

Quote from: Liebo
So I normalize the song, play it back to my satisfaction, then save it. But when I re-open the song in Audition, voila, it has returned to its original version before it was normalized, with its original levels.


Now this, I simply don't understand.  If you have 'nornmalised' a file and saved it, I can see no reason why those changes should not have been saved.  Unless, of course, you are working in multitrack view, where such changes are non-destructive.  If this is the case, then try working in edit view.
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Reply #2
« on: January 31, 2006, 12:26:37 AM »
alanofoz Offline
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One of your statements:-

Quote
Some of the music that I am processing, for any number of reasons, extends beyond the 0 dB level.


I'm interpreting differently from Graeme, and either interpretation is possible. Are you actually clipping at these points, or do you have a 32 bit file which actually goes above 0dBFS? In which case hard limiting might be a reasonable solution if they are occasional peaks. Hard limiting is not repeat not similar to clipping.

Quote
So I normalize the song, play it back to my satisfaction, then save it. But when I re-open the song in Audition, voila, it has returned to its original version before it was normalized, with its original levels.


Like Graeme, I don't understand this either. Are your original files read only? (But the behaviour is not quite right for that either).
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Cheers,
Alan

Bunyip Bush Band
Reply #3
« on: January 31, 2006, 04:25:06 AM »
Liebo Offline
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OK, to clarify what the problem ultimately is...

I open a file in which the waveform reaches the 0 db level; during playback it reaches into the red level. I normalize the wave to 98%, so it doesn't quite reach 0 db and no longer hits the red during playback. Satisfied with the result, I save the file.

But when I reopen the file, for some reason it has returned to its original levels. The waveform again reaches 0 db and playback goes into the red.

The file is 16-bit and is not read-only, so it wouldn't be that.

That is what is so confusing. Any thoughts? Do I still sound silly?

(By the way, the reason I use normalize is to maintain uniform levels throughout my files. Some of them have the problem I have described above, but others had lower levels and needed to be boosted.)

Thanks again for your help.
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Reply #4
« on: January 31, 2006, 08:39:15 AM »
Graeme Offline
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Quote from: Liebo
But when I reopen the file, for some reason it has returned to its original levels. The waveform again reaches 0 db and playback goes into the red.


That's certainly not behaviour we would expect.

Have you tried saving the file under a different name?  (Not that it should make any difference, just trying to limit the possibilities here).

Quote from: Liebo
(By the way, the reason I use normalize is to maintain uniform levels throughout my files. Some of them have the problem I have described above, but others had lower levels and needed to be boosted.)


If you want a group of files to have the same perceived volume, the you should be looking at the Group Waveform Normalise function.  Simply normalising each one to a single specific level wouldn't work, if the material is of differing types.
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