Topic: SF 6.0 normalizing versus rms

[biscorbit]

Can somebody tell me whats better.  Also if you can please tell me the exact process in which you use in normalizing a wav. on soundforge 6.0.  Step by step would be appreciated.


thanks,

nick c.

[nic]

-double click the wave form so that the entire wav is highlited
-goto process and select normalize
-make sure peak level is checked
-on the left hand side, set the normalization to  0.00db(slider to the top)
-click ok
-save waveform

[biscorbit]

I did that last night and the results were minimal.  its very strange.  On the other hand, i used rms and saw the wav get larger and louder immediately.  

[nic]

the results will be minimal if you have a stray peak that is allready near "0".

using RMS is similar to running a compressor(ie, taming those stray loud peaks) on the waveform first, but without the flexibility a true compressor offers

[jctibbitts]

can you elaborate on that LUVEAN?

[nic]

with a waveform you have dynamics, peaks and valleys.
when normalizing(peak level) you are raising the loudest peak to whatever value you specify, normally 0.  well, if the original waveform had a couple stray peaks that were allready near 0 but the rest of the wav was lower, then running the normalization(peak level) would only raise the enitre waveform until those stray peaks reached the desired level, thus appearing to do relatively nothing to the waveform in general.

if you run normalization(rms) on the same waveform it actually takes an average of the peak levels from across the waveform instead of using only the loudest peaks to adjust to 0(or whatever you desired). the problem with rms is that alters the dynamicsof the waveform, but this isnt allways a bad thing. that is why I said this is similar to runing a compressor , because it will take those stray loud peaks and bring them more inline with the rest of the waveform.
a compresoor gives you much more fine detail in controlling the resulting waveform than using normalization(rms) will

[jctibbitts]

+t luvean,

thanks...one more question.  if i have a recording (sbd>ua5>jb3) where my levels are low across the plane, would i want to use the peak or rms normalization?  i'd want the peak right?

and if i had a recording where the levels were a little high i'd want to compress it or even it out with the rms function right?  how exactly does normalize work?  it doesn't screw up the file at all?  

sorry, i am a newb...it just seems logically it is easy to make a file smaller, but when you make a file bigger you lose quality.  for example going to wav > mp3...but going from an original mp3 you cant go back to the lossless wav.  does that analogy make sense?

 i just dont' really understand music editing yet, and i am worried that i might alter the recording rather than just amplify it.  

[nic]

"thanks...one more question.  if i have a recording (sbd>ua5>jb3) where my levels are low across the plane, would i want to use the peak or rms normalization?  i'd want the peak right?"

correct


"and if i had a recording where the levels were a little high i'd want to compress it or even it out with the rms function right?  how exactly does normalize work?  it doesn't screw up the file at all? "

if the levels arent clipping(over 0) then I wouldnt worry about it unless theres a drastic difference in levels across the whole waveform. instead of normalizing(rms) look at Wavehammer...SF6's built in compressor. dont have SF in front of me so I cant tell you exactly where it is located.

compressor how-to basics:
1. Set the threshold level to the point at which you wish compression to take effect. Signals below this level will not be affected. Signal levels above the threshold will be reduced according to the compression ratio.
2. Set the compression ratio. Ratios of 5:1 or less will produce fairly smooth compression; ratios of 10:1 or more will produce more severe cutting off. ratios of 2:1 ad 3:1 are nice in that you generally cant hear the compressor "breathe"
3. Set the attack time. This is the delay between detection of a signal above the threshold, and the commencement of compression (ie. the time it takes to "attack" the signal). measured in ms, I found <1ms produces good results.
4. Set the decay time. This is the time taken to release the signal from compression. measured in ms, I usually set this around 30ms.
5. output gain. you should have level meters showing what the effect of the compressor is. bump up your output gain to get the highest peaks near 0 coming out of the compressor.

some compressors have input gain, but I usually leave this flat.

[AdamZ]

Thanks luvean!!
I'd +t if I could...

[jctibbitts]

i have SF 7...does that function cross over?

[nic]

i have SF 7...does that function cross over?

dont have SF7, but I am assuming it is still there.
they introduced it with SF6

[jctibbitts]

sorry...one more question...total nOOb here:

i have a couple recordings that have audidle clipping and i have just basically stored those recordings and figured them as wasted shows. is there anything i can do to 'fix' them? or is it impossible after the fact?

[nic]

dont know about that in SF.
I used to know how to do it in Cool Edit, but not anymore...sorry

[jctibbitts]

bump for:

i have a couple recordings that have audidle clipping and i have just basically stored those recordings and figured them as wasted shows. is there anything i can do to 'fix' them? or is it impossible after the fact?

[jctibbitts]

if you run normalization(rms) on the same waveform it actually takes an average of the peak levels from across the waveform instead of using only the loudest peaks to adjust to 0(or whatever you desired). the problem with rms is that alters the dynamicsof the waveform, but this isnt allways a bad thing. that is why I said this is similar to runing a compressor , because it will take those stray loud peaks and bring them more inline with the rest of the waveform.
a compresoor gives you much more fine detail in controlling the resulting waveform than using normalization(rms) will

i did both and i think the rms did a better job, more what i am looking for.  it's weird bc everyone has said to do the peak method, and it didn't much to the wav file.  but the rms did too much boosting of the volume. i don't know enough about it to play with the parameters of the rms feature though