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How to amplify a low signal to ‘HARD LIMITING’ clipping?

 
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Joel



Joined: 24 Jan 2009
Posts: 9

PostPosted: Sat Jan 24, 2009 5:07 am    Post subject: How to amplify a low signal to ‘HARD LIMITING’ clipping? Reply with quote

I am using a VST (TLs-3127-LEA.dll) to do ‘hard limiting’ as in CoolEdit2.1. When I have VST:“Processing” checked, w/preset=”Peaks Delight” and turn up the gain to say 30, I hear what sounds like the start of signal clipping & the “Clip” light goes on, which I would expect. However the sound contains more Noise than CoolEdit would have added & is unacceptable.

At this point, “Processing” gets Unchecked, & “Apply” is depressed. A processing image to show that Wavosaur 1.0.0.4 is processing, comes&goes then the selected signal has in fact been changed & is larger with just a small amount of soft clipping (as expected). However the sound is garbled and totally unusable. What am I doing wrong? Wrong VST?? (I am new at audio processing)
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Rex
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Joined: 05 Oct 2006
Posts: 796

PostPosted: Sat Jan 24, 2009 11:55 am    Post subject: Reply with quote

Hello,

When you are in "processing" mode you can turn the knobs while you are playing the sound, and tweak the parameters until you get the sound you want.

with TLs-3127-LEA , make sure to set it on "limiter" (the peak delight preset is in "compressor" mode) and don't forget to adjust the "reduction" knob too.

You may use TLs-Maximizer, its sounds very good and is faster to obtain results. it works well as limiter
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Joel



Joined: 24 Jan 2009
Posts: 9

PostPosted: Sun Jan 25, 2009 10:21 pm    Post subject: Reply with quote

Thank you. Don’t know how I could have missed it. This vst will be usable.

However when I compare the wave forms the CoolEdit waves are never clipped even at 555db increase (totally ridiculous but to show the theory) , the vst waves go to square waves, when the gain is set to around 60.

Also is there another vst that has more analytical control & feedback. In cooledit instead of a ‘pretty’ interface, they have the tools to let me know what is going on that I may or may not hear or see. I can for example ask for the stats to tell me if I apply a certain db gain, that it will result in X% of samples being clipped. Very handy to know. I have noticed that many of the VSTs are pretty but at the expense of usefulness. I don’t think that digital I/O Is horrible. If you could point to there there are more analytical VSTs, I would appreciate that. thanks....
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Rex
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Joined: 05 Oct 2006
Posts: 796

PostPosted: Sun Jan 25, 2009 10:57 pm    Post subject: Reply with quote

I don't use Cool Edit and i don't know how this hard limiting is working nor how it looks.

There must be some VST effects with analytics information, i'll do a search about that.

I know "func shaper" but it's a waveshaper : http://www.rs-met.com/freebies.html

Don't forget that VST works in 32 bits float (or 64 bits), so the "X% of samples being clipped" is not very useful, you can have values > 1.f that are not "clipped" until you convert to 16 bits.
it's a good idea to normalise in Wavosaur before saving to 16 bits.
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Joel



Joined: 24 Jan 2009
Posts: 9

PostPosted: Thu Jan 29, 2009 1:32 am    Post subject: Reply with quote

Thanks again. I m learning much, as I go w/u’r help. I just found out how to control the vst via the input&Output gains together. Working much better.

However u brought up another ? which I had no idea I am dealing with. U said:
“…VST works in 32 bits float (or 64 bits), so the "X% of samples being clipped" is not very useful, you can have values > 1.float that are not "clipped" until you convert to 16 bits.”

My primary input signal is pcm wav @8bits/8kbs (voice/audioBooks w/smallFileSize) on Vista64. I tried audacity first but it could not handle 8bit. I just mention this because u were talking about 32/64 internal word representation that Wavosaur was using. Do I need to be concerned? & could u point to where there might be any further reading/info on this subject. Thank you again….
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Rex
Site Admin


Joined: 05 Oct 2006
Posts: 796

PostPosted: Thu Jan 29, 2009 8:55 am    Post subject: Reply with quote

All VSTs work in 32 or 64 bits, so when you use VST plugins in Wavosaur to process your sound you have 8 bits pcm -> 32 bits floats (-1 +1) conversion and vice versa

you have to be aware that some plugins can go beyond the "limits" (-1 +1) without clipping, but the clipping will occurs when converting from 32 bits to 8 bits, you can avoid that with the normalisation feature of Wavosaur, to bring back the range to -1 +1
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