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Forums     Plugins     Impulse Modeler and Deconvolver Deconvolver Questions

Hi Aleksey!

Great work on Deconvolver, but the html manual does not provide info on these questions:

1.  When using the Test Tone generator should 3 seconds always be used?  Or under what conditions would it be better to use 1 second or 12 seconds?

2.  I've read on this forum that you feel that impulse files above 16 bit are not worth it, but the default bit rate for the test tones are 24 bit.  So if I use a 24 bit test tone must I also make the impulse 24 bit or is it ok to use a 24 bit test tone to produce a 16 bit impulse?

3.  Should I always Normalize to -0.3 dBFS level or are there some situations that I should not?  Does normalizing lower the quality of the impulse in any way?

4.  When playing your test tone through a hardware reverb unit should I try to play the file at the highest possible volume that will not clip the hardware unit?  Or does it not matter?  I'm trying to get the highest quality impulses.

5.  Finally I am trying to get the highest quality impulses, is this the way:

STEP-1: 3 second 24 bit test tone w/ fade in/fade out applied

STEP-2: Input it digitally (via spdif) into the hardware unit at maximum volume that avoids clipping

STEP-3: Capture the digital out via spidf, and make sure you start recording a few seconds before inputing the test tone...

STEP-4: Deconvole using these settings: Reversed Technique, MP Transform and normalize.

Are any of the steps incorrect to achieve the best quality?

Thanks for making such a grea product and for great support.



1.  Answer depends on the audio device 'quality'.  If it's good quality 3 seconds should be enough - if it's a spring reverb, for example, you should use a longer test tone.  I personally tend to use 6 second test tones.

2.  You may mix 24-bits and 16-bits in any proportion.  The end result will have the lowest bit resolution in the chain.  For example, if audio device you are capturing is 8 bit, your final impulse will be 8 bit, too.

3.  Normalization is useful for editing the impulse manually (adding fades, cutting).  Normalization should not be applied when you are building an array of impulses, because otherwise gain level differences between the impulses will be lost.

4.  It does matter whether you playback at the high volume or at the low volume.  It's better to play at the highest volume to preserve bit depth.  At the same time, some tests revealed that by using a longer sine sweep bit depth is preserved as well.  It sounds pretty unbelievable, but it is possible to recover 16-bit impulse using a 8-bit soundcard just by using a suitably long test tone (however, the quality improvement is not huge considering you'll have to use a much longer test tone).

5.  I do not suggest you to use MP transform option - it rarely works good, and can be useful with highly non-linear hardware.

Thanks so much!  As usual your responses are clear and timely, you're the best.

You are welcome!
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