General question on Sigma's Hilbert Transform block

I'm running the ADAU1451 under SigmaStudio 4.2 and have a couple of questions about the Hilbert Transform block.

First, when I click on the appropriate link in the Toolbox Wiki, only an Error page comes up.  Is a discussion of this very useful block in the works?  I suppose it's fairly self-explanatory to those who have Hilbert experience, but it would be nice to have confirmation of the pin names at least, and what to expect at the outputs.

A 'DIY' FIR approach to a Hilbert Transform was offered in these pages prior to this block's appearance in some of the device menu trees, .  More than 4 years ago, KJBob gave a good account of the difference between the menu-tree block and the DIY two-filter approach in this posting: https://ez.analog.com/dsp/sigmadsp/f/q-a/65970/adau1446-hilbert-transform-response.

In that posting, KJBob points out that the menu-tree block (which is an IIR implementation, far more accurate and with less overhead than the DIY one) has a non-constant phase/delay relationship between the input and the 'real' or cosine output.  A filter to mimic the cosine delay characteristic is not a simple matter.  For those of us who need phase coherence between the input signal and the cosine output, we can either use the DIY approach, as KJBob suggests, plus a second fixed delay to match the cosine delay-only, or simply use the cosine output of a second menu-tree block to provide identical time and phase smear.  I believe that, even with a second Hilbert block, the overhead is less this way.  Sound plausible?



Misspelling!
[edited by: electrojim at 3:20 AM (GMT 0) on 14 Jan 2019]
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  •      Hello Jim,

         Just as you expect, the packaged Hilberts use 103 instructions each -- two would still use far less than one FIR.  Below I made a simple headphone effect with two Hilbert transforms, one active and one reference.  By adding a relative 90 degree phase it moves the sound image from inside your head to all around you.  Somewhere in EZ I have also posted a variable audio notch filter using SSB modulator / demod to shift the incoming audio (instead of moving the notch filter, which isn't possible on a ADAU1701 without external coefficient calculation).

         Best regards,

         Bob

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