HOLMImpulse: Measuring Frequency- & Impulse-Response
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diyAudio Member

Join Date: Mar 2003
Location: Denmark, Copenhagen
Quote:
 Originally Posted by john k... From a numerical analysis point of view smoothing always reduces to some form of integration: A'(fc) = integral [W(f)A(f)]/ integral[W(f)] over some limits f1 to f2 where fc is between them. W(f) is a weighting function of choice, typically symmetric about fc. Obviously, in the digital domain the integrals are replaced with their representative summations.
Exactly! As in my appendix A.3.3 Absolute smoothing of discrete frequency values in

So what we need to discuss is "only" W(f)
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gedlee
diyAudio Member

Join Date: Dec 2004
Location: Novi, Michigan
Quote:
 Originally Posted by JohnPM To resample at log spacing you really should first smooth the data, otherwise your resampling introduces aliasing. For example, to convert the data to 96PPO log spaced it is best to first smooth to 48PPO before doing the decimation (which is then simply selecting the desired 96PPO samples from the smoothed set).
Except that I am not resampling in the sense of "decimation" so aliasing is not possible - the "sample rate" is not changing, only the spacing of the data points from linear to log.
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Earl Geddes Gedlee Website

JohnPM
diyAudio Member

Join Date: Oct 2008
Quote:
 Originally Posted by gedlee Except that I am not resampling in the sense of "decimation" so aliasing is not possible - the "sample rate" is not changing, only the spacing of the data points from linear to log.
Well I expect there are far fewer points in your log spaced output than there were in the original linearly spaced FFT data, so that constitutes decimation in my book . The log spaced, filtered result would have a non-uniform sample interval (i.e. the frequency spacing of the samples) but the interval would typically be much greater than that of the FFT data it was derived from, so the usual decimation caveats apply - if the bandwidth is not limited to comply with the effective inter-sample spacing of the final data there will be aliasing artefacts. "Bandwidth" in this context refers to the rate of change in the frequency domain of the frequency response samples, rather than the more usual meaning in the context of a time series.

Last edited by JohnPM; 19th October 2009 at 02:12 PM. Reason: Clarified sentence re sample interval

Pano
diyAudio Moderator

Join Date: Oct 2004
Location: SW Florida
Quote:
 Originally Posted by askbojesen * Device & signal: Invert input has no effect
Thanks for fixing that. I thought I was going crazy!

gedlee
diyAudio Member

Join Date: Dec 2004
Location: Novi, Michigan
Quote:
 Originally Posted by JohnPM Well I expect there are far fewer points in your log spaced output than there were in the original linearly spaced FFT data, so that constitutes decimation in my book . The log spaced, filtered result would have a non-uniform sample interval (i.e. the frequency spacing of the samples) but the interval would typically be much greater than that of the FFT data it was derived from, so the usual decimation caveats apply - if the bandwidth is not limited to comply with the effective inter-sample spacing of the final data there will be aliasing artefacts. "Bandwidth" in this context refers to the rate of change in the frequency domain of the frequency response samples, rather than the more usual meaning in the context of a time series.
John

I have to disagree with you here. Since the bandwidth of the final data is the same as the bandwidth of the original there won't be any aliasing. There may be other issues, but aliasing, as usually defined - out of band components folding back down into the passband - isn't one of them. At least not the way that I do it.
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Earl Geddes Gedlee Website

diyAudio Member

Join Date: Mar 2003
Location: Denmark, Copenhagen
Importing / Exporting FR

Quote:
 Originally Posted by gedlee How can I import a FR curve? is there documentation of the proper format? After smoothing the FR curve is purely real there is no phase. Is this a problem?
No it's not a problem - You can import the amplitude only - see the now updated User Guide v0.0.7
Section "3.1 Importing a frequency response"

Quote:
 If I smooth in Holm, can those curves be exported en-mass like the IRs?
No not yet...
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Last edited by askbojesen; 19th October 2009 at 08:33 PM.

 10th November 2009, 07:36 PM #527 David_Web   diyAudio Member   Join Date: Sep 2008 I just found out I could import/export .wav in LTspice so I had some fun simulating an amp with a music signal. Then convolved it with HOLM and got perfect result. Agrees with spice own FR curve and phase. Did a sweep as well and it agrees with the music version, and the plotted distortion seems to be spot on.
 10th November 2009, 08:34 PM #528 askbojesen   diyAudio Member   Join Date: Mar 2003 Location: Denmark, Copenhagen Thanks! Always nice to hear that HOLMImpulse is consistent __________________ Follow your first Impulse with HOLMImpulse
 22nd November 2009, 10:08 AM #529 Henckel   diyAudio Member   Join Date: Sep 2002 Location: Copenhagen, Denmark First of all - Thank You for providing this great piece of Software -" Godt Gået !" I have an application where i would like HOLMImpulse to work with JACK; JACK | connecting a world of audio However whenever i select JACK as my Asio In/out device in HOLMIpulse my Jack server dies. Since i have been using JACK with a number of other Asio programs with no problems i tend to belive its HOLMImpulse that is causing the problem. I use Windows XP, JACK V 1.9.3 & HOLMImpulse v1.4.2.0 Fault from JACK server when dying: "JackEngine::ClientCloseAux wait error ref = 2" Any Clues ? Morten
 22nd November 2009, 03:19 PM #530 Key   diyAudio Member   Join Date: Jan 2009 I don't think you can use ASIO on the same outputs with 2 apps without some sort of a bridge like ReWire or Reaper's ReRoute. Hmm might be wrong. Does jack work directly on your interface almost like it's the the asio mixer/driver interface? In that case maybe it should be working. Last edited by Key; 22nd November 2009 at 03:23 PM.

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