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Old 3rd March 2013, 06:47 AM   #31
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Quote:
Originally Posted by EUVL View Post
Jan,

Why so complicated ?

You know how Bob's Distortion Magnifier works.
You can use two channels of the sound card to do the same, or even your Handyscope.

Or I complete missed what you want to do (to view the residual error after subtracting the input signal) ?

Patrick
Patrick,

It will work in concept, but the cancellation of the source will not be enough for electronics work. If you recall the Baxandall difference test, they had to tweak not only the amplitude but also the phase response of the cancellation circuit to get a good null, and it drifted like hell.
Especially with systems that have say 0.01% distortion your null must be better than -80dB and that is a tall order, forget scopes or simple circuits.
I am working on a circuit that does it in real time as the basis for a feed-forward error correction system, and I am looking at differencing circuits with at least 200Mhz bandwidth. Yes, that's for audio

For loudspeaker measurements the analog way might be a bit easier as the distortion products are generally higher than in electronics, but Bill's and Joachim's system is in my opinion a very good concept.

jan
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Old 3rd March 2013, 06:52 AM   #32
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Quote:
Originally Posted by Joachim Gerhard View Post
Steven, no. As far as i can tell i am the only one that uses that method. I also helped Bill Waslo with his Distortion Isolation method. Actually i had the idea and Bill did the implementation. I also did the trouble shooting. Maybe this is something to wright up in Jans Linear Audio.
The scale is in Micro Pascal so the distortion is miniscule in absolute terms but quite anoying when you here the WAV. This is a very high quality speaker. Other distort more.
Joachim, you have a habit of coming up with very good and novel ideas all the time! I would be thrilled to publish an article about this in the next Linear Audio - you owe me an article anyway

I can't wait to get home from my 'wintersleep' in Alicante to give this a try. I would be particularly interested to apply it to amplifiers.

jan
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Old 3rd March 2013, 01:30 PM   #33
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Joachim,
Perhaps one of the good things that comes about from using fibrous stuffing in an enclosure is the damping of the reflective waves and the attenuation they bring about leaving the cabinet walls less work to attenuate those high spl levels you speak of, turning some of that energy into heat. One of the advantages I have been told by my friend who has helped me develop my cone material, he has all the B&K lab equipment, is that the cone does damp the back wave quite well compared to a paper pulp cone or most other composite cones and I won't even consider a metallic cone, I just don't like the ringing in the cone those have in the first place.

ps. Jan, this month I will become a subscriber to your magazine, it looks like I should be reading it for education purposes.
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Old 3rd March 2013, 01:46 PM   #34
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Jan, i can talk to Bill. Maybe we can do something together.
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Old 4th March 2013, 01:48 AM   #35
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Default Energy storage and distortion tests -

If you need more hands on deck, i am willing to help.

Thx-RNMarsh

BTW- I also did the tests for filter energy storage decades ago and showed it back then and again last summer on another forum... got nowhere with them. I bought the very first MLSSA. So, its nice to be in touch with similar thinking on that and this idea also.

Thx-RNMarsh

Does not sound flat.jpg

Last edited by RNMarsh; 4th March 2013 at 02:13 AM.
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Old 4th March 2013, 06:39 AM   #36
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It'l be a few weeks before I am back home and again can snuggle up in my lab
But I'll keep in touch.

jan
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Old 4th March 2013, 10:59 AM   #37
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I agree that a time-domain view of the residual is extremely useful, and a weak point of most soundcard-based systems.

The proper approach to implement it would be a tracking notch filter just as done in the analog domain (the mentioned subtraction methods etc. effectively form a notch filter too, just not with the same performance as a direct filter implementation). Probably not very difficult for someone handy with digital filters.

Samuel
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Old 4th March 2013, 11:43 AM   #38
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This is what Virtins tells me:

"Multi-Instrument supports FFT, FIR and IIR digital filters (please refer to Section 2.6.2.8 of the software manual for details). To suppress the fundamental frequency, you can use the Band Stop filter (right click anywhere within the oscilloscope window and choose Oscilloscope Processing). You just need to specify the starting and ending frequencies of the stop band and then evaluate the effect in the oscilloscope and spectrum analyzer windows. If it is OK, then check the Persist check box (in the oscilloscope processing dialog). You can then replay the filtered sound by clicking the Play or Cyclic play button in the Instrument toolbar. You can also save the filtered data as a WAV file into the hard disk.

For FFT filtering, the resolution is the same as the FFT resolution displayed at the lower left corner of the spectrum analyzer window (subject to spectrum leakage effect, depending on the signal frequency and record length). The resolution of FIR or IIR filter depends on the settings/filter coefficients used. The digital filtering is performed in real time but not simultaneously."

Sounds quite useful, but I asked a few more question, waiting for answer.
BTW This software has a 'no spectral leakage' button - when you click it, it finetunes the fundamental that is generated to be synchronous with the FFT so you do not need a window.

jan
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Last edited by jan.didden; 4th March 2013 at 11:47 AM.
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Old 4th March 2013, 01:56 PM   #39
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More:

"If you choose to use FFT filter and want to avoid spectral leakage, then the fundamental frequency should be equal to N × [Sampling Frequency]/[FFT Size], where N is an integer and FFT size is a power of 2. The FFT filter in the oscilloscope does not depend on the FFT size and Window function selected in the spectrum analyzer. It always uses no window function and automatically choose a FFT size equal to (thus no zero padding) or greater than the Record Length (thus padding zero at the end). Therefore your Record Length of the data should be of a power of 2 to avoid zero padding. "

nothing earh-shattering here, but, most interesting:

Question: "I assume that with FFT filtering, the relevant frequency components are completely removed from the particular FFT bins, so that suppression is absolute?"

Answer: "Yes provided that the conditions for no spectrum leakage are met."

jan
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Old 4th March 2013, 05:08 PM   #40
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I found several DSP filter software and hardware by Googling on 'DSP Filters.' Their creators might be able to help. -RNM
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