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Old 7th September 2007, 08:09 PM   #1901
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Quote:
Originally posted by Henckel
Since no speaker has a flat response down to DC ( 0 Hz) the impulse response has to have as much(Area) below zero as above zero.
Correct.

There's a good point about the energy allocation of an impulse - the entire spectrum of the signal is tilted towards HF, thus what you see is mostly HF artifacts. The reverse is true of a step function, which has a spectrum tilted towards LF, so you mostly see the highpass function of the woofer (which is difficult to measure except in nearfield).

Also, like an oscilloscope trace, you can really only see about 30~40 dB of visual resolution, thanks to the linear display. That's one area where a CSD is more useful, since the display is logarithmic in the amplitude and frequency domains. I usually use a CSD and impulse display together, to get a sense of where the worst time-domain faults lie.

Still, I very much like the absence of smoothing in the impulse response - it seems like I always get a million reasons from speaker designers why I'm not intelligent enough to interpret anything but a "smoothed" display - which of course removes all of the bad parts of the measurement technique and small-scale faults in the loudspeaker. If there's a resonance at 2, 4, 8, or 16 kHz, well, I'd like to see it, not have it "smoothed" away.

Maybe I'm a little oversensitive, but I can usually hear problems with HF resonances, even ones I'm not supposed to be able to hear. That's probably why I didn't much care for the first CD players and their discs, which had far more severe HF ringing in the time domain than a good-quality tweeter of the day (back in 1982, when tweeters weren't very good). If a tweeter (typically a metal-dome) has problems around 22~30 kHz, even though I'm not supposed to be able to hear it, I usually don't like it - it'll sound "rough" and kind of sandpapery to me. Thus, I like to measure beyond 20 kHz, and use lowpass fliters that don't degrade the time domain with ringing artifacts.

The "rumbling" quality of some of these horns - Meyersound included - is rather striking. It's all too obvious the air-load of the horn is not damping the diaphragm, contrary to the claims of JBL over many decades.
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Old 7th September 2007, 09:19 PM   #1902
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The audibility of ultrasonic resonances (or high-Q digital-filter artifacts) deserves more discussion. Obviously, if there is no program content to excite the resonance in the first place, it can't be audible - or would it?

Reading Zaph's Web page, I thought that if the 3rd, 5th, or 7th harmonic lands on an ultrasonic resonance, there could be consequences lower down in the spectrum - maybe a very narrowband rise in distortion, only occasionally stimulated by certain types of program material, but lurking in the background all the time.

I suspect there are substantial differences in audibility of narrowband resonances depending on the auditioner. Horns and low-resolution digital systems (like Red Book) are well-known for high-Q narrowband artifacts - and as far as I can tell, quite a few people don't hear these artifacts at all, and others very strongly object and don't like them.

I couldn't accept first-general CD's at all, and didn't buy a player until the Philips 4X oversampling technology came out - even then, I always kind of thought of it as being a more durable alternative to prerecorded Dolby cassettes, but not a truly hifi medium. When the Apogee UV22 dithering and sample-rate-conversion technique became more widespread in the early Nineties, CD's started to sound more acceptable-sounding, although still not as good as my late-Seventies vintage phonograph. But records were becoming harder and harder to find, and I reluctantly made my peace with the medium, although never really liking the sound all that much.

When I first heard really high-quality 96/24 PCM demonstrated by Keith Johnson (who also made the recordings) at the CES and VSAC, I was surprised how little it sounded like 44.1/16 and how close it came to analog mastertapes (also demonstrated by Keith Johnson on Ampex studio recorders). The sound we had all accepted as "digital" was nothing more than the artifacts of a rather poorly implemented low-resolution PCM system - exactly the same thing that was said at the birth of Compact Disc in 1982. When you flip the switch for yourself, all doubt is removed - Red Book is a poorly designed system that never had "mastertape sound" in the first place - and still doesn't, 25 years later.

What I find even more surprising is some people can't hear the difference between 96/24 and 44.1/16 PCM, which I find little short of astonishing. To me, it's like the difference between Dolby-B Compact Cassette and a 15IPS mastertape - obvious and impossible to ignore. The low resolution is one thing - not desirable, but kinda-sorta tolerable - but the metallic-sounding artifacts in the 5~10 kHz region are much harder to accept.

This particular artifact is frequently the result of lazy do-it-by-the-applications-handbook analog-circuit design following the DAC converter chip - and here's just one example: analog-stage op-amps that are too slow by not one, but many orders of magnitude.

Matt Kamna and I measured slew rates well in excess of 1000V/uSec coming out of the Burr-Brown PCM63 converter, along with a flat RF comb spectrum going past 20 MHz - and visible to 50 MHz. But - the common op-amps in even very expensive (more than US$5000) players typically have no more than a 13V/uSec slew rate (Philips/Signetics 5534, 5532, and its successors).

Slewing in the lowpass filter results in the sample interval being the incorrect time duration, which then translates into HF distortion in the audio band. Sampling theory requires that the time duration of the individual samples remains precisely the same as the original sampling operation, otherwise nonlinear distortion will result when the signal is reconstructed. The severity of the digital-reconstruction distortion is a function of the loudness and HF content of the signal (a larger proportion of the signal is slewed, and improperly reconstructed by the lowpass filter).

Again, we come back to HF artifacts, which some people are not bothered by, and others object strongly. They bother me, and since I don't have to report to a boss or the AES, I can design for my own tastes, not a committee.
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Old 7th September 2007, 10:27 PM   #1903
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Default Two Measurements, Two Results

Here's my MLSSA Quad ESL57 measurement and the one posted upstream. This probably shows the value of the ultralow diffraction Spurlock Stand - much cleaner decay performance. I think this demonstrates the difference that measurement protocol can make - absence of ringing in the lowpass filtering (analog or digital, software or hardware), absence of peaking in the measurement microphone, and a low-diffraction stand. If you can't see it in the measurement system, what chance do you have of fixing it in the loudspeaker?

Then again, if the other measurement is a Quad ESL63, that's a sad comment on how much it went downhill. But I suspect measurement artifacts instead.
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Old 7th September 2007, 10:36 PM   #1904
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I though I'd chime in on the impulse response since the impulse of one of my older designs popped up. The system had a low frequency cut off of 25 Hz and was flat out past 20 k Hz. Now if is true that any system that is not flat to DC must have equal area above and below the zero axis. BUT the positive area is that contained under the impulse (finite width). The area below the zero axis isn't always very obvious (as in my impulse) because if represents the decay of the energy of the impulse back to the equilibrium state. With a low, low frequency cut off this takes the form of a very long, and very low amplitude decay (talking in a simplified sense here). So while the area above and below the zero axis are the same, the area above is high amplitude, short time duration (area = A x DT) and the area below has amplitude much, much less than the positive impulse amplitude with a much, much longer duration. Thus, for systems with very low cut off frequency the negative area is not very apparent and can be of such low amplitude that it is below the noise floor. Thus in my impulse the negative swing is not clearly visible at all. What the impulse does tell you is that there is no crossover induced GD in the system. The system is minimum phase.
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Old 7th September 2007, 10:59 PM   #1905
mige0 is offline mige0  Austria
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Hi

Quote:
What the impulse does tell you is that there is no crossover induced GD in the system. The system is minimum phase.
Very clean impulse response for three speakers working together in the plot of post #1898 .
Could we have a x-axes scale please?

Greetings
Michael
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Old 7th September 2007, 11:01 PM   #1906
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Quote:
Originally posted by john k...
I though I'd chime in on the impulse response...
John,

Thank you for the lucid explanation. This is what I believed to be the case.

BTW, I am glad to see that the ICTA system is looking like it will see the light-of-day

Edward

P.S. I would have posted one of my transient-perfect impulse responses, but they were not available to me at the time - so I 'borrowed' the one that I remembered from your orginal "B&O concept" system.
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Old 7th September 2007, 11:12 PM   #1907
Salas is online now Salas  Greece
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Default Re: Two Measurements, Two Results

Quote:
Originally posted by Lynn Olson
Then again, if the other measurement is a Quad ESL63, that's a sad comment on how much it went downhill. But I suspect measurement artifacts instead.
Don't know if it has hardware issues, its from Stereophile. Its ESL63 for sure. Ii just saw that it went visually negative also. ESL63 is widely regarded as time coherent and fast. I posted it to help stimulate commenting on the impulse questions.
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Old 8th September 2007, 12:19 AM   #1908
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I have found another one ESL 63 IR. Maybe it interests Lynn. I got it from here.
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File Type: jpg quad impcoranduncorr.jpg (95.0 KB, 997 views)
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Old 8th September 2007, 10:50 AM   #1909
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Quote:
Originally posted by mige0
Hi



Very clean impulse response for three speakers working together in the plot of post #1898 .
Could we have a x-axes scale please?

Greetings
Michael

That plot, as I recall, was the first 3 msec of the impulse. Nothing much to see after that. Here is the reproduction of a 2k Hz square wave by that speaker.

Click the image to open in full size.

And at 500 Hz

Click the image to open in full size.
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Old 8th September 2007, 04:28 PM   #1910
gedlee is offline gedlee  United States
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Quote:
Originally posted by Henckel
Since no speaker has a flat response down to DC ( 0 Hz) the impulse response has to have as much(Area) below zero as above zero.
This is your answer.

No acoustic source can create a static change in the atmospheric pressure (not even the space shuttle when it enters the atmosphere - it creates a doublet), hence any and all acoustic signals (impulse response or whatever) must average to zero. If it doesn't then its a fake or it didn't come from an acoustic source unmodified by something electronic - or a pencil!
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