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the chronicles of the coyotee-o and miketn khj's (klipschorn jubilee)


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Who,

I think PWK did some listener tests while moving one of the drivers back and fourth closer and farther from the listeners. This was to show that the delay through the bass horn compared to the upper range was not an issue. In the same article he confused time dealy with group dealy though.

Personally, I believe that extreme-slope networks will allow each driver to act alone in every respect. In the case where multiple drivers are intentionally run together over a wide range, taht is a different story alltogether! I think the ES network idea makes that "interferance window" so small that virtually no phasing or such compensation junk will need to be added to the netwrok at all. All this is why am hoping to work with Roy on an ES network for the Jubilee just to experiment with all this. I don't have the facilities to do it myself to any farther degree than I already have. I can only hope that I can work something in it for Klipsch to do it!

Al K.

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Al, here's the best I could do. What I've done in Picture #1 is taken a bunch of sine waves and summed them together, the brownish curve is the sum in each case (the electrical signal coming from the amplifier). All of the curves start at phase=0 at time=0. What I've imagined here is an electric bass playing a 50Hz tone, although that would be much more complex that what I have here (many more overtones), and well, I wanted to keep the math simple. Note that the 2nd harmonic is 6dB higher than the fundamental, but that's common for stringed instruments.

This is what I imagine by steady state, and I won't argue whether phase (or group delay) is audible in the steady state.

Picture #2 is the same sine waves in the same relative amplitude except I have added an exponetial decay (multiplied actually [:)]) to the signal. This is a simplification of a transient.

In Picture #3 I've added some "group delay". Literally I delayed the start of the 100Hz sine by 1.5ms and the 50Hz sine by 10ms, which I think would be reasonable for a "woofer in a box". Certainly though, the resulting curve would not look like this, but I'm keeping the math very simple. Picture #2 is what comes out of the amplifier. Picture #3 is the sound that emanates from the "woofer in a box", which is really just a high pass filter with phase shift (and therefore group delay)

The amplitudes of the various sine waves no longer have the same relation in time, so the summed signal is different. Can a person tell the difference? I don't know. I think I could, with low frequency transients, but I'd like to find out.[:)]

picture #1

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Andy,

Yes, I follow what you are showing me with those waveforms. In fact, I did quite a bit of experimenting with that sort of thing years ago as I was developing the fourier analyis section of my filter design program (PCFILT). I wrote a simple computer program that sumed up a series of sine functions multiplied by amplitude and delay factors. I was able to graphicly simulate a square wave (no dealy factors). The problem here is that it takes real-time summing to come up with the resulting waveform that will be displayed when all the harmonics of a complex waveform are observed on an oscilloscope. My experiments with the three seperate singnal generators shows that the human brain can't do that kind of summing. It's just not fast enough! The precise waveform does not exist in the human perception of the harmonics. It can only percieve that the the harmonics are there. The brain simply does not "sum" them like an oscilloscope can. I believe that the speed of the brains perception to changes is the limit beyond which the actual time delay between to harmonics are percieved as bing two different "things". That is, the classic twin taps of the old tap dancer situation in the big theater horn installation. I really think that below this limit, the brain is simply deff to the actual wave SHAPE. The delay errors are simpy phase errors the brain can't resolve. Above it, you start to percieve a "smearing". Still more delay and you realize it's two distinct sounds in time. You might say the sound "klooch" will eventually become "kaah" - "Loosch", if that makes any sense! Anyhow, the question about steady state versus one impulse in time is really not something I can relate to because I have not dreamed up any way to actually test it. Knowing that the mathematics of one cycle is simply repeated to make the steady state leads me to think it wouldn't matter. BUT: The brain isn't a calculator! You would have to demonstrate it by experiment.

Al K.

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Andy,

I think this is an example of what you are telling me. Here is a 20% duty cycle impulse with a 20 Hz rep-rate. Harmonics are included up to the 60Th. One is with no envelope distortion. The other is with the lowpass half of my ES600 extreme-slope network in the line. All the harmonics are added to the phase and amplitued scattering of the filter. The result is the typical ringing you see if you looked at it with an oscilloscope. I really don't think the brain can add up all those harmonics to make a final waveform like that, even once, as it would have to do with a single "click".

Al K.

post-2934-13819305147426_thumb.gif

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Wow Andy. You have way too much time on your hands. Thanks for posting those [:)]

Al, here's a test for ya...

Disconnect the squaker so there is no crossover overlap. Then build a device with two knobs: one for phase and one for time-delay. Set the knobs to some random position and then have a listener in the room turn the knobs until things sound "correct" - heck, don't even label the knobs. Just be sure to have some way of determining the final transfer function of the box. If it's only a matter of phase then you can expect to see knob positions all over the place. But if timing does matter then you can expect to see the same knob positions over and over. This test would probably be better performed on a 2-way system where the gap is just big enough to prevent overlap (just so the music doesn't sound funky). Probably also want to use a horn for the low-end because it acts like an acoustic filter keeping high-frequency information from coming out (which would prevent harmonic distortion from "doubling up" things).

Btw, isn't "speed" of perception associated with frequency response? And doesn't it make more sense that our ears hear the average amplitude of sound over the period of the smallest window of perception?

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Who,

"build a device with two knobs: one for phase and one for time-delay."

Let's see.. I got a 5 minute break after lunch. I think I'll do that!!! No prblem!

SORRY, BUT I'M NOT QUITE THAT SMART!

I have been trying to think of some way to test all this though. I do have a lot of signal generators that will make pulses and tone bursts. HP 3325A, Wavetek 178 and 166. I may be able to rig several of them to make two pulses or bursts that are delayed in time, one by the other. I doubt if I will ever do anything about it though.

AL K.

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