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so, i created a 3ms time delay convolution (did this at 400hz for the scala)


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As far as spurious double-taps go, the only records I can think of that I own that have clicking heels are Flamenco guitar ones. And the taps of the Spanish dancers are mighty fast and of course there's lots of echo. Hope this helps.

You mean you didn't hear Carlos Santana doing his double-tap dance when I listened to my new Horns earlier? He doesn't just play guitar. You can hear it only if you're a "true[:P]" audiophile.

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Getting back to my prior thread, Al, I tried to make sense of your tables, but I can't decipher them entirely.

But, as to my thread re: the gaps and bleedovers, would this be just matters that pertain only to amplitude (volume) for the given tone represented? If so, I can't imagine how you could pick up amplitude variations that close together. As to harmonies, the gaps and bleedovers would be delays between different instruments and between notes on the same instrument - and assuming there's no sharing of the tone between the mid and woofer. That's a matter that I also think would not be capable of perception.

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It's been real fun, but maybe this will end this debate. Suppose you're listening to stereo from an equidistant point from both. Simply stepping aside 2 steps would accomplish this same argued phase-shift - and thus a "whole new timbre" as it would be argued. If the difference is the same as moving away from the center a couple of steps, big deal!

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There is delay between the squawker and woofer, and then you also need to account for the tweeter -- so triamping using three stereo or six mono amplifiers -- and active crossovers. A cost effective way of doing it would be to use chip amps -- which is why Shawn has been gobbling up all the TEACs going up for sale on the forum. I think I'll wait for his final analysis before deciding if it's worth the trouble or not.

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Dean,

" -- which is why Shawn has been gobbling up all the TEACs going up for sale on the forum. I think I'll wait for his final analysis before deciding if it's worth the trouble or not. "

It is a fair amount of trouble if you are trying to do a good job of optimizing the whole system. Without at the very least a good RTA/mic combo (of varying resolution) I would strongly suggest against anyone taking this approach. And for more indepth analysis time based measurements are needed to test things like the delay between drivers and possibly phasing between drivers. FFT software would be good for finer resolution too for EQing and checking blend at the crossover points. A fair amount of patience and a good amount of understanding the whole setup is important as well.

Having said that I am very happy with the results I'm getting so far. I made a fair number of changes at the same time (tri-amping, adding the JBL tweeter, EQ) so I can't really say how much of the difference is due to which piece of the puzzle... they all added something to the end result.

They system is very listenable as it is, better I think then it ever has been, but I'm not done with what I have been planing on doing yet.

I installed the autoformers on the tweeters yesterday to cut down on their hiss level and ended up using about 10dB of attenuation to really get them quiet. Now the hiss from the mids is louder but they are still pretty quiet. At the same time I took about 10dB of attenuation out of the tweeter amp channels inputs to rebalance the system. Since I have 30 watt...ish... on the tweeters I have power to spare to do things like the attenuation. I do want to see how the Teac reacts to the impedance of the autoformer/tweeter so I will have to measure that at some point.

I haven't verified the time alignment settings that the Behringer measured. I'm not going to do that till I have everything else I want done in case changes in the EQ and such effect latency through the crossover.

My only other planned changes at this point is to lower the crossover from 600hz to something lower. The mid is flat down to about 350hz but I don't think I'm going to push it that low. I have a serious room null right around 500hz so I am thinking I'll center the crossover point right at the null. That in effect may act to make my already very steep crossovers (210dB/octave) even steeper acoustically.

Second, the mid/tweeter crossover is currently only at 48db/octave because I ran out of DSP resources in the crossovers. I can hear and measure the interference between the two as you move around. To go to a very high slope there I need to dedicate one crossover per speaker. Currently they are configured for stereo three way operation. If I set them up for mono I'll have twice the resource available and should have plenty available to do a high slope crossover to the tweeter.

I need to work out the settings for the high slope crossovers before I can do either of the above.

After I do that I'll work on verifying the time alignment/phase between the drivers.

The other thing I might do is move the subwoofer crossovers into the Behringers. If I do that I could go considerably sharper then the fourth order they are currently running.

If you ever happen to be out my way you are welcome to come hear the setup if you like.

Shawn

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All this fretting over 7 millionths of a second. I'll be kind and just not say anything more.

7 thousandths. But point well taken. That's the point I am making - that 7 thousandths doesn't matter. But I like to hear the theory and to hear somebody try to convince me it does... does.. does... does.

Oh come on Jeff you just like a lengthy some what warm debate. Who you kidding it could be about toilet paper[;)]

Craig

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All this fretting over 7 millionths of a second. I'll be kind and just not say anything more.

7 thousandths. But point well taken. That's the point I am making - that 7 thousandths doesn't matter. But I like to hear the theory and to hear somebody try to convince me it does... does.. does... does.

Oh come on Jeff you just like a lengthy some what warm debate. Who you kidding it could be about toilet paper[;)]

Craig

Okay, you caught me, Craig... But I didn't see the thread on toilet paper........................................... my poor Astros![:(]

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"does your processing allow you to easily switch the time delay on and off?"

Should be, I believe you can just turn off the delay correction in the crossover. Otherwise you could just set all the delays to 0.

Come to think of it if I ever wanted to check the audibility of that I can likely make that really slick. I could save everything with the delay ON to PRESET 1 in the crossover. Then disable the delay and save that to PRESET 2.

Then using a gizmo I build use that so that when I select one input on my pre-amp (by remote) the crossovers switch to Preset 1 and selecting another input (both configured for the same source) it would make the crossovers switch to Preset 2.

"And aren't you running "lascalas"?"

Only from about 80hz to currently 600hz. Below that is a pair of subs with a pair of JBL Sub1500s in each. Above 600hz is Altec 288-8k on 805B horns, above 8k is JBL 2404Hs.

Shawn

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First, be warned that plots and samples made with digital equipment may not be the way to go on this. If what you are arguing about is 7 millionths of a second, compare this to the sampling rate and plot resolution. He he.

Second, adding two different frequencies (playing them at the same time) does not make a third complex frequency - it makes a third complex wave, which is not the same thing. There are still just two frequencies - Have you considered that the sum and difference frequencies might not exit in the free air - they only come into existence when measured by a transducer (or heard within the ear)? Ha ha.

If there is a rational lawful external world, any well formed inquiry may be subjected to test. What I see here for page on page are unfounded conjectures, suppositions, assumptions, guesses, half-baked ideas, asertions, elocutions, pet theories, and psudoscience. There may even be some of that in this post! Ho ho.

Paul

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1ms (.001 seconds = thousandths, not millionths) is a lot larger than

the time it takes for a 20kHz sine wave to complete one cycle (0.00005

seconds), which has no problem fitting onto the medium or even being

audible by the human ear. In fact, .001 seconds corresponds to a

frequency of 1kHz.

Btw, the sum and difference frequencies are physically there in the

air. I believe it was sfogg who did some measurements where he recorded

two tones (each playing from a different speaker). He then implemented

an EQ that completely eliminated one of the tones, yet the two

combination tones (and the other original wave) were still present. At the time I was arguing that

perhaps one of the reasons why a 96kHz sampling frequency sounds better

than 44.1kHz sample could be due to the absence of the combination

tones provided by say a 30kHz frequency (which would imply the

combination tones are created when the sound hits the ear). But alas

this is not the case as the combination tones are physically created in

the air. (And let's not confuse this with the fact that combination

tones are still created when the speaker plays back the source

material).

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So, my belief is this delay phenomenon is interesting in terms of picturing graphs and pondering how the "obvious picture" might register in terms of the quality of the audible sound. But I think our brains process sound based upon what the focus of attention is. Ever been watching a news story, and your wife says something, but it doesn't register clearly? Ever been talking to your wife while a news story is on, and the news story did not register clearly? These are the EXACT same sounds, only you heard them qualitatively differently.

The brain is amazing how it can filter out extraneous noise. I find it interesting that if we recorded that same conversation on a tape machine, with the television in the background, we cannot process that information in the same way. When playing this recording back, all we hear is a mish mash of conversation and television, blended together so we can hardly make out either.

What is missing of course, are the audio and visual clues that allows our brain to engage its massive processing power.

While this talk of a miniscule time delay convolution has some anecdotal interest, I can't see such minor considerations having a major bearing on the way loudspeakers reproduce the music. Take a so-called time delay corrected or time and phase-coherent loudspeaker such as say, the Thiel CS2.4. Nice enough loudspeaker, but it doesn't really appear to offer anything special over the competition.

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Guys,

For anybody who read this once, READ IT ONE MORE TIME! .....

Al K.

................

First, be warned that plots and samples made with digital equipment may not be the way to go on this. If what you are arguing about is 7 millionths of a second, compare this to the sampling rate and plot resolution. He he.

Second, adding two different frequencies (playing them at the same time) does not make a third complex frequency - it makes a third complex wave, which is not the same thing. There are still just two frequencies - Have you considered that the sum and difference frequencies might not exit in the free air - they only come into existence when measured by a transducer (or heard within the ear)? Ha ha.

If there is a rational lawful external world, any well formed inquiry may be subjected to test. What I see here for page on page are unfounded conjectures, suppositions, assumptions, guesses, half-baked ideas, asertions, elocutions, pet theories, and psudoscience. There may even be some of that in this post! Ho ho.

Paul

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A couple of thoughts here as we meander about not sure of exactly what we are trying to prove.

First, depending upon how you listen to this, I am curious as to how you account for the phase delay, non-linear latency, etc.!! inherent in your processing and playback system... Isn't science fun!?

Second, and much more significantly, may I suggest you go back and simply review classic waveform superposition! I am amused at the conjecture of how combining the waveforms will be perceived! And just a hint...If you listen to a complex waveform, and then perform FFT (the Fourier transform) on it to identify the component frequencies and their relative levels, do you hear this as 'x' number of separate discreet frequencies? I will leave you to ponder this!

You folks have fallen into the equivalent of Descartes' mind-body problem in audio. And I dare say the majority of the problem is in the framing of the question! It is all well and nice to take a totality and analyze it into its component attributes/characteristics (parts), but like Descartes' mind and body, because you have conceptually labeled them and given them names, do they actually have a life independently of each other? - in other words, as you can now put the 'mind' over here, and the 'body' over there, and explore the attributes of them in isolation, this is all well and good 'conceptually, but how does this relate to 'reality'? Do they actually exist in this manner? Words are great as tools, but at what point do your words cease to 'know' your subject!? Do you have a reality where the mind is radically separate from the body!? (And now that everything is broken into radically separate components - after the 'great fall' - 'how do you put Humpty Dumpty back together again'? ...and if you haven't already realized it, these are all apt metaphors!) At what point does your model take precedence over the integrated whole of reality that you are pretending to explore?

Thirdly, if you ever succeed in doing what it is that you are trying to do with a 3 ms delay, you are going to confirm what is commonly referred to as the Haas effect, but which properly was documented in depth by Faraday more then a hundred years before. Funny how the ancients keep stealing our discoveries!!

Delayed sounds are integrated by the auditory apparatus if they reach the ear within 25 to 35 ms of the direct sound. The level of the delayed components will contribute to the apparent level of the sound, and there may be an apparent change in character. Thus the discrete sounds will seem to fuse, and the first signal will assume precedence.

The bigger question that will remain unanswered here is, although our ears do not process the signals as discrete signals, radically separate from each other, does that mean that the smearing is meaningless. In other words, does complex behavior within this time differential not matter simply because we do not hear the discreet events separately - while we do hear the gross changes resulting form changes in the particular components? And because we do not process them as discrete events, does that mean that the events within that interval are dismissible as meaningless as so many are so quick to do? **

Many complain about the disconnect between perception and the measurement process, and then because they cannot resolve the detail at a particular level, they define the behavior meaningless! Were life so simple!

And you also have quickly discovered that it would indeed be nice if you had some sort of instrument that was capable of viewing complex phenomena in the time domain!

And even more fundamentally, you are encountering the fact that the traditional frames of reference for accounting for both the perception and the measurement of this perception are limited!

The interesting thing - in a weird sort of way [:P]- is that some of you have focused on the frequency domain to attempt to describe the behavior, while others focus on the time domain. The significance is that you are, in a tacit way, acknowledging that you are looking at the same phenomena from two different frames of reference - you are not looking at two radically separate 'things'! And that only one frame of reference is not sufficient to describe and account for all of the characteristics! Well, you will be interested to know that there are still more frames of reference! (But you won't appreciate these just yet!)

What you are encountering is the loose equivalent of folks saying that since bacteria are too small to see, or that since molecules and atoms are too small to see, that all behavior smaller then that which is visible to the naked eye is meaningless. And there will be those of you who will look at one small aspect and because of no world shattering results declare all to be meaningless. The only variable here is not if, but rather how many will do this!

If you want to persist in this belief, be my guest - as you drink your raw sewer water while insisting that there is no difference between that and your dead filtered water. Besides, do THEY taste the same? And if they do taste the same, does that mean they are the same? Hmmm?

So, when you complete your exploration - and I do hope you continue to wrestle with this, I hope it results in the generation of more lingering questions rather then some overly reductionist "answer"! And perhaps you will be intrigued enough to pursue the study further.

And perhaps, just perhaps, someone will wonder why, with all of the great explorers in acoustics and audio, only one person has had their collected works published by the Audio Engineering Society (AES)! And perhaps someone will wonder just what it is that makes those works so special!

An excellent introduction and reference to the mechanics of audio and acoustics is found in Davis' Sound System Engineering - as let's face it, there is no way around understanding the underlying mechanics of sound!!! And then, you might want to pursue an introduction to Heyser's works, which attempt to radically reframe the traditional viewpoint regarding acoustics and things audio - radically shaking up the process leaving neither the traditional objective measurement group nor the traditional subjectivist camp unscathed! As his goal is to bring the objective measured reality subjective closer to the perceived reality - in a way that will make the measured reality correlate with the subjective experience! And with that, you have just entered the world through the looking glass - if you will! Just as quantum mechanics completely reordered the world of classical physics in a way that classical physics' rules only hold for Big things going Slowly! Heyser reorders our conceptualization of the world so that a new paradigm is brought to bear on the fundamental relationships of audio and acoustic phenomena!


I wonder how long it takes for some to venture and risk the look...and I wonder even more how many will persist in actually pursuing the knowledge further to where it will assume some meaning beyond the stage of simply identifying many anomalies not easily accounted for in classical acoustics/audio and pursue it to a point of meaning!?

I wonder![:D]


** And while i will not attempt to address this here in any depth - doesn't the processing of two discreet signals into a smeared signal with only a few apparent character changes constitute a form of distortion? Funny, if this were electronics I am confident that many here who dismiss this would be running about claiming any number of names for the newly discovered distortion! Anyone wonder if this might play a role in overall perception? After all, I can't see nor easily hear many forms of electronic distortion, and 90% (you pick the number!) of equipment has distortion at levels below what I can easily distinguish, so the topic must be simply nonsense! And all equipment must therefore sound the same! Or is there something for which the current system does not account!? Hmmm? I know!!! Let's just declare it a non-issue and ignore it! Sound familiar?

Seems to me that folks should be asking allot more questions rather then declaring everything a none issue!

>And if and when some choose to start exploring Heyser's work, I will be glad to post showing the relationships between the various domains - showing the logical relationships as well as a graphical representation of how they all relate.(map to each other). You will then see the value and significance of the Nyquist & Heyser displays!

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