[AliG]

I believe in bi-wire - only if the two runs are made of different cables..

My speaker cable is Acoustic Zen Double Barrel, which is bi-wire design. Does it make a difference if I just use either one run with a jumper??? The answer is "HELL YES!" aa aa

[BrianM]

Same bell, recorded how? It sounds like you are describing a mono source. You mention center the pan (singular) control.
In the case of a mono signal panned to 45 deg. off axis to the right, then I agree, you would get only amplitude differences from the L/R speakers, but no delay between L/R (assuming you are sitting in a near field listening environment, free of room reflections, or with headphones).

However, in a stereo recording, you have a L and R signal that are assigned to 2 separate channels, and panned hard R and hard L. If you record with a stereo mic configuration pointing directly forward, and you move your "bell" 45 deg, off axis, it will play back as your ears would have heard it in your first example.

I totally get what you are saying about amplitude, and delay between the L/R hearing, but in the second example, with an amplitude only change, then you are talking about a mono signal.

This is exactly what I thought.  Glad to hear I'm not (completely) crazy.

[PhilNYC]

I spent a year with my system using bi-wiring...on a whim, I removed it, and I found that I far preferred the sound without bi-wiring in my setup: more definition, weight, speed (particularly in the bass).

I have no idea if the technical description given in this article is accurate or not, but the description of the effect of bi-wiring on the sound matches my experience exactly:

http://www.sonicdesign.se/biwire.html

[jneutron]

Move the bell to 45 degrees off axis, to the right.  Ring again.

You point to the bell, and again, you made that decision because your right ear heard a louder signal than your left, and, your right ear heard it first.

Amplitude is much stronger at the right ear, and the delay is much shorter at the right ear, also more direct sound vs radiated sound is heard.

Much stronger is of course, relative.  For a spherical wavefront, the difference is actually quite small.  In fact, shifting a spherical wavefront source 10 feet away off axis produces a ear to ear change of  .001dB per inch.  Simply put, moving it a foot will cause .012 dB difference between ears (yes, I know it's really an integration, but those details are beyond this discussion).  Who has the ability to measure that difference with electronics, nevermind microphones???  (Note:  spherical produces the most delta ear to ear...point source, one driver..a line array cylindrical wavefront produces less, a planar....none.  speakingl nearfield)

Put another way..if somebody makes a change to their stereo that causes a movement of a virtual image one foot off center, who on this planet can measure it???

Shorter delay?  Speed of sound is about 1 foot per millisecond.  On axis, the time delay ear to ear at 10 feet is 1.8 usec per 1/4 inch (sorry, my graph is setup with those units)..  To move one inch is a timeshift of 7.2 uSec.  A 40 inches move (from graph) is about 120 uSec.  At 40 inches to one side, a shift of 1 inch is 3.2 uSec, a foot, 38.4 uSec.

Measurement of those kind of numbers using speakers and microphone??  Forget it.

Don't forget what those numbers represent in distance to the drivers, either.  83 uSec per inch in air.  In terms of absolutes, that's just nuts..   That is why I early on abandoned absolute imaging, as that requires a head in vice approach.  Thinking in terms of one image relative to another image is the way to approach the topic.

The density of early and late reflections, and the associated..................

Whoa, wait, stop...

You are introducing far too many confounders.  I am only a simple minded individual (a simple farm boy) , and have to address the problem in really simple terms..

The first part of the problem is to identify the levels of the relevant parameters.  First, consider the problem from an anechoic point of view, as this is essentially how us hunters evolved to find prey...we rarely hunt our prey in a 12 by 12 room from a listening chair..(of course, the brazilian women's vollyball team does come to mind here  ).


The rest is certainly important, but confounds the math a "tad".

Cheers, John

[jneutron]

I believe in bi-wire - only if the two runs are made of different cables..

My speaker cable is Acoustic Zen Double Barrel, which is bi-wire design. Does it make a difference if I just use either one run with a jumper??? The answer is "HELL YES!" aa aa

All anecdotal accounts such as yours, must of course be taken with the "grain of salt", simply because it was an experiment which is confounded by expectation bias, you knew of the change, and could have been biased to think it made a difference.  I do not believe DBT's are well designed for localization discrimination, so also consider null DBT's as suspect as well. I always use caution with both possibilities.

For your experience, you altered all three parameters, LCR, so that is a confounder as well as your knowledge that a change was made.  That blurs the "smoking gun".

But what is important is your perception.  Glad you are having fun..

Cheers, John

[Ethan Winer]

You are refusing to perform a specific measurement simply because of a belief, because you "know".

No, it's because I don't have time to do this. It would take me at least an entire day to disassemble a loudspeaker and tap into the crossover, disconnect one of my big Crown power amps from my other rig, hook up test gear and a signal source, etc. And for what benefit to me?

1. I am saying that the differences will be on the order of the ratio of delivery system resistance to end load resistance.

Differences in what?! Noise? Frequency response? Signal level? Distortion?

2. Audible improvement...I've heard an effect using a contrived not real world test, and am concerned with it possibly being something that is indeed audible.

I can't see how that's possible. My standard reply is it must be comb filtering. In the mean time, please describe your contrived test.

3.  Standard DBT practices fall short ...

Forgive me for stopping you right there. Attacking DBT because it doesn't support one's theories is not a valid defense IMO.

... when it comes to testing of relative localization

As I said, comb filtering.

--Ethan

[jneutron]

If you guys wanna have some fun, I could take some stereo samples, and shift one side of a stereo recording by amounts as small as one sample compared to the other side. If there is any way for the results of that to be analyzed to see if there is anything that can be measured by conventional instruments, hey, then we could have some fun. 
Cheers

I have in my office, a comp setup for listening.  Headphones, as the guys here may not appreciate Inna gadda da vita, Vivaldi, or Darude..

I have a severe imaging problem with the setup...I tracked it down, but it took time...

The output D/A is mux'd...one DAC, two output channels.  They did not restore the interchannel time sync, but allow the held samples to proceed to the analog output.  As a result, there is a built in ~6 uSec delay between channels, the left one is 6 uSec behind the right.

On headphones, that show up as a shift-to-right of all centered images. To alleviate this, I can use the pan control.  Unfortunately, the brain's ability to interpret this shift is frequency dependent.  In other words, by altering the relative amplitude between the channels, I can center some part of the image, but not all.  This results in a lateral spread of the image (using headphones, it is called lateralization), no longer a tightly focussed one.  It blurs, so to speak.  If I were concerned with getting it right, I'd be horribly dissapointed.

In addition, some of my source material was converted using a mux'd input scheme as well.  Sometimes that produces even more shift.  To get adequate centering in that case requires the left channel being turned up waaaay to much to keep a good soundstage.

Cheers, John

[jneutron]

You are refusing to perform a specific measurement simply because of a belief, because you "know".

No, it's because I don't have time to do this. It would take me at least an entire day to disassemble a loudspeaker and tap into the crossover, disconnect one of my big Crown power amps from my other rig, hook up test gear and a signal source, etc. And for what benefit to me?

If you achieved positive results, and published.. you figure it out...

Time is what you make of it.  I also have time constraints, as I also focus on the important tasks in life.

1. I am saying that the differences will be on the order of the ratio of delivery system resistance to end load resistance.

Differences in what?! Noise? Frequency response? Signal level? Distortion?

Why would you say noise?? 

Frequency response as measured conventionally????  Signal level as measured conventionally?  Distortion as measured conventially??

Ya gotta back up, and understand the analysis, what the differences are, and what the caveats are for conventional test setups.  The test I provided looks ONLY for the difference, and does so in a very unique and clear method without confounders.

2. Audible improvement...I've heard an effect using a contrived not real world test, and am concerned with it possibly being something that is indeed audible.

I can't see how that's possible. My standard reply is it must be comb filtering. In the mean time, please describe your contrived test.

Two speakers, each two way, each single pole.

Replace the woofs with resistors.

Feed both channels hf content (I used sine, but decent vocal could be used.)

Feed one channel lf content that is sufficiently lf to eliminate feedthrough to the tweeter, but insufficient to take the amp into nonlinearity or output quadrant vs damping factor problems. (while the resistor eliminates energy return of the woof, it still forces some quadrant 2 and 4 operation by virtue of the crossover inductance).

Listen.  If the lf causes the image to vary, to waver  (to modulate...I cannot provide a better hint than that aa)..then you have two choices..

1.  Ignore it.

2.  Figure it out.

I chose "behind door number 2".

That you don't desire to take the time to choose door number 2, is your decision, and yours alone...I can certainly respect that decision.  I cannot, however, respect the decision to claim something is not correct even though I don't wanna try it  "because"..

3.  Standard DBT practices fall short ...

Forgive me for stopping you right there. Attacking DBT because it doesn't support one's theories is not a valid defense IMO.

Silly.  I have not "attacked" DBT.  Please discontinue strawman arguments.

I have pointed out a science which shows that humans adapt to varied localization stimulus, and have clearly stated that the amount of time humans take to re-adjust to a new interpretation of that stimulus has not been considered in standard DBT protocol.

I do this in my chosen field as well, when I embark on a measurement, I must identify the entity being measured, the expected level of the entity, and most importantly (you forgot this, didn't you)  WHAT IS THE CAPABILITY OF THE EQUIPMENT TO MEASURE THE ENTITY???

How does one measure localization based imaging on a stereo system via DBT, when the stimulus (program material) contains HALF of the content humans use to localize a source in space???

HALF...not 99%, not 80%, not 60%...HALF.

Now what were you saying about DBT's being the cat's meow??  Would you use one test lead on your DVM??

... when it comes to testing of relative localization

As I said, comb filtering.

--Ethan

Comb filtering...chuckle..

If you wish to learn about localization theory, I can send you some links.

BTW, your chosen field of endeavor, that of removal or control of reflective "confounders", is entirely valid.  I would never consider your product as useless, but rather necessary.  But it is clear to me that you are attempting to use "after the fact" science (re:comb filtering) to invalidate control of the initial generation of the soundstage image.  To me, that is akin to running behind an arsonist with a fire extinguisher, and worrying about improving the fire extinguisher.   What I speak of compliments your work, it does not negate it.

Cheers, John

[jneutron]

I have no idea if the technical description given in this article is accurate or not, but the description of the effect of bi-wiring on the sound matches my experience exactly:

http://www.sonicdesign.se/biwire.html

Thanks for the link.

It was going well, up to the point where I read that the graphs were simulations...

I was hoping for actual measurements.

However, note that a 3 degree shift  of a 3 khz wave is :

 1/3000 = 3.33e-4, 333 uSec times 3/360 = 2.7 uSec.

Note that this exceeds Nordmark's measurements of human capability, he measured 1.2uSec.  So even a simple analysis starts to get the timing shifts into a plausible range of audibility thresholds (not necessarily deemed as audible, but not orders of magnitude below thresholds.)

What would be more interesting would be the values of the drop in dB...given the extreme sensitivity of humans to ITD changes, with concurrent signals branching at the crossover.

That is not understood, nor even considered, in any analysis or test.

Cheers, John