problem 1: The availability of the zero phase shift resistor (and associated probing circuitry).
problem 2:How do current probes/hall effect current sensors compare to the zero phase shift resistors for this?
problem 3: ensuring the errors between the voltage and the current are minimized.
There are a lot of places for errors to enter.
I have a Fluke A40A 10A shunt and a Holt 1 A shunt which should be good to 100 KHz at .02%. Is that good enough?
Yup, sounds about right though the folklore keeps on going.
What you are saying is eminently reasonable. For instance, a simple standard multi-tone signal can have "infinite" different temporal envelopes even though the magnitude at each of the 32 tones is identical. Folks are too used to looking at only the FFT magnitudes, throwing the phase away loses 1/2 the information. I still run into folks that don't realize you can synchronously average in both the time and Fourier domain by vector averaging the FFT's.
It doesn't matter now that we know more about the measurement. We didn't know why the signals look they way they do before.
That could be viewed as deflection on your part. Jakob2 is correct about terminology across fields.
I can easily provide a zero b-dot resistor design, either two interleaved pass through Or a simple folded one, gave Scott one of those.
If I am looking in the 1 microsecond range, I would want my test equipment capable of 10 to 100 nanosecond accuracy, I don't know how good the shunts or halls are. I do know that the low value caddocks don't float my boat as the pickup wires pickup the shunt dB/dt.
Agreed that test design is very important here.. Also, we have to figure out exactly what we are looking for, how to proceed. I suspect we can cull eddy dragging out as a modulated third harmonic, I'll still have to go over the full dynamic and magnetic model though.
However, now that I have a mill, I can make some real resistors as I can drill the brass hole pattern using a tight drill bit for the leads, and have a pattern accuracy at the .001 inch level. I think the Seig software on the android tablet has a preprogrammed rectangular hole pattern app.
Once I solder up a resistor using some .090 brass, I can sand the bottom surface and thermally epoxy it to a heatsink, or put some mounting holes in the brass.
Jn
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WIKI: "In simple terms, timbre is what makes a particular musical sound have a different sound from another. For instance, it is the difference in sound between a guitar and a piano playing the same note at the same volume."
So you seem to be on the right track. However, in your previous statement you referred to that the "tone" was "faster" - or something like that - hence my "denier" comment. The attack is not the "note/tone". Yes - here, one must be careful with ones expressions
- not sloppy and hope for a correct(ed) interpretation.//
Yes, this is about a question I asked here https://www.diyaudio.com/forums/multi-way/338239-yes-hear-phase-differences-9.html#post5803380
Huh? This says nothing about what aspects of the sound underlie our perception of timbre differences - it simply says timbre makes one instrument sounds different to another playing the same note. Refer back to my post if you want to know what exact aspect sof the sound determine timbre.
If you could try to be exact about what I posted rather than "or something like that" before you make accusations of "Fourier denier" without being able to quote or understand what I posted
Did I say it was?
You mean this sort of sloppy - suggesting I said "something like that"??Yes - here, one must be careful with ones expressions
//
I still do not understand how the speaker cable impedance would change a phase for the speaker. Depending on speaker impedance changes with frequency, cable contribution to time shifts seems to be no more than tens of nanoseconds. How important is this, compared to all other influences that would be in 3 orders of magnitude higher??
Yes you are.
No he is not. This is not terminology, but a word salad designed to obfuscate any meaning.
There is a discipline called "signal processing" which is as much mathematics as is engineering. It is impossible to have any meaningful discussion outside of the signal processing framework, at least in terminology and basic definitions. The only major difference between analyzing a DSP and your auditory mechanisms is the investigation method; the former is analytic, the later is statistic. But the foundation, the terminology, and the definitions, are (or should be) exactly the same.
Some people seem to have an unusually sensitive obfuscation detector. Can make it easier than otherwise to result in some false positives. Might be worth considering dialing back the sensitivity a little in Jakob2's case. Some useful knowledge and insights might benefit everyone here, not just you.
Now you are starting to ask the right questions. Thank you.
If you do the model I detailed, do the right tests,(or actually believe my t-line analysis) you will see that the delays reach beyond ITD delay thresholds.
The understanding of t-line.....I am not sure if I can teach you what you would need to understand on an online forum. I have understood this for 40 years now, and am quite familiar with how difficult it is to teach this to others..
What is important is...I am not a crazy goofball audio cultist who espouses silly theories that defy physics. I am a person who is applying serious physics and e/m understanding to the problem at hand. The fact that it may not fit your preconceived notions is not my problem, but rather yours.
I welcome the dialog..and I prefer scientific discourse, not silly replies.
Jn
Agreed. Too much is lost when interdisciplinary misunderstandings rule.
Jn
Please stop me when I start down a wrong path. We apply an impulse from a zero source impedance to a pair of wires. The far end is randomly terminated and a (smaller) impulse bounces back, then a (still smaller) impulse bounces forward, etc. until the Sun cools or we get tired of waiting.
Suppose we instead make our impulse generator have the same source impedance as the pair of wires' characteristic impedance. Will it take longer for the terminating end to reach some particular level? Lump LC equivalency would seem to say yes, but a definitive answer would help my thinking a lot.
Much thanks, as always,
Chris
Suppose we instead make our impulse generator have the same source impedance as the pair of wires' characteristic impedance. Will it take longer for the terminating end to reach some particular level? Lump LC equivalency would seem to say yes, but a definitive answer would help my thinking a lot.
Much thanks, as always,
Chris
You seem to be working towards an understanding of the varying delays of various frequency (and level?) components in real audio signals, traveling through a pair of wires to a real loudspeaker, in order to discover if the variations are above detection thresholds.
But I'm just working on the basics. The whole idea of an audio signal in a wire pair "settling" is fresh to me.
Much thanks,
Chris
But I'm just working on the basics. The whole idea of an audio signal in a wire pair "settling" is fresh to me.
Much thanks,
Chris
Then the first reflection back from the load will be fully absorbed into the source/generator (neglecting complex characteristic impedance).
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Just like the case of RF and real transmission lines. An output Zobel looks more and more important in this light.
Much thanks,
Chris
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