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-   -   First cycle distortion - Graham, what is that? (https://www.diyaudio.com/forums/solid-state/32758-cycle-distortion-graham.html)

andy_c 22nd April 2004 08:48 PM

Quote:

Originally posted by PMA
So how about the transient response of the digital filter of the CD player, for example?(...)
Exactly.

Steve Eddy 22nd April 2004 09:10 PM

Quote:

Originally posted by PMA
The explanation hereabove is often used in the DIY community. In fact the electrodynamic speaker can be described by the attached schematics. R2 a L2 are the resistance and the inductance of the voice coil in the "braked" state (not moving). R1, L1 and C1 are the components calculated from mechanical side of the speaker to the electrical one. They describe the resonance effect of the speaker. The component values will differ according to the real speaker.
Exactly.

And you don't hear people referring to capacitors or inductors as "AC generators."

The cone's mass and its compliance are energy storage mechanisms just as inductors and capacitors are. And in fact the cone's mass and compliance have their electrical analogues in inductance and capacitance respectively, which is why dynamic loudspeaker drivers are modeled electrically using R, L and C elements.

se

SY 22nd April 2004 09:12 PM

Quote:

Originally posted by PMA
So how about the transient response of the digital filter of the CD player, for example? It has rise time (10% - 90%) no shorter than 17us and limited initial dv/dt, far below slew rate of the contemporary amplifiers. And how about analysis of the transients of the musical instruments itself? ;)
To square the circle, other sources (tape, phono) give pretty low slews also. This looks like yet one more solution in search of a problem.

Steve Eddy 22nd April 2004 09:29 PM

Quote:

Originally posted by SY
To square the circle, other sources (tape, phono) give pretty low slews also. This looks like yet one more solution in search of a problem.
Or a great business opportunity for someone looking to get into the problem creation industry. :)

se

traderbam 22nd April 2004 09:48 PM

On the subject of speaker electrical models.
The speaker has the additional complexity of being coupled to air and of drive units being physically coupled to one another through the speaker box and the air within the speaker box. Speakers are also microphones, so there will be cone movements which are temporally delayed from the original signal and these will create impedance changes as seen by the amplifier.
Also, the speaker cables form transmission lines which can have very big effects on impedance seen by the amp at HF (as Nelson knows well).
Just for completeness.

subwo1 22nd April 2004 09:59 PM

Quote:

And you don't hear people referring to capacitors or inductors as "AC generators."
That is because they aren't, but a speaker nonetheless is. The model which approximates the real thing is interesting, though.

millwood 22nd April 2004 10:10 PM

Quote:

Originally posted by subwo1


That is because they aren't, but a speaker nonetheless is. The model which approximates the real thing is interesting, though.


that's the TS model that I referred to in the original thread. Nobody including Graham picked it up, tho, :)

yeah, if you write out the differential equiations describing the electro-mechanical movements of a real speaker, you get the same differential equations describing that RLC network presented by TS.

so as far as the amp is concerned, the rlc network and its equivalent speaker are one and the same.

Steve Eddy 22nd April 2004 11:13 PM

Quote:

Originally posted by subwo1
That is because they aren't, but a speaker nonetheless is.
How is it exactly that a speaker is an AC generator but an inductor isn't?

Nelson Pass 23rd April 2004 01:21 AM

Quote:

Originally posted by PMA
http://w3.mit.edu/cheever/www/cheever_thesis.pdf

"Don't bother, it was a terrible piece of work. Amazing he got that thesis passed."

Well I thought the part where he quoted a paragraph of mine
was just great. :cool:

Also, I think there is some merit to the weighting of harmonics
in evaluating harmonic distortion. I know I'd rather listen to
1% 2nd or 3rd than 1% 4th and 5th.

"You cannot get accurate spectral analysis results with an FFT of one cycle....That is a horrible window."

You're right. I'd have to see some evidence of how you can
accurately get Spice to model THD on a waveform start that
has to be full of harmonics by itself, much less steady state.

andy_c 23rd April 2004 03:16 AM

Quote:

Originally posted by Nelson Pass
(...quoting jneutron...)"You cannot get accurate spectral analysis results with an FFT of one cycle....That is a horrible window."

You're right. I'd have to see some evidence of how you can
accurately get Spice to model THD on a waveform start that
has to be full of harmonics by itself, much less steady state.

I believe the problem that jneutron is referring to is the so-called "DFT leakage". Let's assume we have a single cycle of a truly periodic waveform. Ignore for the moment that the first cycle of GM's tone burst, when extended to be periodic, will contain discontinuities. That's because of the transient, which makes the initial value of the "cycle" different from the final value. Anyway, the DFT frequency bins are given by:

fbin= m*fs/N

where m=0, 1, 2, ...N-1 and
N is the number of DFT samples and
fs is the sampling frequency.
The fbin are the frequency "bins", that is, where the DFT is telling us the frequency components of the signal are, as opposed to where they really may be.

Suppose we choose the sampling frequency fs to be exactly N times the fundamental frequency of the periodic signal we're sampling. That is, choose:

ffund=fs/N

Reffering to "Understanding Digital Signal Processing" by Lyons, page 72, he says this:

"The DFT produces correct results only when the input data sequence contains energy precisely at the analysis frequencies given in Eq. 3-24, at integral multiples of our fundamental frequency fs/N"

Andy's note: his equation 3-24 is the same as what I've written above for the fbin.

So if we choose our sampling frequency as exactly N times the fundamental frequency of the signal we're sampling (where N is again the number of DFT points), the bins will line up exactly with the frequency components of the signal we're sampling (assuming that signal is truly periodic). If we don't establish this relationship, the signal will appear spread out in all the DFT bins in general.

How do we make this work with SPICE? SPICE wants to pick its own, often non-uniform time steps based on the time rate of change of the signal. But we can specify a minimum time step, and if that minimum time step is way smaller than what SPICE would compute on its own, we'll get uniform time steps. We can choose these time steps, together with the number of points in the FFT to make the sampling frequency fs an exact integer multiple of the fundamental frequency of the sine wave we're applying.

But the typical approach is to do the analysis on the last cycles of the sinusoid, so that any transient will have died out. Note that this is the opposite of what GM suggests. But GM's measurement really indicates the closeness of the first cycle to that of an ideal sinusoid. This isn't really distortion in the nonlinear sense. Note that I don't necessarily agree with this approach. I'm just trying to explain what I think he's doing.


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