Supposing this is some griefer it seems reasonable to ignore them - but is there a possibility that this is a symptom of some kind of server attack or attempt to profile/track list members?
I've never received any unsub notices myself but it is a little disconcerting that somebody persists at doing this. I'd think that a griefer would give up after a while. E On Tue, Mar 29, 2016 at 7:13 AM, Douglas Repetto <doug...@music.columbia.edu > wrote: > I get reports about this every couple weeks. Because it's a double opt-out > no one is actually being unsubscribed from the list unless they want to be. > So please ignore these bogus unsub messages. It's not worth spending time > worrying about it. > > douglas > > > On Mon, Mar 28, 2016 at 6:37 PM, Evan Balster <e...@imitone.com> wrote: > >> This happened to me also, but I didn't give it much thought. >> >> >> On Mon, Mar 28, 2016 at 4:31 PM, robert bristow-johnson < >> r...@audioimagination.com> wrote: >> >>> >>> >>> hmmmm. i wonder if someone is trying to tell me something.... >>> >>> >>> >>> ---------------------------- Original Message >>> ---------------------------- >>> Subject: confirm a2ab2276c83b0f9c59752d823250447ab4b666 >>> From: music-dsp-requ...@music.columbia.edu >>> Date: Mon, March 28, 2016 2:31 pm >>> To: r...@audioimagination.com >>> >>> -------------------------------------------------------------------------- >>> >>> > Mailing list removal confirmation notice for mailing list music-dsp >>> > >>> > We have received a request for the removal of your email address, >>> > "r...@audioimagination.com" from the music-dsp@music.columbia.edu >>> > mailing list. To confirm that you want to be removed from this >>> > mailing list, simply reply to this message, keeping the Subject: >>> > header intact. Or visit this web page: >>> > >>> > >>> https://lists.columbia.edu/mailman/confirm/music-dsp/a2ab2276c83b0f9c59752d823250447ab4b666 >>> > >>> > >>> > Or include the following line -- and only the following line -- in a >>> > message to music-dsp-requ...@music.columbia.edu: >>> > >>> > confirm a2ab2276c83b0f9c59752d823250447ab4b666 >>> > >>> > Note that simply sending a `reply' to this message should work from >>> > most mail readers, since that usually leaves the Subject: line in the >>> > right form (additional "Re:" text in the Subject: is okay). >>> > >>> > If you do not wish to be removed from this list, please simply >>> > disregard this message. If you think you are being maliciously >>> > removed from the list, or have any other questions, send them to >>> > music-dsp-ow...@music.columbia.edu. >>> > >>> > >>> >>> i think *someone* is being a wee bit malicious. or at least a bit >>> mischievous. >>> >>> (BTW, i changed the number enough that i doubt it will work for anyone. >>> but try it, if you want.) >>> >>> >>> >>> >>> >>> ---------------------------- Original Message >>> ---------------------------- >>> Subject: Re: [music-dsp] Changing Biquad filter coefficients on-the-fly, >>> how to handle filter state? >>> From: "vadim.zavalishin" <vadim.zavalis...@native-instruments.de> >>> Date: Mon, March 28, 2016 2:20 pm >>> To: r...@audioimagination.com >>> music-dsp@music.columbia.edu >>> >>> -------------------------------------------------------------------------- >>> >>> > robert bristow-johnson писал 2016-03-28 17:57: >>> >> using the trapezoid rule to model/approximate the integrator of an >>> >> analog filter is no different than applying bilinear transform >>> >> (without compensation for frequency warping) to the same integrator. >>> >> >>> >> s^(-1) <--- T/2 * (1 + z^(-1)) / (1 - z^(-1)) >>> > >>> > This statement implies the LTI case, where the concept of the transfer >>> > function exists. >>> >>> i didn't say that. i said "applying ... to the same integrator." about >>> each individual "transfer function" that looks like "s^(-1)" >>> >>> >>> >>> > In the topic of this thread we are talking about >>> > time-varying case, this means that the transfer function concept >>> doesn't >>> > apply anymore. >>> >>> >>> >>> well, there's slow time and there's fast time. and the space between >>> the two depends on how wildly one twists the knob. while the filter >>> properties are varying, we want the thing to sound like a filter (with >>> properties that vary). there *is* a concept of frequency response (which >>> may vary). >>> >>> >>> >>> for each individual integrator you are replacing the >>> continuous-time-domain equivalent of s^(-1) with the discrete-time-domain >>> equivalent of T/2 * (1 + z^(-1)) / (1 - z^(-1)), which is the same as the >>> trapezoid rule. >>> >>> >>> >>> > Specifically, filters with identical *formal* transfer >>> > functions will behave differently and this is exactly the topic of the >>> > discussion. >>> >>> i didn't say anything about a "transfer function", until this post. i >>> am saying that the trapezoidal rule for modeling integrators is replacing >>> those integrators (which by themselves are LTI and *do* happen to have a >>> transfer function of "s^(-1)") with whatever "T/2 * (1 + z^(-1)) / (1 - >>> z^(-1))" means. and that's the same as the trapezoid rule. (the stuff >>> connected in-between might be neither L nor TI.) >>> >>> time-varying filters is the topic of the email. your VA paper along >>> with others (like Jean Laroche) speak to it. there are other LTI forms >>> than either emulating a circuit or using simple biquads. there's Hal's SVF >>> (oh, that's emulating a circuit, sorta, but Hal is doing the "Naive" >>> emulation of s^(-1)) and, what i would recommend, is the use of 2nd-order >>> Lattice (if you have floating point) or Normalized Ladder (if you're doing >>> this in fixed point) if you want to wildly modulate resonant frequency. >>> that's actually pretty commonly known in the industry. >>> >>> a "transfer function" is not enough information to fully define a system >>> unless other assumptions are made (like "observability" or >>> "controllability"). that's why the general state-variable system (Hal's >>> title is a little bit of a misnomer), ya know with the A, B, C, D matrices, >>> exists to generalize it. i'm pretty confident that any *linear* circuit >>> (but possibly time-varying) you toss up there, with either gooder or badder >>> emulation of the reactive elements, will come out as this generalized >>> state-variable system. >>> >>> but with some coefficients that can vary. like in >>> http://control.ucsd.edu/mauricio/courses/mae280a/lecture8.pdf . i >>> couldn't easily find a discrete-time version on the web. you might notice >>> that there *is* a concept of a time-variant impulse response h(t, tau) (if >>> it were LTI, h(t, tau) = h(t-tau)). it's the impulse response, h(t), >>> responding to a unit impulse applied at time tau. fix tau and you have an >>> h(t). if you have an h(t), then you also have an H(s) (or in general an >>> H(s,tau)) and i might call that a "transfer function". but it's a >>> time-varying transfer function and if it varies wildly, you can't use >>> Fourier analysis at all. but if it varies slowly enough, you can use >>> Fourier analysis, at least to the point of discussing frequency response >>> and the behavior of the system for short periods of time. >>> >>> >>> >>> -- >>> >>> r b-j r...@audioimagination.com >>> >>> "Imagination is more important than knowledge." >>> >>> >>> >>> >>> _______________________________________________ >>> dupswapdrop: music-dsp mailing list >>> music-dsp@music.columbia.edu >>> https://lists.columbia.edu/mailman/listinfo/music-dsp >>> >> >> >> _______________________________________________ >> dupswapdrop: music-dsp mailing list >> music-dsp@music.columbia.edu >> https://lists.columbia.edu/mailman/listinfo/music-dsp >> > > > _______________________________________________ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp >
_______________________________________________ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
