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Technical and scientific discussion of amps, cables and other topics.

RE: The idea that the

I think you’re on the right track, remember when CD’s were introduced, they “had to” sound better than the LP’s they were targeting to replace and an easy way was making a brighter recording that anyone would notice as “more extended”. Get a copy of Michel Hedges Breakfast in the field and hear an early a very clear but very bright recording.

Part B I think Is that harmonic distortion is normally to the high side of the fundamental producing it so in the case of loudspeakers, one finds the harder they are driven, the brighter they sound and eventually sound bad. As the size of loudspeakers has fallen and power handling climbed and lacking a standard measure of linearity or usable loudness, what we have now are speakers that are often less signal faithful than the old days.
An Altec horn for example had essentially no power compression because if you drove it that hard, the wire came off the voice coil at ~ 125-150C, now days some VC adhesive will tolerate >>350C (Rdc doubles at about 230C).
Also. I do think we “hear through” obstructions without being conscious of it and so if you remove a que that the speakers or room add, what you hear is a more faithful image while you were previously unaware of that same que.
I would offer that for the most part, if you can measure and correct your speakers based on outdoor / anechoic measurement, then you can be pretty sure that what your doing will fix both mag and phase simultaneously.

To the degree what you’re trying to eq is caused by a delayed signal (reflection) combining with the direct, you can’t really fix these as they are not a minimum phase problem. In the old days, they said only cut peaks and bumps, NEVER try to fill a sharp deep notch as that is the signature of a comb filter (caused by a delayed signal).

While the miracle of DSP and incomplete explanation or limited measurement resolution will make it appear you can “fix everything” the absolute best one can do is fix it in one specific spot where the measurement was taken and often making it worse everywhere else.

The “fixed” location is limited to about ¼ wavelength in size for the highest frequency being corrected so it is truly futile (acoustically) fixing it with dsp when the listening area is much larger than the wl in size. At 20Khz, the wave length is about 5/8 inch..
Fixing the source is the best way I think. Can’t help on a plug in but would offer that LSPcad can emulate a number of speaker controllers which have parametric’s , the down side is most dsp units are somewhat different when you call for a given filter set or alignment.
You can also listen to music through that alignment if you want. For work, I take the actual unit and measure / adjust the eq until it overlays on the transfer function or response I need.

Also, you could save the impulse response for the correction and convolve it with the music, the ”Gratisvolver” at Cat acoustics is free anyway. That also appears to be a way of transferring “what a speaker sounds like” as well as the speakers impulse response can be convolved with music too, sort of a software way of doing loudspeaker generation loss recordings we do at work.
Best,
Tom


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