Sloped baffle

Bombaywalla,

The DSP signal processing is touching your music signal in a very fundamental way in that the entire music signal has to go thru the DSP before you can hear it. Same deal with the passive x-over. But the difference is that you change the quality of the cap or the inductor or the hook-up wire & you can change the sound to your liking. It appears that it's not that simple with the DSP software - you cant go in there & change the code. Or, maybe I'm not thinking of this correctly?

I think you have it right; at least from my perspective. However, I don't have the skills to change caps or wire...so I'm basically stuck with what I get. Actually, software provides more flexibility here, in my case.

So, if I'm envisioning this correctly - computer digital out runs into the DSP software which breaks the audio signal into highs, mids, bass. You get 3 digital streams now. You feed these 3 streams into 3 identical DACs or 1 Pro quality DAC able to output 3 analog streams (one box would be better as everything sits in 1 chassis & has a better chance of being matched to the other analog stream). Then 3 analog streams into 3 power amps - you need to match these very well too: same input sensitivity, same gain, same sort of clipping algorithm, same dynamic headroom extension, same power output capability, same current source/sink capability.

Here I would say, not exactly. Inside the computer being used as audio server the DSP software runs as well. On the DSP software (eg, Acourate) you set up XO frequencies, slopes, delays, etc, and perform the driver measurements, do the adjustments, etc, perform digital room correction, and eventually get a sort of digital filter. Then you apply this through a convolver to the audio player software (eg, JRMC). Now the computer is outputting through USB several channels. Eight in my case/plan. A multichannel DAC, such as the exaSound e28 takes USB in and decodes into the 8 channels and outputs 8 analog signals. Simple 1-box solution!

Also the amps don't need to be identical. You adjust gain at the software level. Take a look at the article by Mitchco I linked before. It's an easy read and provides a nice view of his setup.

I have in the past toyed with the idea of multiamping, but always in the analog domain. It always seemed it was too cumbersome, needed too many boxes, and was creating new problems. This newer technology seems to be bridging that gap. Or maybe it's me convincing myself?

Thanks for the clarifications regarding driver time-coherency. Conceptually I understand it. My gut feeling is, though, that lack of coherency is at least one order of magnitude smaller than that introduced by passive XOs. Right? If so, most of the issue would be solved with said software/approach.

Bombaywalla, I reread Roy's White Papers. He speaks to time and phase effects caused by speaker cone mass, suspension elasticity and damping. Nothing about phase shifting (if any) that may be caused by the inductive reactance of the driver itself, namely the voice coil moving in a magnetic field and producing back EMF. Perhaps Roy will catch my Q and share some thoughts.

If the driver's inherent inductance, as a stand alone factor, causes or contributes to nonlinear phase shifting, the challenge becomes a moving target.

Any ESLs out there that don't use X-overs??

@Bifwynne

For the first part of your question, you misunderstand. Pass band is the part of the frequency the driver is covering, unattenuated, within the filter. Actually, I used the term technically incorrectly in the BSC context since that is attenuated long before the crossover point. Driver rolloff caused by inductance usually occurs out of the pass band but is still important. If a driver could, realistically cover from 35 to 20 KHz, than it would require very little inductance. There are drivers with little inductance, relatively, like the Satori MP16, but the numbers you mention are bordering on some AVR brochures :O

The second part is beyond me, even if I could understand the question.

I have a question. If an 18kHz sound left its' source and a 30Hz sound wave left its source at the same time. would they both get to the listener at the same time?

How fast is the train travelling?