focal tweeter tuning

speakermakers wrote: What haemphyst says is quite possibly true. It depends on how the software was written. It can be done both ways, and is. Typically I have found that if your software includes features like selectable crossover types (Butterworth, Linkwitz-Riley, Bessel, etc.) there will be phase characteristics that mimic the analogue design. This can be of great use in the right hands when dealing with asymmetrical configurations and uneven path lengths. But it can cause mass confusion in the wrong hands. There are no hard rules on this though as far as I know. And I have found that its nearly impossible to get this information from the manufacturers. When you think about it, who’s answering the phone. Its not likely to be the guy that designed the processor, or anyone close to his pay grade. So how reliable could any information that you get be?     If you get your hands on those CDs though, it’s easy to determine proper time alignment by ear (after calculations). And like I said after all the calculations are done, and done correctly you will still most likely be off by some amount on a couple if not several of your speakers. The fine tuning here must be done by ear using material specifically recorded for this type of tuning.

Maybe it's not such a good thing that my processor has selectable filter types(Butterworth, Linkwitz-Riley) - I may not be "the right hands"

greenbroncoguy wrote: sedate wrote: Since the function is contained on the integrated circuits and microchips in the headunit, it would sort of have to be 'digital.'  This is why all CD players have a "DAC" - digital-to-analog converter that finishes the signal for the RCA's. I think just the reading of the CD part would be digital. If you were to listen to the radio, it would not have a digital signal I wouldn't think...

If you're controlling time delay, filters, etc. with the headunit it must use an A/D converter on the radio and any other analog input signals first.  If the filters were analog then I don't think you could change them by pushing buttons.  Of course the question of whether the filters were programmed to behave exactly as analog filters (i.e. phase shift) remains.

Don’t worry your self too much about this. On the surface that sounds quite irresponsible but, let me explain. In a perfect world where everything works out just the same in the car as it does on paper everybody would use 24db slopes with no phase altering characteristics and all would be jolly. But this is not the case. In the real world we don’t even know for sure where the acoustic centers (the point which sound starts) of our speakers are. This point is theoretically at or near the dust cap but varies greatly by speaker design. There fore even if you mounted all drivers on the same plane and the same distance from your ear and used phase coherent crossovers you would still not likely achieve a phase coherent acoustic wave front. Now don’t get me wrong. Calculate everything you can. Twice, three times. Account for every possible variable. In the end you should be within one wave length or less of your target on all speakers in the system. After that you can easily determine precise phasing, time alignment, and amplitude per speaker using gated pink noise (no equipment necessary). Using gated pink noise you can quickly achieve elements of music reproduction that most people don’t even know are possible. At this point and only at this point you should also experiment with alternative slopes and types of filters (Butterworth, Linkwitz-Riley, Bessel, etc.) if they are available in your software. In almost all but, not all cases there will be considerable advantages. This is why I encourage you to master your abilities with 24db Butterworth (or what ever your software defaults to) slopes and then move on to experiment with uneven and odd order slopes. You can always go back to a previous setting. Ya gota love digital!

as I suspected, thanks.