New active Satori Textreme

Yes, as ernperkins says... I have both Omnimic software and Hypex HFD filter design software running on the laptop. The laptop is connected to the Hypex amp via USB cable, and the Omnimic is connected to the laptop with another USB cable.

I make a sweep, capture the FR curve, and assess it. Then I make an adjustment to the DSP filter, load it into the amp, and make another sweep. rinse... repeat... it is an iterative process, but with care it can be quite accurate.

This is basically the process recommended by Hypex, except that they want me to use their built-in measurement capability. I prefer to use Omnimic for several reasons. However, I might give the Hypex built-in process a try. I now own a Dayton EMM-6 microphone and a Behringer UMC202HD audio interface.

My expectation is that for each driver, the on-axis response and the listening window response will be nearly the same through each driver’s pass band. If it is not, then I have done something wrong in the design. Once I have confirmed this, there is no longer a need to measure the listening window response until final system testing.

Using the boost/cut filters and shelf filters (first and second order), I adjust the eq of each driver such that it is smooth and flat when viewed with 1/6 octave filtering. I need to be careful to distinguish between driver responses and diffraction responses. I do not correct for diffraction responses. The flat response needs to extend for at least one octave beyond the crossover frequencies.

After the drivers are flat for one octave beyond the crossover points, I set the levels of the three amps so that the drivers are at equal level. The next step is to measure the acoustic delay between the drivers and make that adjustment in the filter software. At this point the drivers have a flat response, have the same level (sensitivity), and due to digital delay, they all originate from the same plane. Since drivers are minimum phase devices, once they are eq'd flat, they have phase alignment. The final step is to apply crossover filtering. Because of the adjustments and manipulations, idealized crossover slopes can be simply applied.

Of course, it is possible to load all the raw driver measurements into a simulation program, such as the very powerful Vituixcad, and develop the DSP filtering algorithm by simulation. However, given the ease of developing the filters based on iteration and rapid feedback, this seems unnecessarily complicated. It is analogous to tuning a guitar. One way is to measure the actual frequency of the string and compare it to the desired frequency, then using physics and math, determine how many degrees the tuning peg must be rotated. The more simple approach is to just start turning the tuning peg until the digital tuning meter says we are in tune.

I use Vituixcad to assess my filters after the fact to make sure I have not done anything stupid. I expect close alignment between simulation and measurement. But the actual development is conducted based on iterative measurements and adjustments.
I will certainly use this method. Edit: and the rest of the thread!
 
Last edited: