Why won’t UA get into the room correction game?

The challenge with "room tuning" lies in the separate auditory perception of the room and the monitor speaker. Impulse tuning, as opposed to pink noise, offers a method to adjust solely the speaker. However, this approach still contends with the effects of microphone capsule diffraction and speaker diffraction, which can skew high-frequency measurements by reducing the top end. These effects are influenced by variables such as placement, capsule, and enclosure design. Our pioneering attempt to tune a control room at Motown in 1965 faced these issues. When the results were unsatisfactory, we reached out to B&K, the creators of our measurement microphone. The engineer responsible for the microphone's design responded, expressing that he had never encountered anyone attempting such tuning and doubted its feasibility due to the aforementioned factors.

Bob I think you hit on why Trinnov uses their own unique mic.
One of the biggest problems I always hear with DSP correction is the fact that the software has no idea what the difference between what is on axis and off axis. A lot of corrections are based on what’s going on off axis and messes up a perfectly reasonable on axis response. As cheap as CPU power is someone is going to realize a program using multiple mics and evaluating off & and on axis performance will produce better results, but you will never be able to simply use DSP to treat them separately.


For the general question of what is better, it’s going to depend on your speakers & room. And DSP power, a lot of the software choices are made in broader strokes than you think, trying to manage DSP usage. Don’t expect a curve like the ones you see in a lot of manufacturers lit that are ruler flat, it’s not going to happen all your filters are going to mostly be a bunch of bells and triangles, put them together with a typical speaker and you get a bumpy curve. If you want to see ruler flat run the smoothing utility.