0
$\begingroup$

So I have this school project where my aim is to improve a loudspeaker in general, my idea is to mesure the impulse response of the room using a mic, then use this mesure to conceive a loudspeaker that can compensate the effects of the room. My question is how could I exploit the room impulse response in conceiving that filter let's, and what kind of filters should I use? would that even be possible?

PS: sorry if the question sounds dumb, I have zero experience and knowledge with this kind of stuff.

Thanks in advance!

$\endgroup$
  • 1
    $\begingroup$ Welcome! Your question doesn't sound dumb at all. What you're looking for is an equalizer :) $\endgroup$ – Marcus Müller Jun 7 '18 at 13:02
  • $\begingroup$ Thanks Marcus! Yes I need an equalizer, but what I am looking for is how this equalizer function, I mean what should I compare the room response too in order to correct it, and is it possible using Matlab :). $\endgroup$ – Mohamed Mahrach Jun 7 '18 at 13:32
  • $\begingroup$ I'd recommend reading up on the zero-forcing equalizer, as I think that's the one easiest to understand without a lot of estimation theory :) $\endgroup$ – Marcus Müller Jun 7 '18 at 13:53
  • $\begingroup$ Pretty much everything is answered here. $\endgroup$ – jojek Jun 7 '18 at 21:36
2
$\begingroup$

This depends a bit on what you want to get out of this and how much effort/work you are willing to put in.

Doing a room EQ that actually works and makes it sound consistently better is quite complicated. There are commercial systems available but they tend to be complicated and expensive or tied to a specific product.

A really good freeware option to play around with is Room EQ Wizard https://www.roomeqwizard.com/

If you want to do this yourself, I suggest you break it down in a few separate steps that can be tackled individually

  1. Figure out how to make a good measurement in a room (noise, reverb, microphone, ADC, SNR estimation, etc.)
  2. Figure out what to measure (using the tools from step 1) that's representative enough of the entire room/speaker/listener situation
  3. Extract perceptually relevant parameters from these measurements
  4. Turn these parameters into some sort of correction that would move the parameters towards some perceptual target
  5. Design a filter and/or algorithm based on this correction
  6. Design the process/HW/system to actually implement and apply the filter or correction

All of these steps have been studied in detail and there is a fair bit of literature available on any of them.

$\endgroup$
1
$\begingroup$

You should understand that a room doesn’t have a single impulse response. The locations of sound sources and listeners matters.

While the term “impulse” response strongly suggests firing a shotgun (with blanks) a wide band continuous noise may be more suitable.

An equalizer that is typically used in communications or radar/sonar is trying to improve SNR and interference rejection. People are not matched filters.

Sound quality is something different.

$\endgroup$

Your Answer

By clicking “Post Your Answer”, you agree to our terms of service, privacy policy and cookie policy

Not the answer you're looking for? Browse other questions tagged or ask your own question.