A ooaurse room simulation would place some dips both above and below 90 Hz when I have the microphone at 90 cm height.
View attachment 32898
The right speaker is having some other stuff around it (LP storage, pushing the rear wall to the right of the speaker forward, which changes the SBIR significantly). So there i100-160 Hz effect is mostly likely SBIR effects from corner/sidewall. The peak and dip below 100 Hz is not affected though.
View attachment 32899
Here is a test with no panels vs two panels behind on the wall behind the speakers. Blue=panels, red=no panels. (You can also see my intentional voicing with a little more boost 1-2 kHz compared to 2-5 kHz, which I have applied according to Shirley et al. But that is my preference to get a more neutral response for stereo listening.)
First the frequency response 100-20,000 Hz:
View attachment 32904
1/6 octave 20-20,000 Hz
View attachment 32906
And the reverberation time:
View attachment 32905
Very interesting! The panels seem to help with reverb time noticeably but with FR it's a kind of a mixed bag, don't you think?
Yeah that's what I figure... but as the post was preceded by:
I found it a little confusing... but it is late and I am quite tired and maybe a little stupid (well a little more than normal I mean).
Yes that's what I figured the case was. I just thought your "already fixed" and "design and setup" - meant that you were saying you had already handled cancellations of that nature and therefore everything shown was only what you would fix via DSP... which seemed in conflict with the rest of the post. Especially confusing considering trims are much more likely to be effective than boosts via EQ in most situations.It may have been confusing, but in design meaning that the speaker (driver to wall) distance is around 35-40 cm, pushing the first SBIR cancelation from back wall higher up in frequency, enough to be dealt with using a thinner damping panel. Since there is a boost below that 1st cancellation frequency, you need to calculate with that in crossover and driver output.
giphy.gif
(This is not related to room resonant modes, or later comb filtering effects by sound waves bouncing around.)
Yes that's what I figured the case was. I just thought your "already fixed" and "design and setup" - meant that you were saying you had already handled cancellations of that nature and therefore everything shown was only what you would fix via DSP... which seemed in conflict with the rest of the post. Especially confusing considering trims not boosts are usually all that's effective via EQ.
Correct, I edited my original statement to be more clear.Boosts are also effective, but not if you are trying to correct SBIR related dip.
Yes that's what I figured the case was. I just thought your "already fixed" and "design and setup" - meant that you were saying you had already handled cancellations of that nature and therefore everything shown was only what you would fix via DSP... which seemed in conflict with the rest of the post. Especially confusing considering trims are much more likely to be effective than boosts via EQ in most situations.
I'm using Mathaudio room eq incombination with foobar2000 as add In under Win 10. It is free (incombination with foobar2000) work simpel with excellent results. https://mathaudio.com/room-eq.htm
Thanx for your explanation this is what I'm after (as long as i can comprehend it). All these solutions like REW, Dirac live. Mathaudio, Lyngdorf, Sonarworks, Minidsp an probably more it would be handy to have a sort of overview/test what are the benefits or drawbacks of all these solutions for instance Mathaudio avoids pre-echo which will have an affect on the neutrality of the sound whatever that means?. Does that mean that other solutions like Dirac live also avoids pre-echo and if not why, are ther certain disadvantages to do so?. Mine impression is that most of these solutions don't want to reveal how there solution (algo's) really works what is understandable considering competitive advantage. What for me as a consumer important is is that i can choose the best solution for my needs. This could be Mathaudio but another solution fit mabey better. For that i need tests/comparisons, overviews an reviews. IMO room correction is the biggest change atleast for me in the last 50 years in reproducing music (much much) better.Being this a thread dedicated to "Dirac and others" it gives the opportunity to detail about different approaches.
MathAudio specifies that it works with zero latency, as a result (and correctly stated) it doesn't delay the audio track... but "no latency at all" also points to a major difference.
A speaker in a room can be measured and showed to have some frequency (magnitude) response.
What we try to do with Dirac live is to improve this response in frequency (to have the desired target) and in time (we want the impulse response to be as close to a perfect impulse as possible).
When considering only the frequency response we can (in theory) apply any filter that has the inverse frequency response and the resulting response will be flat.
This filter is often a minimum phase filter, and if the system (the speaker together with the room) was already minimum phase, the result will be a flat frequency response and the impulse response will be perfect.
If the system is not minimum phase (and this is the case in our normal listening rooms) the frequency response will still be flat, but the impulse response will not be ideal (exactly how it looks will depend on the phase).
In order to get the impulse response correct for a non minimum phase system you need to use a filter that is not minimum phase.
What this filter will do is shift certain frequencies in time in such a way that they all arrive at the same time, thereby achieving the desired result.
Obviously we cannot move them all to time zero, as this requires us to know about the future.... we have advanced technology but not that advanced
Instead we shift all of them to the latest frequency, that is we delay frequencies to match the latest one (within reasonable limits of course).
As a result the Dirac correction requires some latency which is irrelevant for audio only applications and normally low enough for audio&video.