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Article: Audiolense Digital Loudspeaker and Room Correction Software Walkthrough

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2 hours ago, mitchco said:

Not to be confused with that the correction filter never exceeds 0 dBFS. This is what I was referring to in the video. As soon as you make a correction filter and examine it in AL, you will see what I mean. It is not "boosting" in the traditional sense, like a  +PEQ for example.


Thanks for clarifying @mitchco .


I saw Acourate gives a similar explanation and some cool info:




2 hours ago, mitchco said:

I think you said you were playing with Audiolense? I would recommend creating a few corrections filters with different so-called boost settings and examine what the correction filter looks. And most importantly listen to the impact of increasing and decreasing this setting and what it does to the dynamic range. There usually is a "sweetspot" for any given loudspeaker/room setup, plus what each persons preference is. 


Yes but these technical queries/discussions can be separate to experimenting/listening.


I like learning about what's happening 'under the hood' along the way, even while enjoying the music.


As previously mentioned, discussions/questions don't always mean something is wrong or broken.


Learning is fun. Thanks for sharing your knowledge ! Very very helpful to me and so many others.

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Hi again @mitchco


A technical query here. Disclaimer: this is not an audible problem ! Just some fun learning and discussion.


In your section of this article where you are attempting to correct group delay between left and right speakers: "We can clearly see group delay at 50 Hz on the left speaker and 65 Hz on the right speaker. Again, because the speakers are offset left from center of room. If I could set up the speakers to be perfectly symmetrical on either side of the center line in the room, the group delay would still be present, but at the same frequencies."


You don't show the step responses in this Audiolense article, when fixing group delay but I assume that fixing the group delays did introduce some pre-ringing in your step responses?


I see in your book (Acourate) you do say "the excess phase FDW window size will affect pre-ringing" and you showed the step responses, with increased pre-ringing as you corrected the group delay.


So I assume it's the same (minor, inaudible) trade-off here with Audiolense, where again fixing group delays will introduce pre-ringing in step responses?


The above query only pertains to passive crossover speakers of course.


When I can get some active speakers and do perfect time alignment of all drivers, that will be fun !


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Both Acourate and Audiolense have preringing compensation. There is a big section on preringing in the Audiolense help manual that I highly recommend reading. This allows you to tune the group delay without introducing preringing and therefore the step response pretty much stays the same.


In my book, I was showing examples of varying the excess phase to show what that does, but I was not varying the preringing control. In the end of that example, you can see that both long and short excess low frequency excess phase correction windows, with preringing compensation engaged, shows virtually the same step response with next to no preringing.

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9 hours ago, mitchco said:

Hi @asdf1000 yes, you can find more about in the Audiolense help file. As mentioned previously, if you read the preringing section, gives good insight on how to use the group delay view. I think you already read my post on why I use REW for validation measurements.


Thanks @mitchco. I know why you use REW (for verification) but hadn't seen any Group Delay plots from Audiolense itself in your article, preceding your verification section... hence why I was asking if the exact same can be done in Audiolense. No doubt it's always great to verify with a separate app.


Before messaging you I saw the below in the Help File but this is different to your article. It's one speaker and it looks quite different to the REW plots in your article. Also some of the written text on this section in the Help File is hard to understand.


For example, the issue with this plot isn't entirely clear to me (and not explained in the Help File), maybe because I can't see the red?




Also this text from the Help File isn't clear to me either:


"You can choose the size of the time window in the bass (here: 10 Hz) and in the high treble. You can also specify an optional mid frequency and the associated duration. This will often come in handy if you wish to progressively increase the scope of the time domain correction towards the lowest frequencies. You can also adjust the low frequency (here set to 10 Hz) can also be adjusted. This is the end point of the window, and anything that falls outside (in this case 800 ms). If we change the low frequency to 100 Hz, the filtering will stop at 80 ms."


The underlined reads like the 10Hz here is upper limit (end point) of the window?


But in your article it reads 10Hz is the lower limit (you say starts from 10Hz - which makes more sense).



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@asdf1000End point means lower limit in this case. So at 10 Hz with an 8 cycle window will yield 800 milliseconds of TTD at that frequency. So for frequencies at 10 Hz and above, an 800 ms window is applied until we reach the upper limit of 24 kHz, which in your example above is a 2 cycle window which equals 0,083ms. It is frequency dependent windowing.


So you can adjust the lower and upper frequency limits and for anything outside those frequencies, no windowing is applied. Within the frequency limits, you can also vary the amount of windowing applied. And anything inside the frequency limits and withing the time domain windowing, will get TTD applied.


As mentioned, we are talking about frequency dependant windowing. You can read more about what that is here: http://drc-fir.sourceforge.net/doc/drc.html#sec32  May not be exactly what Audiolense is using, but gives you an idea. Also, you can work out the FDW math here: https://www.audiovero.de/acourateforum/viewtopic.php?f=11&t=14  Again, may not exactly represent what Audiolense is doing, but again gives you an idea.


As far as understanding group delay, minimum phase, excess phase, I would recommend John Mulcahy's article on minimum phase: https://www.roomeqwizard.com/help/help_en-GB/html/minimumphase.html


For product related questions, please reach out to Bernt on the Audiolense forum (Edit: I see you have, good).

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8 hours ago, mitchco said:

@asdf1000End point means lower limit in this case. So at 10 Hz with an 8 cycle window will yield 800 milliseconds of TTD at that frequency. So for frequencies at 10 Hz and above, an 800 ms window is applied until we reach the upper limit of 24 kHz, which in your example above is a 2 cycle window which equals 0,083ms. It is frequency dependent windowing.


Thanks @mitchco


I understood the maths of frequency, cycles and correction window in milliseconds at 10 Hz and 24 kHz but not what was happening in between.


What didn't make sense to me was talking about a correction at 10 Hz (not clear from the Help File or any articles).  What happens at more meaningful bass frequencies like 20 Hz, 30 Hz, 40 Hz etc etc ?


But Bernt answered this today as you saw with this bit of info:


"a gradual transition (in cycles) between the two frequencies."


So in the example below, it isn't a constant 200ms window from 10 Hz to 24 kHz - Bernt clarified that there is the gradual transition. Which now makes all the sense in the world of course - hindsight !


It's nice to understand what is happening between the fields of the program because it isn't that obvious initially (to me anyway)



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  • 7 months later...

Hi Mitch (@mitchco),


I ‘ve been re-reading your book and wondered where you sit at this point on near field corrections and beam forming. Both concepts make a lot of sense to me. Just wondering with time, experience and switching to Audiolense whether you feel they are worthwhile. 

I recently did do a Nearfield measurement and correction in Audiolense. I loaded that in Audiolense convolver, then I did a new measurement at a distance . I then loaded nearfield and farfield cascading filters in Roon, changing the CFG file as needed, but was disappointed to see that Roon was not properly handling it (or maybe I’m doing something wrong). Before I spend too much time on this I thought I’d ask you if it’s a waste of time. 

The second part of my question is about multi seat measurements. I usually measure in a few locations right, left, front and back of my normal sitting position. In order to take advantage of beam forming would it be smarter to do these measurements front to back exactly at the midline using the accurate microphone alignment tool?


Thank you 



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Happy New Year Omid! 


Wow, well I could write a chapter or two about that :-) If I had a passive speaker system, I would just apply room correction and be done with it. The reason being is that the drivers are already hardwired (i.e. designed) with a passive XO and with the tradeoffs between on and of axis response, etc., already baked in.


Now if it is was a digital XO system, where you control the XO and each individual driver, that opens the door to making some real system improvements and worth the effort. Reason being, you have full control over all aspects of the XO design and full control over each individual driver. 


With Acourate, you have read the pattern and how to use the tools to achieve the results. Audiolense does something similar but using a more automated approach. From the Help file: "When True Time Domain Correction is checked, the whole speaker will be time domain corrected as one single entity. This is the only TTD option for passive speakers. The option “TTD Correction per driver” will also be visible when digital crossovers are used. Checking this option will also time domain correct each driver individually before the speaker is assembled and corrected as a whole."


Folks should know that these are linear phase digital XO's and unlike passive XO's, sum perfectly in the frequency and time domain. With each drivers frequency response convolved with the digital XO is how they still sum, now acoustically, perfectly.


So if you are using Audiolense, it was intended to be used with the mic at the LP as it's process is a bit different than Acourate.


Beamforming is a very specific approach and worth reading the paper referenced in the book. As you can see in the book example, it really is a semi-anechoic response in the top end. i.e. major reduction in the mid and high frequency reflections along the beamwidth. But again, a very specific procedure goes with it and needs to be followed to get the result advertised. I tried it with horn loudspeakers that are a bit higher in directivity and the clarity and closeness of the sound was amazing. But did not sound really natural to my ears. Some room reflections are of the good kind above Schroeder ;-) It is a worthy experiment to hear it, but the formula needs to be followed exactly for it to work.




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Thank you for the thorough answer. 


I’m using Audiolense for my digital XO. Based on your answer I will not complicate things and keep my filters as they are, ie based on the listening position sweep rather than breaking it in  two parts for Nearfield and farfield. The multi seat filter will include positions to the right and left not just in the middle line.


Thanks to you, I don’t have to worry about cascading filters, and losing 7 to 10 dB with each filter. 


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