Thanks! I didn’t only mean compare end result but compare the entire process of getting there.
And the differences in features.
AL XO tackles group delay in a different way to Acourate (I’ve seen mitch’s latest video showing how V2 does this).
Plus my setup is not straight forward, it’s 8-channels of 3-way DSP crossover plus 2 subs.
I have been meaning to get Acourate V2 when I get spare time. Hopefully this year some time I’ll get it
Please, do share your experience. I myself am very curious about AL as well.
@dabassgoesboomboom hi, now we meet here. 
I would like to try using AL to mix the speakers in the car, respectively, since you are sitting on the left or right and not like in the room in the middle, the delays will be slightly different. For starters, these are 4 channels, 2 tweeters and 2 midbass. As I understand it, AL can easily cope with this task, and HQPE will be a convolver directly for listening to music.
I have a UMIK-1 microphone. If I’m not mistaken, first you need to make measurements for each speaker separately.
The link is the following:
A laptop on which AL + is connected a USB microphone UMIK-1 + USB sound card Asus U7 which has a TOSLINK output, which in turn will go to TOSLINK ASUS XONAR D2X, which in turn is connected to HQPE and it will also be a 4-channel DAC for speakers .
Correct !
Yes Audiolense XO measurement process runs automatically - it plays sweeps through each speaker separately. This measurement part is automatic. Only 1 measurement position but it works very well.
After the measurement which is fast, you need to spend lots of time ‘optimising’ the correction filters till you are happy with the simulation results that AL provides.
Yep but for the measurement process, HQP is obviously not in the picture in this step.
AL is Windows only, so you need to connect your 4-channel DAC directly to the Windows laptop, for measurements.
Then AL can export mono WAV files, which you import to HQPlayer’s Matrix Pipeline section
See these 2 great detailed articles running through how to use Audiolense XO:
and
Anyway, yes, I’ll try following the instructions on the link (thank you very much, just what you need), this is for a start, but it seems to me that for more reliable measurements it is necessary to pass the test signal through the device that will be used for listening in the future. Elementary, there may be different circuitry for the output stage of the DAC, operational ones can rotate the phase in their own way, a different frequency response, other delays, etc. IMHO of course.
Since TOSLINK in the Asus U7 can only output 2 channels (and most likely the Xonar D2X can only receive 2 channels), a possible option would be to measure in turn (in the order indicated in the instructions), first the bottom left / right Midbass (before that in mixer by turning off the tweeters), then run the tweeters by turning off the midbass, AL even has a separate metering function, and not one after the other in a row.
Although no, I was wrong, AL does not allow you to put the same channels on
both on Twitter and midbass. So I will run it on the same equipment where Win + AL music will be played, it will be more correct
@dabassgoesboomboom I read the first link, I roughly understand, but I have one question for you. It turns out that when AL generates WAV pulses for each of the 4 channels, do they already include time delays? Or will it be necessary to add them manually to HQPE?
Highlighted in red is it necessary to set the distance to the speakers in HQPE?
Also, even if mono .wav is selected, configuration files with numbers are saved, should data be used from them?
No need to do anything - the correction filters already incorporate all time domain and frequency domain corrections
Those delays you show there are just some fast visual feedback
You will see the same time delay when you look at measurement impulse responses for each driver seperately
And the simulated impulse response corrects these delays
No need to setup time delays or distance in HQP or any convolver. The mono WAV has everything built-in.
The config files help with understanding the channel mapping in convolver.
So you don’t import those files for HQP but you need to look at them to understand how to setup pipeline matrix in HQP
Only mono WAV get imported into HQP but of course you need to send correct sound to correct driver. Thats where the config file helps. You don’t want to send bass to tweeter for example.
So have to be very careful with the first music playback, start at very low volume and listen if wrong sound is going to wrong driver.
Listen at really low volume until you are 100% sure your channel mapping is setup correctly in HQP
Yes indeed I just downloaded the impulses. And I was very impressed, the system started playing in my car at a very serious level … The stage appeared in front in depth and wide, the tonal balance was evened out (although there is still something to work on), in general, it’s not very bad, and this is on a simple sound albeit tweaked Xonar. I think when I finish the 8-channel DAC on multi-bit pcm1702, it will sing awesome. I still have to play around with the target curve and determine the optimal cutoff frequencies for my speakers, since the lowest frequencies have clearly become less, but in general, yes, it works very cool.
That was fast progress!
Yes you see the measurement takes 60 seconds.
But the ‘optimisation’ of correction flters can take hours/days/weeks
True. Especially to produce your target curve that can spend YEARS. 
And before that, there are still years to eliminate all the jambs of the influence of the salon and get good measurements))
@dabassgoesboomboom In what format do you save pulses from AL to HQPE? HQPE will understand 64bit float?
yep 64bit mono wav
I create 64bit 352.8 kHz mono wav
HQPlayer will automagically convolve at the music source sample rate
So you don’t have to create different filters for different sample rates.
Only create for the highest sample rate of music you listen to
And during measurements, what parameters are set in the Sample Srrings window?
I’m just thinking, what is the right way to set the frequency with which the HPQE stream will then output? Or the one with which the microphone works?
Measure with your UMIK-1 at 48kHz
Set 48kHz sample rate in AL for measurement
UMIK-1 works in 48/24 mode
By the way, for the Audiolense Convolver, no or minimal delay is declared. It’s possible to watch the video … But this is only for windows
Yes I mentioned before there are other zero latency convolvers better for video.
On macOS I use free multichannel convolver:
http://www.angelofarina.it/Public/X-MCFX_convolver/
You just need to make a minimum phase crossover and minimum phase target filter
It can still be 65k tap length
