3ll3d00d

yes that is correct, something downstream has to add 10dB to the SW channel and this is behaviour is independent of content (i.e. it assumes the ch4 on the input is an LFE channel at all times). I didn't know whether that was normal behaviour or not for an AV receiver/processor, assuming it is then such users will be ok as you say.

I updated their wiki recently and included measurements of all the various DSP options (https://wiki.jriver.com/index.php/DSP) as well as a guide on how to verify this yourself (https://wiki.jriver.com/index.php/Verifying_DSP_Studio) using freely available tools (though I since notice REW has an offline measurement mode that makes that even easier to do). There are a few things in there that don't look like they work or exhibit some unusual behaviour, one of which was bass management (as reported in https://yabb.jriver.com/interact/index.php/topic,114602.msg792538.html#msg792538 and on the wiki in https://wiki.jriver.com/index.php/Room_Correction#Bass_Management_In_Action) FWIW I do my bass management manually (using a mix of PEQ and acourate filters) but I would be wary about using the built in function (i.e. make sure the end to end signal chain behaves as you expect)

Hi @mitchco , you have identified me correctly In case it's not clear from the docs, MSO is basically an automated search algorithm which varies the filter parameters in each iteration and then tests the results against a score (where the score is IIRC the RMS error against the target across the listening positions subject to the per seat weights), the top score is selected as the solution. This repeats for as long as you like to run the algorithm for and the time required can be very long indeed (e.g. if you have a lot of subs and/or give each channel a lot of filters). IME the final outcome (with respect to step response etc) is similar to any other method because I run acourate on top of MSO though I also take care with the constraints put on the MSO filter ranges (inc delay). I'm not sure whether I have good comparison graphs for my current setup, the sub end of it (e.g. from a group delay point of view) basically looks like a textbook (minimum phase) response. My workflow is basically - measure subs at various positions as usual for MSO - configure MSO with those measurements, listening positions and with whatever filters you want to let it play with - run that for as long as you see fit and as many times as you see fit (til you get a result that looks promising) - translate the resulting filters into an appropriate format for acourate (I used rephase for this, life is too short to manually create those filters in acourate and convolve them one by one!), convolve with your sub XO(s) - now run acourate as normal i.e. MSO is playing the "driver linearisation" role in this setup where the driver is actually composed of many independent subs. FWIW I decided to anchor the MSO timing relative to my front subs (i.e. it could vary the delay on the other subs but not what I consider to be my "main" sub which is at the front) and then I aligned the result to my mains. It's quite time consuming but the results are excellent IME. Does audiolense allow you to add arbitrary filters to the XO (or via prefilters) like acourate does? If so I would think the same basic process would work.

@dziemian you can use MSO with acourate, no need for any external devices. @mitchco is audiolense equivalent to MSO? I thought audiolense produced independent per channel filters rather than filters that allow multiple independent sources to integrate as one mono source (via independent eq/delay).

add Macro4RightFDW=1 in the [Expert Mode] block in \Documents\Acourate\Acourate.ini then restart acourate

FWIW the latest version of JRiver (20.0.27) has added this feature 11. NEW: Added dsp studio to DLNA server audio advanced options. REQUIRES the output format to be set to "Specified Output Format".

1. not sure 2. looks like 10th 3. yes

OK I see what you mean (I think). If you are sitting at the 50% (acoustic) distance in a cuboid room then the output from those subs will completely cancel the 1st axial length mode so you don't need EQ in that case.

Cancelling a modal resonance at ~18Hz is a job for an IIR notch filter, I don't see the length/resolution of the dirac filter as being relevant to that (as dirac AIUI employs both FIR and IIR when creating the final filter).

AIUI Dirac uses IIR for the low frequencies so I don't think filter length makes any difference. The filter itself is ~15-20ms long which equates to a frequency resolution of ~25Hz in FIR terms hence pretty useless for a sub.

Does your current setup work well? By that I mean do you get a smooth transition from mains to sub with no obvious integration related response anomalies? If so, I don't know what you have to gain by changing anything. FWIW I think miniDSP posted a good summary of the difference approaches on AVS - 8 - IN 8 - OUT HDMI Audio processor with Dirac Live® technology at CEDIA - Page 3 - AVS Forum If you wanted to do bass management in jriver then I think you could try combining that with peq in jriver to attempt to fix any anomalies in the crossover range. I suspect this will be a fair bit more work though as you would need to iterate over Dirac then peq then Dirac to optimise it.

Definitely agree with this. The spectral balance can reveal/his detail and being different bits of a mix to the fore.

I think what I was thinking is that you are chasing the HF rolloff by treating the 12kHz peak as the anomaly rather than the dip just before it. Your ears rule though, I am just looking at a graph

Bass management and room correction are not related except that they concern the same frequency range (and can be optimised if done together). This repeats the AVS thread on this subject but I think it is definitely suboptimal but likely to be perfectly workable in practice (especially if you know what you are doing). It feels like a case of perfect is the enemy of good tbh (though that might be the very definition of our interest!)

I think we can all agree that good EQ is well worth the entrance fee I think it would be v interesting to hear your thoughts on a listening session based on your existing target curve & the acourate demo. Why do you roll off so sharply above 12kHz btw?