Deqx pre8

Hi Ted,

Like me, you must be on the second wave.  Mine showed up a little less than a month ago and it been a tedious process.  I have been told by someone on the first beta wave that we're fortunate to be on the second wave since some of significant the first wave issues have now been worked out.  As you have probably figured out by now the beta UI is stark and not all that friendly, but as I've been assured the final consumer version will be much improved over what we are seeing now.

Have you made your first correction profile yet?  The videos made by Matt at DEQX show the essential basics. 

I've made a few profiles so far but I'm not there yet.

I hope to see more flexibility in the web-based cloud software in the future.  I'm so use to what you can do in the old Windows based DEQX-Cal V3, but Alan told me  "to rethink how the software now works and it has not relationship to the original DEQX-Cal software".  That was good advice.  Yes, it is fundamentally different than DEQX-Cal.

DEQX Pre-8 to using 6 channels for The Apogee original Full Range fully rebuilt and upgraded, powered by 2x H2O S250 SE with two subs.

Thanks for responding. I suggested to DEQX that they start a forum for users to communicate experiences and suggestions, but we’ll see.

i made some profiles and got it up and functioning. It’s very transparent. I plugged REW into it and got some strange measurements though, vey confusing. 
I’ve been using a manual DSP, Xilica, so I’m used to setting driver timing, crossover slopes, and PEQ’s, so I’m missing the ability to control more parameter.  
mic placement seems important.  
I know it’s still in beta, but I’m hoping for increased ability for user tweaking.  
I’m running a usb cable from my laptop into usb input to run REW through DEQX and I’m getting the REW sweep sounds, so I think my measurements are accurate, but like I say, the results are not what I’m used to.

Typically I like a 8db or so slope from 100-15khz, not sure how to achieve that yet. 
the overall sound is very good, but I’m trying tailor the sound more to my liking, and the unit seems so focused on automation I’m having trouble. I’m optimistic about it though, wondering what the final software release will change.

Hi Fellows,

I was in the first wave. There was a plan to set up a forum for users. Have either one of you tried to contact DEQX lately. They have not answered any of my inquiries for three weeks. I'm a bit worried that COVID took to much of a financial toll on them. 

I have a difficult system for the Pre 8. I use dipole ESLs and the comb filtering they produce confuses it. I'm working on a solution to absorb most of the back wave so the 8 can get a measurement on it.

@caglioti you are right. we need more flexibility, manual control of delays and separate EQ for each channel. Right now I have set the crossovers and run it in manual mode, EQing manually. The imaging is not yet what it should be. Rome was not built in a day.
 

@forrestc Which Apogees? I use to have Divas. You have the same basic problem I have, Dipoles, perhaps not as bad as mine as the ribbon is not as transparent to the rear wall reflection. Place the microphone as close as you can to the ribbons, within and inch!  Look at the response curve. If you see it bouncing rhythmically along the x axis you are picking up the comb filtering.  If you go to the Sound Labs web site check out the "Sallie" rear wave absorption panel. I am building a 24" wide version with 8" wedges instead of the stock 6" wedges. If you are not comb filtering then it is a mute issue and you should be able to get a decent measurement.  

Hi mijostyn, Alan replied back to an mail I sent may 30 on June 3. I figured that he must be swamped between further development and answering user questions. I questioned him because was perplexed at the final response curve that I was seeing. I saw significant dips in the final response curve at the crossover points and moreover, it just didn’t sound right. I also mentioned few items relating to the legacy DEQX-Cal v3 that they may want to consider to eventually integrating into the software. Alan assured me that the response curve we are now seeing with dips at the crossover point(s) is not accurate and in reality those dips do not exist in the actual response. He went on to say the they are presently working to correct that issue.

FYI. I get my best measurements at about 16" (40cm) from the ribbons. For me this yields right at 10mS to the first reflection on each speaker and I set the Trim in the Chirp Config there. I really cannot go closer with the mic since the mid-range to tweeter ribbons are passively crossed over and I’m happy at 10mS. I also set the lower limit of correction to 200Hz and not 20Hz as is the factory default. It probably wouldn’t hurt to go as high as 400 Hz I’m thinking, and I’m sure that I’ll try it eventually.

In the meantime, I did work out the issue of bad sound. The issue turned out to be a slight to sever phase difference between the left and right speaker at different frequencies. I ended up playing with the driver distance relationships under the "Create a Speaker" tab, starting by equalizing the distance left to right of the non-zero drivers. It’s probably not perfect, but it now sounds really good now. I have not integrated the subs as of yet. I’m trying to work out as many of the issues with just the main speaker first before adding a whole new mix of issues surrounding sub integration since it is done way differently than in DEQX-Cal.

Of yes, the Apogees I’m using are the original "The Apogee" which was their very first model with all steel frames and weigh about 300 lbs each.

http://www.reality-audio.com/full_range.html

Oh, one more thing. Does anyone know how to do a Volumio Plug-in on the Pre-8?

I opened a ticket with Volumio  about two weeks ago but it's still in the "pending" state.

@forrestc I have a problem with volumio. I reached out to Volumio and they referred me back to DEQX who have not been helpful either. I use Volumio to plug in Qobuz which works everywhere in my house except on the Pre 8. I went to up date Volumio as I was behind in versions. The unit crashed during the update. I reset it to factory settings and now other than the volume control the front panel does not work. I operate the system from my computer. There is a set up procedure for Volumio and I assume until this is done the Volumio attached functions will not work.

You are right staying back from the midrange and tweeter ribbons. Tri amping that speaker is difficult because the impedance of the individual ribbons is close to zero. My Divas had a huge back of resistors to compensate. 

10 ms is 10 feet. You are 10 feet from the rear and side wall? 

I see the same problem at the crossover points. I always check the results with an independent USB measurement system and there is a dip in the response at the high frequency crossover points. This only occurs at the midrange/tweeter crossover. The subwoofer cross seems normal although the delays on the subwoofer are OTL (Out to Lunch). 

I also have a phase problem. Usually, one of the high frequency channels will be 180 degrees out of phase, but it can happen with a lower frequency channel. I set the microphone positions with a tape measure. I have no idea why this is happening. 

I have not used any previous DEQX models. I used a Tact 2.2X for 20 years and it's correction system was faultless. It also offered much more flexibility, but the unit finally expired and the company is out of business. I looked at the Trinnov Amethyst, but I did not think it's bass management was flexible enough. 

mijostyn, Im glad that you pointed that out to me. You got me thinking. Even though I used lots of very sound absorbent padding on the floor, walls, etc, and I’m sure that the cloud processing uses software algorithms to minimize first reflection interactions, I’m still about six feet from the side walls and just under four feet from the floor with the mic. I just now took another very close look at the impulse response in Lin X mode and i noticed some rising "fuzz" on the decaying curve starting at about 5.5mS. That’s probably more realistic. Later tonight, I’ll go back and run it through to the end and take a listen. I appreciate the heads up!

Oh no, I didn’t want to hear that about the Volumio installation. I was hoping to use the Softsqueeze plugin in order to use LMS (formally known as Logitech Media Server) directly to Volumio. Right now, I’m using LMS with the UPnP/DLNA Bridge plug in (on the LMS side) developed by an LMS community member.. It works but making a connection isn’t always smooth when connecting to Volumio. Sometimes you need to mess around with it to get it working.

@forrestc 

The problem comes with the first step, defining the drivers. In this mode the DEQX assumes everything is coming from the driver. It uses this measurement to determine what is room effect and what is the driver. Getting a near anechoic measurement on a dipole is next to impossible unless you drag them outside. Remember, dipoles rediate in a figure 8 fashion and sharply beam vertically. You only have to worry about the rear wall and reflections coming from it.  I tried blocking the rear wave with a quilt draped over a tall tripod. It worked up to about 1000 Hz then the comb filtering resumed. I am going to build a 24" wide "Sallie" 7 feet tall which should do the trick. If I can't get a decent measurement then I'm afraid that will end my relationship with DEQX. 

Don’t give up too quickly. I know that it can be done. Yes, dipoles do radiate to the rear. Some less than others, but like you state, it is a major factor.

The Pre-8 is my fourth DEQX since 2012 and I have great success not only with with this pair of Apogees but earlier with a pair of Duetta Sigs as well. I’ve always used "conventional" indoor DEQX recommended methods to measure the ribbons. I will admit that there has been a lot of trial error with mic placement, crossover point and slope selection as well as speaker placement and toe-in (or in my case no toe-in). I’ve also had great success with precise subwoofer integration with both sets of Apogees as well.

This Pre-8 was purchased to replace my third DEQX which is a Larry Owens heavily modded HDP-Express II. To be honest, there was nothing at all wrong with the sound. In fact, the sound was drop-dead great. Fantastic imaging, great slam, deep smooth bass, very realistic sounding piano and voices - sometimes I had to wonder why am I messing with something that ain’t broken!

I’m thinking that the fourth generation DEQX will actually go to that next level. It may not initially as the software is still a beta release. I fully believe that the hardware is a SOTA as humanly possible at this date. No other manufacture of ANY audio product of ANY type is at the technological level of DEQX. Yes, you’ll see plenty of new, latest and greatest stuff for sale out there saying that they have the latest technology, promising to take you to the next level and built with unobtanium suspended in pure ether, and of course at a price that you would need a second mortgage in order to purchase. And in the end, it’s all the same stuff but with new lipstick.

I know that this doesn’t apply to anyone on this thread, but most audiophiles I talk to don’t mind spending the money but they want it simple. They just want to plug a few components together, read a review or two about a power cord or such, buy it, two minutes to install it, a week to break it in, and life is good.

If anyone who has electrostatic, planer magnetic, ribbon, MBL, or other non-conventional speakers AND DEQX, it’s never going to be plug-n-play. DEQX wants to make it easier for their customer and they gone a long way to make that happen. I’m sure that many potential DEQX customers have been scared off in the past by its level of hands-on technological prowess required. On the other hand, they can’t abandon their nuts-and-bolts type customers like us as well. I doubt many any beta tester customers expected plug and play.

The knowledge may be proprietary, but at this point, I for one would like to know what, at least in general, goes on in each step of the cloud processing. A block diagram would be great. That way we look at it and say, "ah-ha" that where _____ happens. Right now it’s a black box. Data in, data out. If it doesn’t come out right, change a parameter and try again.

OK, I’m done.

ForrestC in Tallahassee

@forrestc 

I was not expecting plug and play at all. However, there are a lot of bugs and idiosyncrasies. The basic functionality is there and you learn your way around the idiosyncrasies. What worries me most is the lack of communication lately. Something happened. Hopefully nobody is ill or hurt. Maybe they are just tired of listening to me b-tch. 

ESLs have an extremely light diaphragm. It is transparent to reflections. You can hold a perfectly normal conversation with someone standing behind the speaker. The microphone picks up everything coming off the back wall and assumes it is coming directly from the loudspeaker. The curves are a beautiful example of comb filtering from 500 Hz up it looks like waves with +3-4 dB peaks.  Anyway, I'm building the solution. Not much above 200 Hz is going to get by this device. 

This is my first DEQX. I used a TacT 2.2X for 20 years. About a year ago it died. The 8 was not available yet. I got a MiniDSP SHD to hold me over. Dirac Live works surprisingly well and for $1500 the SHD is an amazing value, but it's DACs are not up to the quality a top system demands. Benchmark Media systems uses a SHD Studio with their own DACs and are very happy with the results. The SHD is now in a friend's system. The DACs in the 8 are top notch and even in bypass mode it is sonically superior to the SHD. 

@caglioti 

You should be able to tailor your curves with the EQ function. I roll the top end off from 1000 Hz amount depending on the volume. I put a corner at 20 kHz. and sweep the Q all the way over as far as it will. Then all I have to do is move the corner (or very low Q filter) up or down depending on the roll off suitable for the situation. I also EQ the bass up 10 dB at 20 Hz. What are you using for speakers? 

After some offline discussions with mijostyn about the difficulties in taking accurate measurements of dipolar speakers such a the Soundlabs that he has and the Apogees that I have, as well as any other dipolar speaker for that matter, I found that his experience and advice in this area proved to be correct and very helpful.

>> YOU MUST BLOCK THE REAR WAVE OF A DIPOLE SPEAKER FROM REFLECTING FROM THE WALL BEHIND IT IN BOTH STEP 1 AND STEP 2. <<. Otherwise, you will get comb filtering which produces all sorts of undesirable side effects. Even though your graphic results may not show what appears to a traditional repeating comb "tooth" pattern over a wide frequency band, you will probably immediately notice a problem in step 2 when the driver distances appear. Your first clue is when you see unrealistic relative distances between the drivers in your main speakers. I have found that these driver distances should be within the hundredths-of-a-meter without manual adjustment. I’ve also found that manual adjustment won’t fix this problem either. If you take these unrealistic defaults you will most likely have very noticeable phase issues. If you manually correct the distances, you may correct the noticeable phase issues, but it will still not sound right. This problem may be because there are driver phase/group-delay correction issues that cannot be corrected by manual relative driver-distance manipulations alone. Remote subwoofers are a different story and may need manual relative distance intervention. I’m still looking into the best way to deal with remote subs. Any suggestions in this area are certainly welcome.

I my case I was able to effectively block the rear sound wave from my Apogees with a heavy sound-absorbent blanket similar to what you would find in under-hood sound insulation in a automobile but of higher density. Please don;t ask exactly what it is or where I got it, it just appeared in garage one day, but any heavy-weight sound absorbent material such as a heavy quilt should work just as well. Just be sure to position the material in such a way in order to cover ALL of the rear of your dipole speaker. As I understand from our conversations, due to the exceptional acoustic transparency of the Solundlabs and probably most other ESL speakers as well, mijostyn informs me that simply blocking the rear of those speakers with a quilt was not enough and for this reason, he is building a Sallie for his unique situation.

Finally, be sure to remove the rear covering before starting step 3 since in this step acoustic properties of your entire room are quantified from your listening position.

Here a tip to help minimize aggravation when measuring speakers.  

If it's going to take a bit of time to reposition blankets, wall coverings, sofa cushions, furniture, etc, between measuring your left and right speakers in step 1, consider LOGGING OUT while moving these items to set up for measuring second speaker then log back in just before your start to take measurements again.   What I have found is that after taking so much time to move and reposition these items, the Security Token would expire right in the middle taking driver measurements requiring you to start all over again on that speaker.  This is especially aggravating when you have completed your main speaker and the token expires after you move to its associated remote subwoofer.

Welp, I'm going to throw out what I know may be a terrible idea.  Sorry.

What about taking a cue from ground-plane measurements?  Often used for subwoofers.  The idea is to put the subwoofer up against the floor so there is no reflection point.  The microphone is also nearly at floor level. 

What I am thinking you might want to do instead is to put your planars up against a wall and attempt the same thing? 

Sorry I'm not familiar enough with DEQX to help more.

Actually that sounds like it may be a great idea for measuring the subs and I'm going to be thinking about doing just that, but unfortunately that's not going to work, at least in my case, for a pair of 81-inch high 300lb (each) speakers.  Even a couple of inches to the right/left or forward/back becomes a quite a chore.  Thanks, every idea helps!

The problem that I run into during the integration of a pair of stereo subwoofers using the Pre-8 is that all drivers’ relative sound levels, for both the main speakers and subs, are measured during the first step. Essentially, you place the measurement mic very close to each driver for left channels then repeat the process for the right. After all the driver measurements are collected for the left channels, graphic results are displayed of the response curve for each driver relative to each other. This in my case includes the left subwoofer sitting about five-feet behind the left main speaker and the same is on the right side as well. This seems to work well for drivers on, or nearly on the same plane, but for the subwoofer that I just took a near-field measurement, five feet (1.5m) behind my main speaker, is not going to have the same relative sound-level when measured at the plane of the main speaker.

In DEQX-Cal, used for legacy DEQX processors, you could match the level of your subwoofer to your main speakers by matching the subwoofer curve amplitude to the main speaker’s curve as measured from the sweet spot, but this was a manual process. You could also match impulse response of your subs to each other then both to the impulse response to your main speakers in order to set the correct delay. It appears now that subwoofer time alignment as well as time alignment of all drivers is performed automatically in step two with a provision to override (in distance).

But how could step two possibly set the subwoofer’s relative loudness when relative levels are set in step one and the subwoofer is substantially on different plane than the main speakers?

For me when I perform driver measurements in step one and close-mic my subs, the final outcome is low bass "lite".  As a kluge, which actually returned fairly good results, I made step-one measurements of the main speakers, in my case at 16" (40cm) then moved the mic along the same plane as I measured my main speakers to where the mic was directly in front of my sub and still on the same plane as my main speaker. This placed the measurement mic about 66" away from the front of the sub and I then took the measurement. I also limited the lower speaker correction (but not room correction) to 200Hz crossing over at 60Hz12dB/oct since the Apogee woofer panels go very low. The bass is pretty damn good now, but I know this couldn’t be best way.

Always open to suggestions.

Has anybody found a way to EQ individual drivers?  As opposed to EQing the summed frequency response. Thanks

I don’t think the software is there as of this latest release. You may want to submit this as a "Suggest a New Feature" request.

So far I’ve only submitted two request but I’m sure that I'll be adding more  soon.

The following are the two request’s that I have submitted:

1. For a provision to display the room response showing any automatic PEQ correction (if any) applied after all calibrations have completed. Also with the ability to auto-correct AND manually PEQ correct the room response.

2. For a provision in Volumio on the Pre-8 to allow installation plug-ins for the Volumio plug-in store.

As for the first request, if you have ever used the legacy DEQX-Cal, it has this feature. What was nice about it, is that it did have the Auto-Correct feature which by default would apply PEQ correction very conservatively. It gave you a great starting point and from there you tweak those setting or change it up completely. It applied the updates in real time where you could immediately hear any change. Right now as it stands, you cannot visualize room response at all. I assume that room response measurements are taken and adjusted automatically in step 3 of the calibration sequence.

The second request is asking for a feature at all other implementations of Volumio already have. Hopefully the Volumio folks can code this one for DEQX so that can keep working on enhancing the core product’s software.

Hi, looking for any “best practices” with using the Deqx.  We’ve gotten several software updates, still waiting for the final “retail” release. 
I’m trying to understand the mic positioning.  Has anybody tried to create quasi anechoic conditions through placement and room treatment surrounding speaker/mic? Was this noticeably beneficial? 
Also, has anybody found success with adjustments of “trim” , “smoothing(?)”, or attenuation/boost?  
just looking to get some anecdotal opinions from users who’ve found some better ways of using the unit.  DEQX so far is giving very little help in how to best use the unit, outside of the most basic operating requirements.

thanks, Ted

By placing the measurement microphone about six-inches directly in front of each driver to be measured should minimize its first reflection back to the microphone.  Of course the best, but usually not practical solution would be to place the speakers to be measured either outdoors or in a very large indoor space short of in an anechoic chamber.  In my case with the Apogee FRs, the mid-range and tweeter ribbons are crossed over passively and are treated as one driver by the Pre-8.  For that reason, I have to set the measurement microphone directly between the mid-range and tweeter ribbons about 16- to 18-inches away from the area in front of the two ribbons.  For consistency, I also measure the bass panels from the same distance as well.  Unfortunately, moving the measurement mic away from the driver, such as in my case, is not ideal since by doing this allows more time for the first reflection to be picked-up by the measurement microphone. 

During the ownership of four DEQXs over the last 12 years, I have found that trying to create a quasi-anechoic environment with easy or economically to obtain items is almost impossible.  A specific material may be great at absorbing energy at some specific frequency but actually reflects sound quite efficiency at other frequencies.  I have always taken my best measurements when the measurement mic as close to the drivers a practical.  Again, about six-inches seems to work best for conventional come and small drivers, but you may need to back away  a little when measuring ESL, magnetic planer, ribbons, etc.

For the "trim" adjustment, as I understand it, this setting is used to truncate the lowest frequency of speaker correction.  Trim is set so the DEQX will not correct frequencies that occur after the first reflection.  You can determine the first reflection after you complete the "Measure Drivers": task by changing the graphical view to.from Frequency Response to Impulse Response and the expand the X-axis noting where there is a small second impulse after the decay of the initial impulse.  At this point you can locate your trim slider to position it immediately before the first reflection point. This will create the "perfect" trim or truncation.  That being said, there is always some wiggle room either up or down test for what sounds best.  Moving the trim slider to the right past the reflection point can make the bass woolly; to the left the bass can get thin.  What’s nice is that you don’t need to take a new measurement each time you want to change the trim point, or any other setting for that matter.  Just reload the existing left- or right-driver measurement file, make the adjustment and save it as a different file name.

As for attenuation/boost, I too am at a loss on how to use these adjustments. I think that attenuation/boost controls the amplitude, up and down, limits of speaker correction.   I played around with these settings a little and the sound always got worse as I moved away from default.   I also played with smoothing a little but didn’t find any real difference - but I only moved it down to 9 from 10.  So that’s not a real test.

Right now, I've created my best profiles with all the settings in the default positions, except for trim.

I think it would be great if DEQX would document all of these setting in detail.  It would sure be a great help to the Gen 4 community..

I agree with measuring drivers up close is a good practice. Last, (3rd), measurement I’ve been doing at listening position. 2nd measurement I’ve been doing about 1 meter, assuming this is only for time alignment, not sure if this is best.  
trim feature, I tend to get a reflected “bump” at 7 ms.  Would we assume .007 is 7ms? 

I’m also wondering about the PEQ frequency response.  I’m under the belief DEQX creates a flat response through its automatic EQ. Not sure what I’m looking at. Also it doesn’t match the FR I see from REW.

Unit sounds great, would love a little more manual control of crossovers, but overall it’s great.

Yes, 0.007 is 7ms which really isn’t all that bad. It’s bouncing back from the nearest surface - wall, floor, ceiling, etc  Move your trim pointer immediately before the bump and go from there.  Later on, you can reload that same driver file without taking another measurement and move the trim pointer to the right may to 10 to 12ms, save to a different file name and continue with steps 2 and 3 then give that a try.

In v1.4.0 the frequency response heard by the measurement microphone in step 3 in now superimposed on to the the PEQ screen. Whow!  Like you, it’s not what I expected!  It looks like a profile of the Smoky Mountains.  I thought the Pre-8 would flatten it out for the most part - and without any PEQ the profile’s sound left a lot to be desired.  I haven’t pulled out my XLR-phantom-power-to-USB Roland Rubix22 to connect my Earthworks M23R to my REW laptop yet to check it, but you do need to use the same measurement mic placed in the same exact position to do a direct comparison.  I may try it soon.

Well anyway with PEQ, I can flatten the response a bit - quite a bit - returning great sounding results.

In step 2, "Create a Speaker",  I’m finding that somewhere about 40 to 60% of the distance on-axis to the listening position works best for me.  What I found was that if my measurement mic was set up just slightly right or slightly left I would get vastly different relative distance measurements between the bass and the MRT section. The bass panel would lag the MRT section anywhere  between 0 to over 100 mm.  I’m not sure if the following way is correct, but after much trial and error, measuring each side accurately as possible with the measurement mic set the same distance and pointing at the same exact place on both the right and left speakers, I found that if one side has a very low relative distance number, say under 30mm and the other is much higher, I will ultimately manually set the speaker with the higher distance to match the speaker with the lower distance.  Right now, I’m listening to a profile with both speakers’ bass panels set to 27mm and is among the best sounding profile created to date, BUT I still think that there is still a better sounding profile to be had either with another software update or me finding that ah-ha moment with taking measurements - I thinking probably both now. 

Subwoofers are another story.  You can see some very unrealistic distance numbers here - especially if you has multiple subs.  In the past Alan said to take a physical measurement of the distance of your subs to you main speakers then add 0.6m then override the sub relative distance with this new number.  This procedure really doesn’t work if you have multiple subs in different locations. For this reason you can only input your measurements as a WAG (wild ass guess).  I did notice however that the relative distances returned in step 2 for my subs appear to be somewhat realistic now in v 1.4.0, and for that reason I went with that.  The bass seems to be quite good and tightly integrated.

There is not much to say when doing step three, "Creating a System".  Set the measurement microphone where your head would be at your listening position and take the measurements.  If all goes well and you sit down for that first listen, you either say OMG that sound unbelievably great OR that’s just down right awful!

The first thing I do now after I create a new profile is listen to track 2 of the Sheffield Labs XLO Test CD to quickly make sure that everything is totally in phase.  I do this because there was a bug in the software (version 1.37.21) where the final profile had the right and left speaker drivers out-of-phase more often than not. Alan said that they were aware of the issue and had me jump on their development server to test the fix.  I was please to find that all the test profile that i created were totally in phase; however, I did manage to create a couple of profiles recently under v1.4.0 that were out of phase.  For that reason, I just deleted everything and started over again and all profiles created since has been in phase.  I’m not totally sure but when it happens during the calibration process, but I think the out-of-phase situation occurs during step 2.

Bottom line is that there is a lot of time and work involved in properly implementing a four-, six-, or eight-channel BETA DEQX Pre-8.  It’s not for the impatient.  I’ve done hours of work just to take one listen and then delete everything and start all over again - and moreover with every new software release you will need to do it all over again. But so far, like you I’m sure, I think that is the most technologically advanced piece of audio equipment ever produced.  It’s still being developed and I always look forwarded to next software update and ultimately, the final product.