Is DEQX a game changer?

Yes, an excellent post by Bombaywalla, and an excellent response by Michael (Swampwalker). Thanks, gentlemen!

I am in full agreement with both posts, aside from what I believe is an inadvertent and minor misstatement in Bombaywalla's post:

A linear power supply is expensive from a power dissipation perspective - you have to design its max voltage for the max peak voltage of the program material but in normal operation the linear power supply operates mostly at the average voltage of the program material. The difference in the peak & average voltage is dissipated as heat. Of course, you don't know what the max voltage of the program material is so you have to over-design further leading to more heat dissipation.

Shouldn't it be the output of the amplifier that has to operate mostly at the average voltage of the program material, not the output of the power supply? With the difference between the average output voltage and the voltage supplied to that stage (which as you indicated has to provide headroom relative to the maximum anticipated output voltage), multiplied by current, corresponding to the heat dissipated in the output stage, not the power supply? Although the heat dissipated in the power supply will also vary with current demand. And although there are a few amplifier designs in which the output voltage of the power supply is actually varied among a number of discrete levels as a function of signal level, some of Bob Carver's older designs being examples.

Again, though, an excellent and informative post. Thanks!

Best regards,
-- Al

Drewan & Ptss, thanks for your latest comments, which are valued as always.

Pete, a point to keep in mind is that a downside, or at least a potential downside, of the extraordinarily wide bandwidth of your Spectral equipment can be expected to be increased sensitivity to any RFI/high frequency interference that may find its way into the circuitry of those components, compared to components having more typical bandwidths.

Also, as I pointed out earlier the PreMate, which was well reviewed by Kal and has been found to be beneficial in Bruce's (Bifwynne's) very high quality system, uses a switching power supply.

Finally, it would seem understandable that the internal real estate and the additional power required to support inclusion in the HDP-5 of the touchscreen and its associated CPU and other circuitry could very well have necessitated going to a switching power supply.

Best regards,
-- Al

P.S. to my last paragraph above: Bruce, I think that in my step response plot (the expanded version, covering the first 10 ms) there may be a natural tendency to judge the time alignments of the drivers based on the zero crossings of the waveform, which would be misleading and cause the woofer outputs to be interpreted as being excessively delayed from those of the other drivers. However that would not be a meaningful comparison, because the slopes (risetimes and falltimes) of the woofer outputs will inherently be far slower than those of the other drivers, due to the limited bandwidths of both the woofers and their associated low pass crossover network.

So a more meaningful way to assess that plot would be to compare the points at which the tweeter outputs, the mid-range outputs, and the woofer outputs BEGIN to appear. Which in turn, for the woofers, would be where the slope of the initial output from the mid-range appears to begin to slow.

In this case the location of that point is made a bit ambiguous by the glitch at 0.7 ms, but it is clearly somewhere around the time of the first negative peak in the output of the mid-range, somewhat less than 1 ms after the start of the initial sound arrival (that being from the tweeter, of course). And as I indicated earlier, that delay between arrival times from the woofers and the other drivers will be considerably greater at the closeup mic position that was used for the speaker measurements than it would be at the listening position, due to the much smaller angle between the drivers as perceived from the listening position.

Best regards,
-- Al

Andrew (Drewan) thanks for the clarification. That would certainly explain why your group delay plot appears as it does. Also, I note that the excursions in the plot appear to be greatest in the vicinity of what is probably the crossover point between the two drivers. Which makes sense, as in that vicinity the measurement mic would be receiving comparably strong outputs from two different drivers which are at two different physical locations.

Also, since DEQX was providing for you a lot of the optimizations that would be provided by the manufacturer in the case of speakers such as mine and Bruce's, the criticality of your speaker measurements being as accurate as possible (i.e., taken outdoors) figures to have been increased correspondingly.

Best regards,
-- Al

I've performed the experiment I indicated I would be doing to investigate the glitch at 0.7 ms which was evident in the measurements I posted the other day, as well as in the measurements I described taking from other distances.

What I did today was to measure one speaker in its normal position, from a distance of 30 inches, with and without a boom on the mic stand. The resulting impulse response plots made clear that the glitch had indeed been caused by a reflection from the mic stand. However, in comparing today's results with and without the boom, there was little or no difference in the step response at or near that point in time. Also, despite the much greater room reflection content of today's measurements compared with the ones I made the other day with the speakers in the center of the room and with acoustic panels placed as shown in the photos I posted, aside from the deep bass region the frequency response plot derived from today's measurement with the boom matched remarkably well with the frequency plot taken from the same distance the other day (without the boom, with the speaker in the center of the room, and with the acoustic panels in place).

So, thankfully, I can proceed without having to re-do the measurements, which in my case involved a laborious 5 hour effort. :-)

Best regards,
-- Al

I’ve spent some time studying the measurements I previously described having taken, and observing on the computer screen the results of applying various window durations to them. As a result I’ve chosen two specific correction filters to evaluate sonically in the coming days. I’ve uploaded jpg files depicting those filters, which can be viewed here. There are ten files, five for each filter, depicting the corrections in terms of frequency response, group delay, impulse response, step response, and phase response. In interpreting them, be sure to take into account the scales marked on the vertical axes. (To see the markings clearly, click on the image thumbnails to expand them). The measurements, window durations, and correction limits that were used in creating these filters are described below.

The ways in which I narrowed down the many possible correction filters to these two are as follows:

As you may recall, I had performed measurements of each speaker at distances of 30, 36, and 42 inches, with the grilles removed, and also at 36 inches with the grilles in place. The grilles appeared to make essentially no difference.

I found that both 36 inch measurements of the right speaker had a huge group delay spike at about 420 Hz, which was not helped significantly by smoothing, and which did not appear in the 30 or 42 inch measurements of that speaker, and which did not appear in any of the measurements of the other speaker. I have no idea why that occurred, as the two speakers and the mic were positioned identically when they were measured, within perhaps 1/8” or less. The only variable that seemed to be present, which in turn seems very unlikely to have anything to do with that spike, is that the design of the speakers is such that their rear surfaces, rather than being parallel to the front baffle, or being otherwise identical between the two speakers, are mirror-imaged at an angle such that the side of each speaker that is closest to the other speaker (in their normal positions) is 1.5 inches shorter than the other side. In any event, due to that spike I eliminated the 36 inch measurements from consideration.

I experimented on the computer with windowing of the 30 and 42 inch measurements at three different points, each of them just prior to what appeared to be significant reflections or increases in reflections, at about 13, 17, and 21 milliseconds. The 21 ms window resulted in significant frequency response wiggles in the 500 Hz to 1000 Hz area, so I eliminated that choice.

The 13 and 17 ms window durations provided results that looked fairly similar, but 13 ms (which terminated just prior to what I’m pretty certain was a ceiling reflection, based on its timing) looked slightly more promising. So that’s what I went with, for both measurement distances, that also being exactly what Andrew (Drewan77) had suggested the other day after looking at the measurements I had posted.

The (approximately) 13 ms window duration (actually 13.1 ms for the 30 inch measurement and 13.2 ms for the 42 inch measurement) corresponds to 7.4 ms after the initial sound arrival for the 30 inch measurement, and 6.7 ms after the initial sound arrival for the 42 inch measurement.

Regarding the correction limits I set, I used the default amplitude limits, which in turn were not called into play at all within the frequency response limits that I set. For both correction filters, Filter 1 corresponding to the 30 inch measurements and Filter 2 corresponding to the 42 inch measurements, I set frequency response limits such that corrections were only performed between about 400 Hz and 10.5 kHz. Those choices being made taking into account suggestions from both Nyal Mellor of Acoustic Frontiers and Alan Langford of DEQX to be conservative in dealing with the top octave, and to avoid correcting further into the bass region than seems reasonable in relation to the window duration. With the latter determination being made in the manner I described in my post dated 6-22-15. And, also, taking into account a presumption I made that both the high frequency and low frequency limits should be chosen such that abrupt discontinuities in frequency response are not introduced at the limit points.

Finally, in deriving the correction filters all parameters which I haven’t mentioned were used at their default values, including 100% smoothing.

Best regards,
--Al

Thanks, Andrew (Drewan), for your always valuable inputs.

For the time being, at least, I’ve completed my assessment of the two correction filters I described in my previous post. The clear winner was filter 2 (the one that was created from measurements taken at a 42 inch distance), vs. filter 1 (created identically except from measurements taken at a 30 inch distance), and vs. bypass mode (which was outperformed by both filters), and vs. the several filters I had tried some time ago which were created from measurements that were compromised by close placement of the acoustical panels I used.

Most of the evaluation was performed with classical music, which is what I and my wife primarily listen to. Some rock, pop, and folk was also included. The degree of the differences between filter 2, filter 1, and bypass mode varied widely depending on the recording, ranging from barely perceptible to quite dramatic.

Perhaps most notable among the differences that I and my wife perceived were on some recordings having overly bright string sound, including some string quartets as well as symphonic recordings. Those became much more enjoyable with the filters engaged. Not because the sound was dulled down, but because there seemed to be increased detail and improved definition in the upper midrange and lower treble, as opposed to a more homogenized presentation of those notes, which in turn resulted in the brightness being less objectionable. I recall that some time ago, either in this thread or in the “sloped baffle” thread, Bombaywalla had commented that time coherence will provide benefits along those lines. Both this experience and many previous experiences I’ve had comparing sonics between my speakers and my Stax electrostatic headphones have me convinced that he was right.

Room corrections, which I haven’t yet addressed at all, are next in the queue!

Best regards,

--Al

An update for those who have been following my progress with the HDP-5: There will be a further delay until I perform the room corrections, due to an unrelated issue that has arisen in my system.

I've recently been noticing significant loss of definition on high frequency percussion, especially on high frequency piano notes. That has not been evident with my Stax headphones, however, just via the speakers. Since the headphone amp is driven by an output of the DEQX, in bypass mode of course, that pointed to my power amp as being responsible. And sure enough, when I lightly tapped on its tubes with an eraser, with the amp powered up, I found that one of its four vintage Sylvania 6SN7GTB's had become highly microphonic.

I have a number of other 6SN7GTB's on hand, but I don't want to use them for anything involving critical listening, such as the room corrections, as I had tried them in the amp a couple of years ago and didn't care for their sonics. So I'm ordering some additional tubes ("tubes" plural, as I'll want to replace the corresponding tube in the other channel with one that matches).

I'm pretty certain, btw, that this issue would not have affected the speaker measurements or speaker corrections I have performed. The problematical tube is only in the path of one channel, and the speaker measurements and correction profiles turned out to be very similar for the two speakers. Also, the tube still measures fine on my Hickok tester, and when I performed the speaker measurements the speaker was considerably further away from the amp than when in its normal position, and was pointed in the opposite direction of the amp.

Best regards,
-- Al

Thanks, Roscoe.

When listening to LPs via the Stax headphones, which are of course very revealing, the DEQX continues to seem to me to be perfectly transparent. Which is amazing, of course, considering the A/D and D/A conversions it puts into the signal path.

With CDs, for which I'm now using the DEQX as the DAC and my Bryston BCD-1 just as a transport, on some recordings it has provided a slight benefit compared to using the BCD-1's analog outputs routed to the Stax headphone amp via tape outputs of my previous preamp (a Classe CP-60), while on others there seems to be little if any difference. I suspect, though, that eventually I'll want to experiment with some different digital cables, and also different cable lengths, between the BCD-1 and the DEQX. I'm presently using an inexpensive Mogami AES/EBU cable, in a 6 foot length.

How has your HDP-4 been working out lately?

Best regards,
-- Al

Hi Unsound,

No, I still haven't addressed the room correction part of the effort, and I probably won't during the next few weeks. The microphonic tube issue I mentioned a few weeks ago was readily resolved with the purchase of a couple of NOS tubes. But subsequently an unrelated hardware issue arose in my system, which among other things has prompted me to put into motion a phono stage upgrade I've had in mind for some time.

So I've just placed an order with Herron Audio for his highly regarded VTPH-2 phono stage. It will replace what I have been using for that purpose for many years, the phono section of a vintage Mark Levinson ML-1 preamplifier (accessed via the ML-1's tape outputs). I should receive the VTPH-2 in about two weeks or so. I've also ordered an Adona isolation platform for it to sit on, and I may have to also make some changes in the system's power distribution setup.

I'll then want to spend some time familiarizing myself with the sonics the VTPH-2 will provide, and perhaps also give it some additional burn-in (although the burn-in and test Keith Herron provides prior to shipment amounts to 48 hours, according to the manual).

After the dust settles on all of that, I'll resume the process of dialing in the DEQX.

Best regards,
-- Al

Not yet, Unsound. I should be addressing room corrections next week. Meanwhile my listening has been focused on the upgrades I alluded to above, most notably the new phono stage, and I've also had to devote a lot of time to activities unrelated to audio.

BTW, you can check out updated photos and descriptions of the recent system changes in my system description thread.

Best regards,
-- Al

Hi Unsound,

I presume that your Thiel speakers are designed to be time coherent, and if so presumably the DEQX speaker calibration function would not be beneficial to you. I would expect, however, that the equalization functions a DEQX can provide would allow you to eliminate the speaker's line-level analog equalizer from your system, and avoid the sonic tradeoffs that are inherent in that equalizer. (As you've probably seen in the past, Larry Archibald commented on those tradeoffs in this 1989 review).

And even though I haven't yet addressed room corrections in my system, I don't doubt that with sufficient tweaking (which can be done in real time, while listening), that function can provide significant benefit in most circumstances.

Also, as you no doubt realize a DEQX, depending on model, may be able to serve as a preamplifier, as well as providing remarkably transparent A/D and D/A converter functions.

Whether all of those potential benefits would to you be worth the not inconsiderable cost is, of course, your call. (And figure on $745 for the optional reference microphone kit on top of the cost of whichever DEQX model you might choose. Having some experience with both inexpensive and high quality professional mics for other applications I would not rely on the inexpensive mic that is included in the base price of the unit).

FWIW, in my own case I'm glad I made this investment, notwithstanding that as I've described earlier in the thread it has in my case turned out to be a bit of a science project at times. Albeit one that to me has been interesting and educational.

Best regards,
-- Al

I’ve posted a couple of screenshots of the room corrections/equalizations that I’ve settled on, at least for the time being, at this link.  One depicts the full spectrum, and one has an expanded view covering just up to 500 Hz. The room corrections I’ve settled on just extend up to 400 Hz, which just by coincidence happens to be the lower limit I had chosen for the speaker calibrations, for which I previously provided screenshots.

As you can see in the listings on the left side of the screenshots, I had made a total of 11 different measurements of each speaker, all in the vicinity of the listening position but at various distances and heights (all of them centered between the speakers). (“D” in the measurement position listings indicates distance from the plane of the speakers in feet, and “H” represents the measurement height in inches). Eventually I chose the measurements that were performed at a 12 foot distance and a 36 inch height to work with in deriving the corrections. The screenshots show those particular measurements, together with the equalization curves I’m now using.

The 12 foot distance corresponds just about exactly to my usual listening distance. The 36 inch height is about 2 inches above the mid-point between the two tweeters that are approximately centrally located on the vertical axis of the baffle area of my speakers. That height is also within a couple of inches or so of the height of my ears and my wife’s ears, when we are seated in the (quite low) Wassily chairs you can see in my system description photos.

You’ll note in the full spectrum screenshot that the upper treble rolls off significantly at the listening position. That occurs even though, as you can see in the speaker measurements I posted some time ago, the speakers themselves have a significant rise in frequency response in the top octave (the magnitude of that rise, remarkably, being similar to the magnitude of the dip at the listening position), and it occurs even though a mild dip in the 5 to 10 kHz area which the speakers exhibited in the close up measurements was pretty much compensated for in the speaker calibration process. But I’ve nevertheless chosen not to introduce any room correction equalizations in the treble region, after taking into account recommendations I had received from both Nyal Mellor of Acoustic Frontiers and Alan Langford of DEQX suggesting that it is generally best to leave the upper treble region alone, as well as my ongoing satisfaction with the tonal balance of my speakers in that region, as well as the impression I’ve formed over the years that a system which measures flat at the listening position throughout the treble region will sound too bright to most listeners.

Another point I found to be particularly striking, when I compared the measurements I had taken at the various distances, is that as might be expected the numerous peaks and valleys in frequency response occurring in the bass region as a result of room reflections will be located at frequencies that differ significantly depending on the specific measurement position. Meaning that basing a correction on a measurement performed at a distance that is a foot or two greater or less than the listening distance can result in an equalization boost being applied to a peak instead of a valley, or an equalization cut being applied to a valley instead of a peak. Particularly if one is aggressive in trying to correct for relatively narrow peaks or valleys.

Summing up my experience with the DEQX to this point, I would say that it has provided significant benefit to me as a result of its speaker calibration function, its room correction function, and as a preamp, and as a DAC. As I alluded to in an earlier post, speaker calibration seems to be especially beneficial in my system on recordings that are sonically mediocre, or worse. @Bombaywalla had said some time ago, either in this thread or in the “Sloped Baffle” thread, that that can be expected to be a consequence of improvements in time coherence, and my experience with the DEQX appears to confirm that.  

So I am glad I chose to make the substantial investments of money and time that have been involved. The time factor presumably would have been considerably less if I had chosen to utilize the DEQXpert service, but while I have no way of knowing whether the results of doing that would have ended up being better than, worse than, or similar to what I have accomplished, I feel that the learning experience resulting from doing it all myself has been sufficiently valuable that I don’t regret not having utilized that service.  

Next on my list, now that I have added the excellent and LOMC-capable Herron phono stage to my system (as I mentioned in an earlier post), is that in the next couple of weeks I’m planning to purchase an LOMC cartridge, to use in place of or as an alternate to my Soundsmith re-tipped vintage Grace F9E moving magnet. I’m planning on getting a Dynavector 17D3, in part because of the excellent reviews it has received over the years, but particularly because member @Rodman99999 had mentioned to me in a thread a while back that it is “magic” in the vintage Magnepan Unitrac tonearm he and I both use.

Thanks for reading. Best regards,

-- Al

Thanks for the comments, gentlemen. Sounds like a plan, Roscoe!

Yes, I would think that a DEQX will generally make speaker positioning at least a bit less critical than it would otherwise be, as a result of the corrections and equalizations it can introduce with seemingly no sonic downside.

Unrelated to the DEQX, I’ve made one change to what I said in my 12-2-2015 post. Instead of purchasing a Dynavector 17D3 phono cartridge, based largely on the many accolades that have been posted here recently I’ve purchased an Audio Technica AT-ART9. I just started listening to it today, with promising results. But the reported experiences seem to indicate that it improves considerably during the first 50 or so hours of breakin, so I’ll wait until I’m significantly into that period before attempting to fine tune the various tonearm adjustments.

Roscoe, best of luck as you proceed. And thanks very much for the kind words you posted recently in another thread.

Enjoy! Best regards,
-- Al

Both the HDP5 I own and the Premate+ are described as using low noise switching power supplies, and I suspect that the same supply is used in both.  The power supply in the HDP5 is a commercially available sealed unit capable of supplying a maximum output of 30 watts.  Since it is a switching supply and since the power drawn from it is undoubtedly somewhat less than 30 watts I would expect that the unit's AC consumption is not a great deal more than that amount.  Which means that it won't run particularly hot, and will just get a bit warm as I have found with my HDP5.

That said, 0.35 inches of clearance by the vents does sound a little uncomfortable to me.  My guess is you'll be ok over the long term, but I'm unsure.

I have no specific knowledge regarding your second question.

Good luck.  Regards,
-- Al
  

Todd, if I understand correctly you are supplying the Premate with a digital input, and when you connect the analog "main speaker" outputs of the Premate to your (monoblock?) amplifiers everything works fine. But when you connect the analog "main speaker" outputs of the Premate to analog inputs of your preamp, and connect the preamp’s analog outputs to your amplifiers, you get nothing.

If that is a correct interpretation, the only possibilities I can think of (assuming you are using similar settings of the Premate in the two cases) are that the preamp’s input select switch is not set to the right input, or there is a connection problem, or the preamp has a problem.

In any event, as you alluded to I suspect the cause of this problem will turn out to be something simple.

Wondering if any could share their connection setup.

FWIW, I use my DEQX HDP-5 as my preamp.

Good luck, and enjoy! Regards,
-- Al

FWIW my HDP-5, which has superseded the HDP-4 as DEQX’s top-of-the-line model, uses a switching power supply. I don’t know if it is the same supply that is used in your Mate, but it is an XP Power model ECL30UD02-S. Here is the datasheet for that series of supplies.

I use the HDP-5 as my preamp. I have never sensed that it is causing any glare whatsoever in any part of the spectrum. I am connecting my phono stage (a Herron VTPH-2) to it via unbalanced RCA cables. I am connecting my main digital source (a Bryston BCD-1 CD player used just as a transport) via an AES/EBU digital cable. Those are my two critical sources. I am connecting the HDP-5 to my power amp via unbalanced RCA cables.

I don’t recall what connectivity the Mate provides, but if you are using balanced analog connections to its inputs or outputs perhaps a contributing factor to the issue you described might be an internal balanced/unbalanced converter stage or transformer needed to interface to its presumably unbalanced internal signal path. Or if you are connecting to a digital input, perhaps waveform degradation due to signal reflections resulting from small impedance mismatches is occurring, resulting in jitter that might be improved by changing the length and/or type of the digital cable.

That’s about all I can think of in response to your question.

Good luck. Regards,
-- Al

Andrew (Drewan77), congratulations on the new HDP-5!  And thanks once again for the many valuable inputs you have provided to this thread.

Best regards,
-- Al
 

Good info, Kingrex.  Thanks!

BTW, although it undoubtedly has no relevance in the case of your Mate, I'll mention that according to a communication I had with Alan Langford of DEQX last year, incorporation of Larry's mods into a unit that is under factory warranty will invalidate that warranty.

Best of luck as you proceed.  Regards,
-- Al
 

Ozzy, looking at the photos of your setup and room in your system description thread I suspect that you would want to do something along the lines of what I did for the speaker calibration measurements. Namely moving the speakers to the center of the room for the measurements (one at a time, of course). And perhaps considering doing as I did in placing large sound absorbent panels against the nearest reflective surfaces (e.g. the side walls and the floor), when the measurements are performed.

In my case the resulting impulse response measurements were not nearly as "clean" as, for example, the results linked to earlier in the thread by Andrew (Drewan77), which he obtained by measuring outdoors. That in turn necessitated that I "window" the impulse response many milliseconds earlier than he did, which in turn resulted in me choosing to just implement speaker corrections at frequencies above 400 Hz, rather than down to 200 Hz which seems to be the optimal goal. (Lower frequencies are addressed in the room correction process, based on measurements at the listening position with the speakers in their normal positions).

So while I suspect that my results would have been at least a little better if I had measured outdoors, as I said in a post dated 12-2-2015 on the previous page of this thread:

Summing up my experience with the DEQX to this point, I would say that it has provided significant benefit to me as a result of its speaker calibration function, its room correction function, and as a preamp, and as a DAC. As I alluded to in an earlier post, speaker calibration seems to be especially beneficial in my system on recordings that are sonically mediocre, or worse. Bombaywalla had said some time ago, either in this thread or in the “Sloped Baffle” thread, that that can be expected to be a consequence of improvements in time coherence, and my experience with the DEQX appears to confirm that.

Also, regarding your own system, one thing I would wonder about is how the phase-related processing of your BSG QOL would interact with the timing corrections DEQX implements.

Finally, I should mention that for my critical sources I am using unbalanced analog inputs and outputs on my DEQX (as well as digital inputs), and so I have no meaningful knowledge of the sonics of its balanced analog inputs and outputs, one or both of which I suspect you might be using if you were to introduce a DEQX into your setup.

Regards,
-- Al

Hi Andrew,

I’m not sure I understand the first paragraph of your post just above. If you are using an external DAC via the HDP-5’s balanced analog inputs, aren’t you then still using the HDP-5’s built-in DAC as well, and also inserting the HDP-5’s A/D converter function into the signal path?

Thanks.  Best regards,
-- Al

Jeff, yes, all of the questions you raised in your previous post seem to me to be logical and valid concerns. Although as you’ve probably seen earlier in the thread many DEQX users having very high quality systems consider the DAC function and the overall transparency of their units to be excellent.

Also, while I’m not in a position to comment on how the sonic quality of the HDP-5’s D/A converter compares to the sonic quality of its A/D converter, as an electrical engineer who has designed several A/D and D/A circuits over the years (for defense electronics), I can say that generally speaking it is a considerably greater challenge to design a good performing A/D circuit than a comparably good D/A circuit, assuming both are required to perform at similar sample rates and resolutions/bit depths.

So I wouldn’t be surprised if keeping a DEQX’s A/D converter out of the signal path would have the potential to provide greater benefit than keeping its D/A converter out of the signal path. Although, again, I’ve personally been very pleased with the transparency and overall performance of my HDP-5 with both in the signal path.

Best regards,
-- Al

Ozzy, to find the manual go to DEQX.com, and at the top click "Owners," and then "Upgrades" in the drop-down menu that appears.  Scroll down to the bottom of the box that will appear, and you'll see a link for the manual.

Good luck.  Regards,
-- Al

Ozzy, no, I didn't use the DEQXpert service.  I did, though, receive a useful 1 hour walk-through of the calibration software that was provided via web + phone by Nyal Mellor of Acoustic Frontiers, from whom I purchased the unit.  And I then spent several months patiently and methodically further familiarizing myself with everything, performing measurements, re-doing some measurements, tweaking settings, listening to a variety of material, etc., before I considered the process done.  And my application is simpler than most of the others that have been mentioned in the thread, with just a single pair of speakers, a single stereo amp, and no sub.

My experiences during that process are documented in posts on pages 6 through 9 of this thread.

The amount of time I invested in this process would obviously have been far less if I had chosen to utilize the DEQXpert service, and obviously I have no way of knowing whether the results of doing that would have been better than, worse than, or similar to what I ended up with.  But personally I feel that the learning experience resulting from doing it all myself was sufficiently valuable that I don’t regret not having utilized that service.

Good luck.  Regards,
-- Al
 

Hi Ozzy,

I can't answer your question about the download CD you referred to. I was provided with the calibration file for my microphone by the dealer from whom I purchased my HDP-5 together with the mic kit (see below), Nyal Mellor of AcousticFrontiers.com. The microphone calibration file would have a .mic file extension, and if it is not on the CD I would expect that the person you bought the Premate from should be able to email it to you. Or perhaps DEQX could supply it, if you were to ask them via their support link.

Once you have that file on your computer, the mic would be "installed" into the calibration software by selecting "Install Microphone" under the "File" menu, and then navigating to the calibration file for the particular mic.

That said, it appears that the EMM-6 is the cheapie (approx. $50) microphone that is included with the DEQX units as standard. I chose the much more expensive ($745) Earthworks "Reference Calibration Kit" alternative. Based on my experience with mics in those two price ranges for unrelated applications I suspect that you would get significantly better results if you were to purchase the Earthworks kit. A good source from whom to purchase it would be Nyal Mellor:

http://store.acousticfrontiers.com/Digital-Signal-Processors/deqx-earthworks-m23-cal-kit.html

Good luck. Regards,
-- Al

P.S: I just took a look at the CD that came with my HDP-5. Viewing the files on it via the File Explorer in Windows I see that there is a folder called "Microphones," and under that "Dayton," and under that "Dayton" again, and under that "6780EMM6-INV.mic" among many other listings. If you can find that file on your CD, copy it to a location on your hard drive and proceed as I indicated above.

Regards,
-- Al

Hi Ozzy,

I believe these paragraphs from version 2.98 of the manual may explain the issue you referred to just above.  (I assume version 3.02 of the manual provides similar information, but I haven't looked for it there):

Note - when the Save All to DEQX button is clicked, all four profiles from the open configuration are saved to the DEQX.  Since all profiles of a new configuration are disabled until you change them, if you install a correction filter in just one profile of a new configuration, then save it to the DEQX, the other profiles of the DEQX will be disabled, irrespective of what those profiles may have been before. 

Note that the DEQX Configurations saved to the DEQX unit have been generated in 'non-real time' i.e. they only affect the processing by the DEQX unit once the Save All to DEQX button is clicked.  This contrasts to changes made in the DEQX Control Panel, which affect the processing on the DEQX in ‘real time’ i.e. any change made in the DEQX Control Panel instantaneously affects the DEQX. For more on this subject see IO Manager in the Reference section.

To save a DEQX Configuration on your PC (not on the DEQX), click on the Save button once you have set up the DEQX Configuration. This also saves the measurement(s) and correction(s) you have created.

HTH.  Regards,
-- Al
 

No, but when you create a profile and do a "file/save" (which updates the .mzd file that is currently open on your computer) or a "file/save as" (which creates a new .mzd file on your computer), and then at a later time you create an additional profile and do a "file/save" or "file/save as," when you "save all to DEQX" what will be uploaded to the DEQX are the four profiles that exist (or don’t exist) in the .mzd file that is currently open.

So you can create a profile, store it in a .mzd file via a "file/save" or a "file/save as," and at a later time create an additional profile in that same file, and then a "save all to DEQX" will upload both profiles to the DEQX.

Regards,
-- Al

Blang11, thanks for providing the thorough and extremely well composed report.  Glad it worked out so well.  Enjoy!

Best regards,
-- Al

Hi Steve,

As usual Andrew (Drewan77) provides great answers, with which I agree completely.

I too purchased the M23 as part of the $745 DEQX "Reference Calibration Kit." Mainly out of curiosity I subsequently requested a calibration file corresponding to the serial number of the particular mic from Earthworks, via their website, and it was in a plain text format rather than the proprietary DEQX format supplied with the kit that is necessary for importation into the DEQX cal software.

Regarding your questions about how to best configure the system, I would just add to Andrew’s comments that using a particularly long digital cable could conceivably increase susceptibility to ground loop issues between the CDP and the DEQX, which in turn might adversely affect timing jitter at the circuit point within the DEQX where D/A conversion is performed. So when the time comes it may be worthwhile to compare sonics with and without a cheater plug temporarily applied to the CDP’s power plug, to defeat its safety ground connection. That would break any ground loop that may exist between the CDP and the DEQX, and allow you to determine if this possibility is an issue.

BTW, for the benefit of others who may read your post and may wonder, I’ll mention that I assume the word "apron" was intended to be "approx."

Best of luck as you proceed. Regards,
-- Al

From page 20 of the manual, referring to the single digital output of the PreMate and PreMate+:

This is the same signal as provided on the Main Speakers analog output. It carries the corrected audio signal for the speakers, optionally with limit filters for subwoofer integration. It can be used to connect an external DAC instead of using the internal DAC of the DEQX.

I would infer from this statement that the digital output is volume controlled, as well as being subjected to all of the corrections and other signal processing that are applied to the main analog outputs, other than D/A conversion.

I’ve never used the digital outputs of my HDP-5. But what I’m wondering is **if** the PreMate was purchased used, and **if** the previous owner was just using it for subwoofer duties, he may have disabled the main analog outputs in the configuration he set up. Which presumably would have also disabled the digital output, given the paragraph I quoted from the manual.

In that case a computer would have to be connected to the DEQX and the calibration software would have to be used to revise the configuration.

Best regards,
-- Al

Hi Ozzy,

I’ve never adjusted the gain on my HDP-5 via the Control Panel, and I believe that increasing the gain via the Control Panel would reduce the headroom that is available for frequency response boosts that may be introduced for purposes of speaker calibration, room correction, or equalizations that may be desired. In turn resulting in the possibility of clipping the output circuits of the DEQX. I believe that would apply to both the analog and digital outputs.

My present amplifier (a Pass XA25) has relatively low gain (20 db), and what I have done to add some gain is to change the internal jumpers in the DEQX which control the voltage range of its analog outputs that I use to drive the amp. See page 166 of the manual. Relative to the default position of the jumpers that results in a 4.9 db gain increase on the single-ended outputs I use, and I’m pretty certain that is accomplished without any sacrifice in headroom. Those jumpers have no relevance to digital outputs, however.

Best regards,
-- Al

Hi Steve,

For the speaker calibration measurements what I initially tried was placing large sound-absorbent panels behind and to the sides of the measurement mic, with each speaker having been moved to the center of the room for purposes of that measurement.  The panels were placed something like one or two feet from the mic.  That did NOT provide good results, because reflections from the panels themselves, while small in amplitude, were so close in arrival time time to the direct sound that the "booth" did more harm than good.

I then placed the panels against the nearest reflective surfaces.  One being a stone fireplace on the wall on one side, and the other being a large piece of furniture on the other side.  That was definitely worthwhile in my case, as the room is only 13 feet wide and the piece of furniture (actually an antique radio/phono console) extends out about 2.5 feet from the wall on that side.

In your case whether doing something similar would be worthwhile presumably depends on the distance to the nearest walls or other large surfaces, and their reflectivity.  Perhaps consider trying it initially without any such measures, and see on the resulting impulse response/time-domain plots how many milliseconds from the direct sound arrivals you can "window" the measurements, before reflections become prominent. 

In my case, if I recall correctly the duration of the "window" I applied to the measurements was limited by reflections from the ceiling, occurring about 8 ms after the direct sound arrival.  (Reflections from the floor were not significant because in addition to it being covered with a thick rug, when making the measurements I had placed a pillow on it, directly in front of the speakers). 


The panels I used were these:

https://www.bhphotovideo.com/c/product/401266-REG/ClearSonic_S5_2_S5_2_Dark_Grey_SORBER.html

Good luck!  Regards,
-- Al

Thanks for the update, Steve. I’m curious, though, as to why the use of two monoblocks (I assume you meant per channel, in a biamp configuration) would make any difference with respect to the ability of the DEQX to improve the time coherence of the speakers.

Best regards,
-- Al

Sincere condolences on the loss of your friend, Steve.

Not moving the speakers to the center of the room for the speaker calibration measurements certainly figures to be a reason the PreMate didn’t provide much improvement. I had pointed out on 9-5-2018 in the DEQX thread you had started that might be an issue given the 410 pound weight of your speakers.

Again, my condolences. Best regards,
-- Al