Archimago on SPDIF

Archimago seems to want it both ways.

1) Jitter, caused by decent quality modern devices, is not audible.  (This is a standard audio objectivist position.  A JAES paper written long ago showed that jitter must be 100-1000 times higher than produced by modern digital devices to be audible.)

2) Nevertheless, you should always use USB, if possible, because it has lower jitter.

I will not hide my feelings here.  I hate USB audio interfaces.  And I am angry that Archimago is hypocritically using his platform to promote FUD (Fear, Uncertainty, and Doubt) of SPDIF type interfaces, while at the same time claiming it's an unimportant issue.  This is weasel hypocrisy.  (Note: in other ways, I truly appreciate Archimago's audio journalism.  He is the best I know.)

I hate USB interfaces because they are a product of the tyrannical Computer world, and will be endlessly subject to planned obsolescence, mandatory "upgrades," computer failures, software failures, complicated hardware, firmware issues, IPP, UPP, Wepp and so on.  It might be possible to make those systems needlessly technically better than the alternative, but at the cost of complexity and brittleness.  When things like this work, fine, but the problem is there may be times when instead of working needlessly but slightly measureably better, they simply don't work at all.  That's what I mean by brittle--rather than bending they break.  That could mean dropouts, or it could mean hours of frustrated fiddling followed by another high dollar trip to the computer store to get another mandated upgrade, including possibly a whole new computer.

SPDIF is a simple and open and unchanging interface which is Good Enough.  Period.  (And this applies to AES as well, but not Toslink--always avoid Toslink if you can--though sometimes it is useful for breaking ground loops.)

SPDIF interfaces also inherently permit an endless chain of devices, with essentially zero loss.  That is mainly why I love them.  I used to dream of having power like this.  You can add any number of DSP processors in line, including devices like Behringer DEQ 2496 which are cheap and powerful, and miniDSP.  Devices like these tend to be open, like the SPDIF interface itself, and let you play with them, rather than being a magic box which does stuff you are just supposed to sit down and like.

Generally, with computer interfaces like USB, everything is supposed to be done by your computer, which generally also means proprietary software, such as Archimago's beloved Accourate, but also things like USB Drivers and USB Firmware, which often require updates on the day of purchase, and months if not years into the future.

There is little or no flexibility in the hardware configuration with USB.  You can't chain devices endlessly (or, plug in devices to record the output of other devices--in fact that is the whole reason why things are sometimes supported on USB and not in free and open interfaces--it's all about the IPP).

With USB, your hardware setup is going to be like this: computer connects to DAC over USB, end of story.

Archimago complains that the Oppo doesn't permit SPDIF or HDMI output when USB sources are used.  Yes, that's kind of the nature of the USB beast, generally speaking, and getting around it requires fancy footwork rarely seen (and I wonder if IPP would get in the way also).  Archimago hints at what Oppo should have done for USB, to permit SPDIF and HDMI downstream and it doesn't sound trivial.

Possibly Archimago's endless pushing of USB interfaces is they are the ones best suited, at this time, for multichannel audio.  There are already existing standards for high resolution multichannel audio over USB.  (Which will change every few years, of course, which is back to one of the big reasons I hate USB.)  And multichannel audio is one of his big things.  I think objectivist audiophiles tend to oversell multichannel audio.  I think it's barely worth the effort for music.  I'm working on it myself as a fun extra thing, but only that.  Not worth the bother, at this time, for most people, IMO.)

Just take a look at Archimago's peerless and wonderful review of the Oppo BDP 205, where he shows the performance of the UDP 205 as a DAC.  Here you can compare the performance of his beloved USB and the SPDIF/Coax interface.  Is there a difference?  Yes but smaller than with practically anything else.  We're talking a handful of tiny ticks at the reaching from the -155dB noise floor to about -145dB.

24/96 is also Good Enough.  What is the theoretical S/N ratio of 24 bits?  144dB.

Is there an asynchronous type interface which I like for audio?  Yes.  Ethernet and wifi, which are also basically open interfaces.  Ethernet never locks you down to a specific computer like USB does.  It isn't specifically part of the Personal Computer world.

I don't like using computers directly as audio sources either.  I like dedicated players, streamers that connect via Ethernet, and things like that, not things that are going to tell you that another update is going to be needed today.

I suppose, if you are going to be using a computer as your audio source, not mediated by a player connected through ethernet, etc, then USB is fine, and yes it may be the way to go.  So use USB if it's applicable like that.  I can't say you should never use USB.  I use USB audio solely for doing audio measurements because computer based measurement systems are almost unavoidable.  And every time, it's a pain, because the computer needs an upgrade or whatever.  Unlike traditional audio gear, which is near timeless and universal, computers are cranky and immediately obsolete after purchase.

But you should not be badgered into buying a new DAC simply to attach via USB because "it's better."  Or abandoning non-Personal Computer playback systems to reach that goal.  The alleged benefit in lower jitter isn't worth it.  In fact, it isn't worth anything at all.

While Archimago dabbles in the like of DSD and 24/386, which may require USB (or, gasp, HDMI), he also believes that nobody needs a sampling rate higher than 50kHz.  Well then SPDIF is just fine, again.

It should be understood, anyway, that I truly appreciate the very fine work Archimago continues to do.  In this incredible blind test, involving a large number of listeners, he shows, once again, that the differences between DACs, even much larger and more obvious differences than the ones caused by SPDIF jitter, are so small that they appear to be inaudible in fair tests.  But this, once again, shows the tiny differences caused by the "jittery" SPDIF interface are inconsequential.

Next Priorities

Before putting away the measurement rig, which I've decided I must do at least once a week to maintain sanity, I made one last measurement of the Left channel, with 0 degree calibration file loaded.  (Actually, part of the purpose was to be sure I was loading the calibration file correctly, which requires remembering to tick the box "Use with frequency response" and the "OK.")

It's only slightly different from the uncalibrated response, with some differences above 10kHz and especially below 20 Hz.  To see below 20 Hz or above 20kHz with ARTA, one has to remember to use the scroll or zoom controls on the right side.  It extends to respectable limits beyond.  Until I discovered that, I was cursing the limited 20-20kHz range.

The frequency response graph shows a few problems that should be tackled before moving on to phase linear crossovers, at least with regards to likely audible consequences.

Probably the most important if the rear wall reflection notch at 200 Hz.  I thought I had fixed that by moving speakers farther from the wall.  Indeed, the notch used to be at 250 Hz and bigger.  Also it gets covered up when the region around 100 Hz isn't equalized properly.  But when 100 Hz is equalized to roughly flat, then it is obvious there is a large notch at 200 Hz.  Currently this problem is limited to the left channel for some reason.

It would be nice if I could ameliorate this without going to greater complexity, such as using the midrange driver in the supertweeter box for fill-in just in this notch range.  (That would make the linear crossovers thing even more complicated.)

Another bit that bugs me, but probably not very important, is the depression just before the supertweeter takes over.  That relates strongly to the high frequency crossover.  I possibly need to equalize the Acoustat output around 14kHz where there may be some kind of resonance, so I can make the acoustic crossover "tighter" without causing a huge peak.  That's been the limitation so far, I can't fix the dip without creating a huge peak at 14kHz instead.  That needs to be fixed before trying to make the high frequency crossover linear phase.

Finally, the Linkwitz dip from 2-6kHz is not well done on the left channel.  Instead of a dip, it has a bulge around  4 kHz.  Most of the EQ tuning for that was done on the right channel, and I haven't used separate L and R settings.  I will need to have separate Left and Right equalizations going forwards to deal with things like this.

But meanwhile, I also need to apply time alignment to my two other subwoofered systems, the kitchen and the bedroom.  The time alignment for those systems has never been properly measured or adjusted.  (That was true of the living room system also, until last week.)

I also want to check out the digital jitter resulting from the use of the miniDSP's and other new features, such as my Kanex Pro de-embedder which is now used to extract PCM from HDMI while I am playing SACDs.

Time Alignment using ARTA (description and pictures)

Time Alignment showing microphone at listening position

ARTA running on laptop and Focusrite Scarlett

For many years up until now I had not been satisfied with my methods of measuring time alignment.   Measurement is the hard part.  Actually adjusting the time alignment is trivial, I simply dial in time delay values into the separate Behringer DEQ 2496 units that digitally EQ the signals for my SVS PB13 Ultra subwoofer (the bass below 100 Hz), the Acoustat 2+2's (the midrange from 100Hz to 17kHz), and the super tweeters (two small dome tweeters with response to 40kHz mounted on a wood box).  Now that I am also using miniDSP OpenDRC-DI units to perform the actual crossover functions for each of these 3 ways, I could dial the delays required into the MiniDSP's instead, but I find it easer to make adjustments with the Behringer DEQ units because they don't require a computer to be attached.

Stack of MiniDSPs on top of stack of Behringer DEQ's

Previously I had used the measurement program that is part of my Tact Room Correction System 2.0 Preamp (I don't use the Room Correction itself, but I have often used the RCS measurements).  That is strange and hard to use.  Last year I started using Room EQ Wizard (REW), which is OK for doing loudspeaker+room measurements, but didn't seem helpful specifically for the time alignment.  Just last week I downloaded and registered the ARTA program used by many loudspeaker designers.  For only $100 it is a far better program IMO than REW and after a week of fiddling around with it, I came up with a very intuitive, repeatable, and I believe accurate methodology for time alignment.  The method has two parts:

1) Align the leading edge of the acoustic output of each driver.  I measure each driver separately with the Signal Time Record feature of ARTA.  For midrange and super tweeter, I use a pulse with width "1".  I expand the vertical and horizontal scales of the time record display, and put the cursor at the exact position where the signal begins, as differentiated from room noise.  This is easy to see with the scales expanded, though less easy for the subwoofer output.  I adjust the delay times in my Behringer DEQ's so that the signal beginning times for each driver are the same, within about 0.02 ms for the panels and 0.06ms for the subwoofer (which is hard to see as clearly).  When measuring the subwoofer, I change the pulse width to 1000, otherwise it doesn't show up at all.

Leading edge of Acoustat signal

The ARTA Signal Time Record is an unprocessed display, similar in principle to a storage oscilloscope.  It can be used with a simple up and down pulse, which results in the measurement shown above.

The Impulse and Step displays are derived from longer term signals using FFT, and therefore show the room response as well as the speaker response, and not just from the very first instant.  For the first phase of time alignment, we need to examine ONLY the very first instant of the signal.

2) For the subwoofer and electrostatic panel alignment, I further fine tune the delay adjustments to either maximize the output at the 100 Hz crossover frequency, or minimize the output with the polarity of the panels reversed.  I think the minimization method is the better one, and IIRC was specifically recommended by Linkwitz himself.

When the polarity of the panels is reversed, the subwoofer and the panels are "cancelling" each other, because normally the LR4 crossover has them "in phase" at every frequency, especially the crossover.  I was pleased to achieve deep null using the cancellation method.

Deep Null at 100 Hz Crossofer Frequency (Green Line)

To measure the output at 100 Hz, I run an Impulse measurement with ARTA, and then run the Frequency Response and Distortion analysis.  This gives a remarkably stable frequency response graph, so stable that averaging his hardly needed, but I usually averaged 3 runs (a setting in the Impulse Response dialog) to be sure the measurements were not contaminated by ambient noise.  Two runs in a row typically show the same value at 100.3 Hz with 0.1dB precision.

*****

I had never been able to do the leading edge alignment before with any feeling of confidence.  No other tool has given me as clear a display of the signal vs time as ARTA.  You may be able to find pictures of my previous attempts farther back in this blog.  In every previous case, a lot of guesswork and interpretation was involved.  Not so with the Signal Time Record feature.

Although the ultimate alignment is done with the frequency output at the crossover frequency, the leading edge alignment is still important, because it is a good place to start.  When starting the alignment on a system like mine, one doesn't really know the latency of processors and DACs in line.  It all has to be included in the "starting point" which can then be further optimized at the crossover frequency.

Sometimes I run the signal time record feature several times if the transient looks affected by nearby noise.  I didn't at first realize how to zoom and scroll the ARTA window, so my earliest pictures are not as revealing.

Here is the final determination of the leading edge of the Acoustat panel output.  I put the marker where the signal begins abruptly downwards first.  (I do not know why it goes downwards first, by all other measures all my speakers are unambiguously "in polarity."  I've spent much time verifying that by various technical means, including and hand-made asymmetric signal and a smart phone app.)

Leading edge of Acoustat signal

Here is the final determination of the leading edge of the supertweeter output (with the miniDSP OpenDRC-DI running at 24/96 using minisharc 4x10 plugin).  I line up the marker with the leading large edge, ignoring smaller pre-ringing.

Leading edge of Supertweeter signal

Here is determination of the leading edge of the subwoofer signal, after I finally figured out to expand the horizontal and vertical scales first (previously, it was looking for the first pixel).  Prior to exampding the scale, it was virtually impossible to see where it started closer than 1ms or so, because the initial start is so low in level.  There is still some ambiguity because of noise, and sometimes I will try another run to see more clearly.

Leading edge of Subwoofer Response

The subwoofer start time does not actually optimize its phase at the crossover frequency.  That is best measured at the crossover frequency by putting one way (the midrange) out of polarity.  The supertweeters are deactivated.  At the crossover frequency, if the levels are set properly, there should be a null in the response.  I was pleasantly surprised that there was a deep null.  And then that null is made deeper by moving around in 0.02ms increments until the optimal delay value is determined.  First I tested positive and negative 0.02ms increments away from the leading edge alignment, to determine which direction to go in.  Then larger increments in delay time are tried, backing up when the notch starts getting less deep.  Ultimately, the optimal delay adjustment causes a very deep null, differing significantly from 0.02ms forward and backward delay adjustments around it.

In this picture the green line is the system response, and the lines below are % distortion (which is very high where the fundamental cancellation is occuring, an artifact of the measurement being done).  I set the marker as close as possible to the crossover frequency (it is set to 100.3 Hz).  Then I can read the amplitude very precisely (from one run to the next, it is often exactly the same to 0.1dB, except at the deep null where it goes lowest, it actually varies a lot, but always lower than the surrounding points).

You may note the notch in the response actually appears to be centered below 100Hz.  This is because of room acoustics and rear wall reflection.   I measure the level as close as possible to 100 Hz, ignoring any deeper notch below 100 Hz.

The high end response in the graph above rolls off because supertweeter is disconnected.


Here are the delay adjustments dialed in on June 19, before converting the supertweeter miniDSP to 96kHz operation.

Here are the ultimate delay adjustments dialed in, after the miniDSP for the supertweeter was switched to 96/24 operation on the afternoon of June 20.  Note that the super tweeters require 1.3 ms more delay than before because the 96/24 path has lost 1.3 ms latency for some reason, compared to when it was running at 48/24.  The change in latency might be from the miniDSP itself, or in the following Behringer EQ, or in the DAC, or all of these combined.

The final delay adjustments on June 20

I was unhappy to see the following left channel system response when all was done.  It was not very flat in the bass, with 100 Hz itself still in a notch (after correcting the Acoustat polarity), with a huge rise over 15dB from there throughout the deep bass below 100 Hz.  This was not my intention (I intend to have "flat bass", or what I call "electrostatic bass" even though realized with a dynamic subwoofer).

After adjusting the magnitudes of my pre-existing parametric EQ's, but NOT their frequencies (which were adjusted by hand tuning an oscillator, and are therefore "real" resonances and not digital artifacts) I was able to make the bass much flatter.  I still couldn't totally flatten the deepest bass without creating undesired holes there.  So I left it like this:

This graph is also showing the effect of the super tweeter in the rise abov 15kHz.  The super tweeter is optimized through other means, mainly using the microphones of my smartphone (which use algorithms to simulate a perfect spherical omnidirectional response).  The rise in on-axis response shown here above 15kHz is not representative of the average room response, or even at ear positions a few inches to the sides, because of the high degree of beaming from the supertweeter at those frequencies.  The on-axis level must be exaggerated at the exact center (where the microphone is) to create the equivalent auditory sensation as a live performance.  This is the opposite of the situation which calls for the Gundry (aka Linkwitz) Dip, which is another useful alteration of flat on-axis response,  because small rooms reflect unnatural amounts of 2-6kHz directly into the ear from side reflections.  Also, I have not as yet incorporated my microphone calibration into ARTA, and my microphone has a slight HF resonance contributing to the rise shown here.

Whenever I've lowered the supertweeter level to make it appear flatter with some kinds of measurements (and not so much my smartphone held at the listing position, which shows almost flat response at 20kHz) it sounds wrong to me, dull and more irritating actually.  The supertweeter takes away any sense of "strain," making things more rather than less listenable, stridency is reduced when the super tweeter is active.  But when the super tweeter is too low, the strain comes back, the magic is gone, and I'm just listening to 30 year old electrostats not a live band.

Here are the new PEQ adjustments for the left channel bass:

Notably I reduced the notch filter at 106Hz down to 0.5dB, to help neutralize the dip around 100 Hz.  I didn't zero out the notch at 106 Hz so it could be increased again if it later seems this decision was not altogether for the best.  I also increased the depth of the notches at 45.3 and 71 Hz and the broad cut around 28.3 Hz to help flatten the deepest bass.


The right channel looked better, especially in the bass, so I didn't make any EQ changes there:

MiniDSP now in 24/96

I am now running the MiniDSP product OpenDRC-DI (digital I/O only) for my super tweeters at 96kHz.  A helpful poster at the miniDSP forums gave me the correct information.  I needed to obtain the miniShark 4x10 plugin.  That was not the plugin linked at the OpenDRC-DI product page.

Once I got the correct plug-in, it should have been simple, but I messed things up and I spent 4 hours sorting it all out.  Now it is running a LR4 crossover at 17220 Hz just as with the previous plugin, but now at 24/96 instead of 24/48.  This is more suited to a super tweeter setup which (supposedly) responds up to 40kHz.

Once I finally got things sorted out, i re-measured the time alignment and readjusted the time alignment of the leading large edge of the supertweeter response (assuming the tiny edges to be digital pre-ringing).   This seems to require 1.30 ms more delay than with the 24/48 plugin, possibly partly because of reduced latency downstream.

There is little use in doing a "cancellation" type of delay adjustment with the super tweeter, and the crossover isn't really that well worked out.  The acoustic crossover of the Acoustats is a combination of their own falling response (down at 18kHz) and the LR4 lowpass I have added at 20kHz (to avoid wasting amplifier power into the capacitive load of the Acoustats).  This approximates, but not perfectly, a lowpass around 17 kHz, but it needs to be fine tuned better.

Anyway, the leading edge of the output of both Acoustats and super tweeters are fairly easily measured, and I believe it is quite close to exact time alignment required for these drivers (the subwoofer needed 1ms less delay than the leading edge alignment, but subs are different).

  It's possible the best type of delay optimization for the super tweeters would come from looking at the impulse response.  But that should be done in combination with analyzing better crossover curves.

New Time Alignment. This Time for Sure

In my own thinking, the time of first air movement was more important than alignment at the crossover frequency, and I "figured" if there were any correction, it would be additional woofer delay.  I tried testing the other direction only briefly with a big difference.

Anyway, Thursday night's alignment ended up in total chaos, things were not looking correct at all (though it sounded ok).

So I decided to start all over on the subwoofer/panel alignment.  Starting from the time of first air movement.  THEN, from that point, I "fairly" looked fore and aft, starting with the smallest increments in delay adjustments so as not to miss something close (as I had apparently done on Thursday night).  The smallest increment is 0.02ms.

On the right speaker, I optimized the total output at the crossover frequency, 100 Hz.  At first I worried how I would do this, knowing acoustical measurements to be hard to make sufficiently precise.  But I simply decided to use my new program in the ordinary frequency response calculation, and see what it calculates for 100 Hz.  It is precise to 0.1% (in linear voltage) and apparently quite repeatable.

Technically I run an impulse, using the sweep signal, and then convert it to frequency response (with distortion harmonic estimations also, an added bonus).  To help improve the repeatability and accuracy, I set it to run 3 sweeps and average.

It was quite clear from the beginning that I had been optimizing the bass in the wrong direction.  Instead of adding MORE delay to the bass, to move the alignment further out past the "phase lead" or whatever it is, it turned out I needed to subtract even more delay from the bass.

After seeing which direction to go, I scaled up the changes and then scaled them back down as nearing what appeared to be the optimum, peak response at 100 Hz.

Satisfied with right channel, I moved on to left channel.  But this time, after a few measurements, I decided to try something different.  I inverted the panels on the panels (sadly I have no easy way to do this for the bass) by swapping speaker connections, and then maximized the null.  I was surprised and very pleased that in fact there is a very deep null at the crossover frequency when I do this.  And I made it as deep as it can be, within the closest 0.02ms of delay.

The nulling approach is really the better one, I think, and if I remember correctly it's also what Linkwitz recommends.

The result of doing one channel by peaking and the other channel by nulling nevertheless makes sense.  The difference between the two seems about what I would expect based on the fact that one subwoofer is about 5.5 inches further back (a situation not easy to change, it's very cramped with equipment in the front of the room).  Correspondingly, the delay for the closer subwoofer is 0.3ms more.

(On the previous day's adjustment, the delay difference was opposite from expectations, leading me to believe something was wrong.)

But then, seeing generally how poor the curve below 200 Hz was in the left channel, I decided to optimize the parametric EQ settings (PEQs).  I didn't add or subtract any PEQ's, just change their magnitudes, in some cases by fairly large amounts.  There is little doubt that ARTA is showing a more accurate frequency response curve than my phone's RTA app, which is generally what I have used before  (though, combined with very slow hand sweeping, which I still believe is the best way to find the critical points where adjustments should be centered).

I wasn't sure this was going to work out well, because it was already late and I wasn't going to have much time, but it did work out pretty well.  There had been about a 20dB rise below 100 Hz, 100Hz being at a low point (even after time delay adjustment and putting the polarity back to normal).  I lowered the excess deep bass, and raised the area around and just above 100 Hz, so it's much closer to flat overall.

I've also been working on getting the miniDSP used for the supertweeter changed from 48kHz sampling rate (response to about 24kHz at best) to 96kHz sampling rate.  These units convert all inputs to their internal sampling rate--whatever it is.  Using the standard "plugin" supplied by miniDSP, you get 48khz.  Actually the website says it differently, it says a different plug-in in standard and will get you 96kHz, not even mentioning the old standard 48kHz one I got last year.

I ordered a new miniDSP a few weeks ago for experimentation, and I thought therefore I would get the new 96kHz plug in.  But the website info was grong, I still got the old 48kHz plugin.  I tried buying the one they indicated for 96kHz.  It only costs $10 so I didn't want to wait to try to send get them to me for free.

Well, sadly, that didn't work at all, I just got error messages.  I also tried another plug in that seemed close to what had been recommended in forums.  That didn't work either.

So finally yesterday I posted my problem as a question to the miniDSP user forum.  Within a few hours, a veteran user suggested what I needed.  So I've obtained that plug in, and hope to get the 96kHz working today.

That will require me to readjust the delay because likely the new plugin will have different latency.  But now I have good ways of setting the time alignment.

Problem getting miniDSP to do 96kHz sampling rate

I have just now posted this at the miniDSP help forum.  [Problem was resolved later by a helpful poster who told me I needed the minisharc 4x8 plugin and he provided a link.]

I have four OpenDRC-DI units, including one new one purchased this month. I would like to use one of them at 96khz, which seems to be an advertised feature now, but the (formerly?) standard OpenDRC-2x2 plug in does not support it. The product page for the OpenDRC-DI now says that when you buy an OpenDRC-DI, you should get the miniDSP-4x10 plugin. However, when I purchased a new OpenDRC-DI this month, the only thing that actually appeared in my personal download folders (and I looked at each one, couldn't this be simpler???) was simply the same old OpenDRC-2x2, which I already had from previous purchases. So I clicked on the link to the miniDSP-4x10 plugin on the OpenDRC-DI product page and purchased the miniDSP-4x10 plugin for $10. I then go to the my personal download folders and it appears that I can download the 4x10 plug in. BUT when I click on the image which says "miniDSP 4x10", I don't get anything that says 4x10, what is actually downloaded to my computer (through Chrome browser) the miniDSP 2x8 plugin. I can't figure out how I'm supposed to obtain the miniDSP4x10 plugin ! When I run the miniDSP 2x8 that I downloaded as above (trying to get the 4x10), it fails to communicate with any of my OpenDRC-DI units, including the brand new one. I get this error message: Incorrect Device connected to this plugin software! Please use the correct plugin software HWID supported by the software: 1 HWID of the device:4 I also tried purchasing and downloading the nanoshark 2x8 plug in. That simply fails to find the device and doesn't give error message. I have no trouble using the OpenDRC 2x2 plugin. But it gives me 48kHz sampling rate, and I want 96kHz sampling rate for my super tweeters. I am using MacOS computers though I do also have a PC if needed.

One DEQ Weirdness figured out for good, I hope

I've figured out (again?) why the Behringer DEQ delay settings, on the I/O page 4, often have Left and Right lumped together and you can't switch channels by simply pressing the "Left/Right" button.  This is maddening because for the subwoofer, I need to put in different delay values for each channel.

If the button has both Left And Right highlighted, it needs to be held down for a few seconds then either Left or Right is selected, and the other becomes selectable with a tap.

To go back to delay settings in both channels, you hold down the button again (technically it is "Button A") for a few seconds, and then it reverts to Left And Right again.

It has nothing to do with the "Stereo-Link" (vs "Dual Mono") setting in the Utility manual, something that the DEQ manuals back to 2007 have emphasized.

Funny I thought I did this last night, after desperately trying other things, and then trying to remember what I had done before, but it didn't seem to help.

Then I tried it this morning, and voila it worked.  I got the idea after re-reading these ambiguous lines in the manual:

With the A key you can select the left or right stereo side. Independently of the Stereo LINK mode both sides can be processed separately. Keep the key pressed to edit both sides simultaneously.

"Keep the key pressed" sounds like you keep the key pressed while turning the knob.  But actually just keeping the key pressed for a few seconds changes from single channel to dual channel delay setting mode or back.

This is sadly typical of how most manuals are written.  And also how I forget the magic tricks I may have used before.