There's a depression right around the crossover frequency (125 Hz) in both channels and in both Movie and Hotseat modes. This depression seemed to respond to an adjustment (done in Movie mode) that was counterintuitive. I delayed the subs LESS so that they sub gets the signal only 0.6 ms earlier (it was 1.6 ms earlier before). That raised the output at the crossover frequency.
But this made no sense at all, since the sub is about two feet back from the panel. Making the delay compensation much LESS than 2 ms should make it worse, not better, I figured. But if I brought it down to 2ms, it got even worse than 1.6ms.
Because of this anomaly, I decided to reopen this time alignment issue which I haven't investigated for about 6 years.
Back in 2020 I made 3 or more posts on the topic. I tried various things, and concluded that you couldn't rely on measurements at the crossover frequency. There were too many confounding factors, I decided, and measurements never made any sense. I ended up concluding that simply using the distance between sub and panel was as good as anything you could do. And that was the inspiration for the 1.6 ms delay, since the right sub (then) was about 1.6 feet back (back then).
Well that was before a lot of changes. And I mean a lot. I changed the speakers, I added the miniDSP FIR based linear crossovers, I re-did the EQ's many times (including just last week) and I moved the speakers farther out, about 2 feet from the subs now. By the old reasoning, I should have increased the delay compensation to 2 ms, not reduced it to 0.6ms.
I decided this week the time alignment needed a whole new investigation. And there's one thing I never thought of before. The subs, which are SVS PB13 Ultras, (now, at least*) have internal DSP based processing. All such processing introduces a time delay. I asked Google and it replied I could expect a 5-6 ms latency for the DSP in an SVS PB sub. That was based on an investigation by ASR (Audio Science Review) which I have a lot of respect for.
https://www.audiosciencereview.com/forum/index.php?threads/svs-subwoofers.17378/
(*When I first bought the SVS subs in 2009 they had all analog controls, and might not have used DSP, but I've replaced the plate amplifier boards since then, almost always under their warranty. As of now, the right sub has the newest version I have, which is fully app controllable. The left sub has digital controls, but I have to go to the back of the speaker to control it. Since they are different plate amplifier boards, they might also have different latencies, though I'd expect them to be similar because SVS probably tries to keep the latency as low as possible since many people don't have a way of correcting it like I do.)
Since I never took the latency of the sub DSP into account before, it's no wonder the whole thing seemed confusing. The expected delay compensation should be about 5-6 ms (for the DSP) plus 2 ms (for the 2 foot difference) or around 7-8ms total (which is close to my experimental result below).
Furthermore, given the crossover frequency of 125 Hz, the response at 125 Hz should cycle through a range of values that repeats every 8 ms of applied delay compensation. So if the correct delay compensation is 8 ms, it should also give a local peak response at 0 ms, which is close to the 0.6 ms I determined from a few quick and not very careful measurements I did last week. (There is quite a bit of randomness in these measurements, and I didn't take them all the way to 0 ms because I knew that would be wrong.)
This does not mean that 0, 8, or 16 ms of delay would be equivalent in every way, only one of them would be the correct delay compensation value, but they would yield identical (or at least similar) response with continuous signals.
(I use pink noise, which is not really a continuous signal, but in some ways it acts like one. Actually, given the fact that it is random, the very highest amplitude peak should be at the true correct compensation value, and the other local peaks at 8ms intervals should be increasingly lower, but they would be local peaks higher than at delay values in between.)
This seems to be confirmed by a series of measurements I did yesterday. I played pink noise and measured the 1/6 octave band at 125 Hz while I changed the delay added by the sub EQ.
(Note that I have the the panel delay fixed at 10ms, so if the sub delay is set to 2ms, the signal to the sub would therefore "lead" by 8ms because it has 8 ms less delay being applied. By using a 10 ms fixed delay for the panels, which are equidistant from the listening position, I can compensate for the subs independently, since they are at slightly different distances, and I can compensate up to a 10 ms difference.)
I was starting at "9.42 ms" for the the right sub which I had determined in more crude measurements last week. (That would correspond to 0.58ms relative delay.). That made me guess that the correct setting would be 1.42ms (7.58 ms relative delay which is close to the predicted 8 ms).
So assuming my optimization last week was worth something, I started at 1.42 ms sub delay, tried 0.42, and then 2.52. It was very confusing, and I ended up going all the way from 0.42 to 9.42 to see the pattern. But the repeating pattern is unmistakeable.
Sadly my Analyzer app doesn't show the actual measured level after you freeze the analyzer. (I let it run for about 30 seconds on random pink noise.). Instead, it shows the suggested EQ level. As the response goes down, the EQ goes up. So what we want is the lowest suggested EQ level, and since the bass level is generally raised for my room curve, this means we are looking for the largest NEGATIVE number (which corresponds to the highest actual output at the crossover frequency):
Sub Delay Suggested EQ for 125 Hz (smaller means better--higher output)
9.42 ms -1.0 dB
8.42 +0.6
7.42 +2.0
6.42 +3.5 ***worst***
5.42 +0.8
4.42 -0.6
3.42 -3.0
2.42 -3.9 ***best***
1.42 -3.4
0.42 +0
This shows exactly the pattern I was expecting. The amplitude goes through a sinusoidal cycle in about 8 ms (determined by the crossover frequency of 125 Hz, because 1000ms/125 is 8ms). It reaches its best value (among the delays tested) at 2.42, which corresponds to 7.58 ms relative panel to sub delay compensation, right in the expected 7-8 ms range. I would expect it to hit another peak at 10.42 (but in that case, the signal would be going to the panels first, which is clearly wrong). The worst value is at 6.42, which is right in between 2.42 and 10.42. At 6.42 ms delay the amplitude is so low it needs +3.5dB of EQ to reach the spectrum average, while most of the bass is above the spectrum average. The output at the best delay tested, 2.42ms, is 7.4dB higher, as indicated by the suggested EQ of -3.9dB.
When I was making these measurements, I was getting very confused, not seeing the pattern correctly, so I mistakenly decided to a finer grid between 2.42 and 3.42. I did not get a better result than at 2.42ms. But it looks here like there could be an even higher value between 2.42 and 1.42, so I need to continue this investigation there to find the true optimal delay compensation value, or at least closer than just the nearest 1 ms. My determination of a local peak around 9.42ms was probably mistaken, it would be more like 10.42ms.
(When I was making the measurements I often confused positive and negative EQ values, and I also figured that what might be important is to compare the EQ suggestion for 125 Hz with that at 112 Hz--that made things very confusing. When I laid all the data out in a table it became clear that the expected pattern is shown most clearly in the actual 125 Hz value itself, which is where it should be clearest, because that's the crossover frequency. It's still confounded by the fact that changing the delay does change other frequency bins too, which then affect the baseline the suggested EQ at 125 is computed against. It would be nice if I could read the measured amplitude value rather than the suggested EQ value after the display is frozen, but I have seen no way to do that.)
While 2.42 ms is the best delay tested so far for the right channel, a quicker series of measurements suggested that 0.5 ms is the correct sub delay for the left channel:
Sub Delay Suggested EQ
8.72 ms -5.4 db
2 -2.5
1 -4.4
0.72 -4.7
0.5 -5
0.26 -4.7
0 -4.4
Funny that my original 8.72ms setting is still better than 0.5, there may be a still better point between 0.26 and 0.72 but 0.5 is close, +/- 0.25ms.
The left sub is actually closer to its panel than the right sub. Why it should need about 2 ms more relative delay difference (from the 10ms used on the panels) for time alignment is an open question. I have three ideas:
1) The plate amplifier board in my left sub is a slightly older generation. (The Left sub is a Sledge STA-1000D and the right sub is a Sledge STA-1200D.). The STA-1000D is controlled by a digital menu on the plate itself, whereas the later STA-1200D can only be controlled by an app. However, online sources say they both use DSP and that all SVS DSP plate amplifiers have a 5-6 ms delay. Nevertheless, it seems plausible that the later generation DSP has lower latency, since latency is something SVS would try to minimize. If this is the correct answer, and my measurements are correct, the STA-1000D has about a 5ms delay, and the STA-1200D has about a 5ms delay, and I should go with my measurements in the expected range 0-3 ms (10-7 ms delay relative to panels).
2) Online sources could be wrong and perhaps the earlier STA-1000D may not use DSP under the settings I am using (I am only adjusting volume and setting the port mode). In this case, I would be best using the 8.72ms setting for the left sub which only provides 1.28 ms relative delay, the delay being caused by having the sub about 1.5 feet back from the panel, with no digital latency.
3). I cannot measure the delay correctly with an amplitude measurement because the Left sub is in a corner while the Right sub is not. Possibly the corner has a horn-like amplifying effect on an acoustically delayed response which overwhelms the direct response. (This was what I concluded in 2020, not thinking about the possibility of DSP at all.). It is not obvious what I should do in this case. Possibly adjusting for the maximum amplitude in the left as well as the right would be the best thing to do, in the 0-3 ms range.
I have emailed SVS customer support for info on the digital latency of the two different plate amplifiers. If they cannot tell me I may try using one of the many SVS chat groups.
A test I could, in theory, do myself would be to change the crossover frequency to 60 Hz, and then compare the 8.72 ms and 0.5 ms settings. If the plate amplifier has digital delay, this would ruin the response at 8.72 ms but leave it unchanged at 0.5 ms. Because of the way I have the crossover patched in through FIR filters in miniDSP boxes, this would be a little complicated. I would have to bypass or neutralize the miniDSP boxes and then add in 4th order LR filters in the Behringer DEQ's. It would be complicated but do-able.
Direct measurements using digital recording would likely be more complicated. I could create a series of 125ms pulses spaced at least 1 sec apart. Then make a digital recording with high end microphone and see if I can figure out the time alignment. It's difficult because low-passed subs only rise slowly from the beginning of signal onset, where as high-passed panels rise immediately then fade away. Where the low passed sub signal begins is a judgement call, especially given background noise and non-anechoic conditions. I've tried to do things like this before with very ambiguous results.
I've also previously tried to do time alignment using programs based on Maximum Length Sequences (MLS) like ARTA or the Tact 2.0. That is complicated by the fact these programs were not designed to assess different speakers with highpass and lowpass crossovers. My original MLS program, LAUD 3.0, is now inoperable because it required 1998 vintage hardware like the Turtle Beach Fiji or Monterrey cards which have become inoperable due to capacitor aging, and even when they worked with a vintage PC, it took days and days to get them configured correctly on an ISA bus based system. Even one of my tireless vintage PC friends (RIP) got tired of my endless attempts to get this working again circa 2016. I once figured it took about 50 re-installs of Windows 95 to get one of these boards correctly configured, because the PC would "remember" the erroneous bus settings and you'd have to start all over from scratch to try all the possibilities and zero in on the one that works. But I miss LAUD dearly because it gave much better looking impulse responses than ARTA. The MLS used by Tact 2.0 are basically simple pulses, LAUD and ARTA use quick sweeps which have more information, and therefore less noise, but ARTA fails to map this to a virtual square pulse the comprehensible way that LAUD did. I've never been able to afford the true state of the art MLSSA system. DIRAC, I don't even want to think about DIRAC. I hate programs that try to do everything for you, including frequency response and phase correction, and just give you thousands of FIR coefficients instead of comprehensible information (typically loading them directly in your processor). REW doesn't do time alignment type things any differently than an RTA like my iPhone analyzer app; it is meant for setting Programmable EQ's, not designing or time aligning multi-way speakers (at least this was true the last time I tried it in 2020).












