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Principles: Time of Arrival Delays, vs Phase tuning ?

Relevant

 
I interpreted it as overall system latency?

But also, some types of delay affect the music signal as a whole, all frequencies equally?

Then it may be better to use the term "latency", which is specific to what you describe.
 
If I add delay to all speakers it does not affect overall sound so absolute. I can add 50ms too all speakers or 5ms delay sub relative to main or woofer relative to tweeter. 50ms delay despite being a large number no audible change to sound where as 5 ms relative delay makes a difference. But I don't think that's the issue anymore John clearly understands concept of delay. I think more important whether it's integrating speaker to a sub or woofer to a tweeter the same physics/principles likely apply
 
So, latency is an unintended side effect, but delays are intentional?
 
Relevant

I think of it as four stages: the source creates the sound, the sound travels through the air, the room changes it, and the listener hears the final mix of direct sound plus room sound. WiiM’s sub delay recommendation seems mostly focused on arrival time, or which sound gets there earlier or later. With REW, it’s more focused on what the listener actually hears once the speakers, sub, and room are all combined.
 
I think more important whether it's integrating speaker to a sub or woofer to a tweeter the same physics/principles likely apply
As far as I am concerned I am leaving all the upper frequency crossovers - internal to the passive multi-way speakers - alone, also no HPF on any.

...

I'll continue to use this thread as a "my use case" scratchpad, maybe even graphics one day.

Estimates of my self-managed crossover points to / between the four types of DIY LF single-driver boxen

(Within the F-L/R colocated stacks)

between the LS50s and the top MBM couplers, ~160-200Hz

between the MBM couplers and Sub2 type, ~80Hz

Between Sub2 and Sub1, ~50Hz

Rumble filter HP ~18Hz

...

Center stack likely the same, but hopefully no MBM needed

If subs used on rear surround channels, likely just one "Sub3" type, wider bandwidth lower SPL

crossed at ~80Hz

...

For the mono-summed freely placed trueSubs, will need testing to see if Sub 3 is sufficient, or a mix

...

If not sealed, I am considering bandpass enclosures, possibly a PR design like @Wolfs' "Overdrive 10" project, aka Kilauea

I figure standardising on the "don't span" boundary ~80Hz will give more flexibility in testing summed mono or matrix'd bass for the stereo channels

leaving open the possibility of a simpler "all boxen under that point go mono" implementation.

...

Anticipating a "why four types of LF boxen?" Q

I am limiting the form factor to around a 12x12" face, 1-1.5 cu ft max internally - can be over 15" deep if needed, more internal volume means deeper bass, right?
 
As far as I am concerned I am leaving all the upper frequency crossovers - internal to the passive multi-way speakers - alone, also no HPF on any.

...

I'll continue to use this thread as a "my use case" scratchpad, maybe even graphics one day.

Estimates of my self-managed crossover points to / between the four types of DIY LF single-driver boxen

(Within the F-L/R colocated stacks)

between the LS50s and the top MBM couplers, ~160-200Hz

between the MBM couplers and Sub2 type, ~80Hz

Between Sub2 and Sub1, ~50Hz

Rumble filter HP ~18Hz

...

Center stack likely the same, but hopefully no MBM needed

If subs used on rear surround channels, likely just one "Sub3" type, wider bandwidth lower SPL

crossed at ~80Hz

...

For the mono-summed freely placed trueSubs, will need testing to see if Sub 3 is sufficient, or a mix

...

If not sealed, I am considering bandpass enclosures, possibly a PR design like @Wolfs' "Overdrive 10" project, aka Kilauea

I figure standardising on the "don't span" boundary ~80Hz will give more flexibility in testing summed mono or matrix'd bass for the stereo channels

leaving open the possibility of a simpler "all boxen under that point go mono" implementation.

...

Anticipating a "why four types of LF boxen?" Q

I am limiting the form factor to around a 12x12" face, 1-1.5 cu ft max internally - can be over 15" deep if needed, more internal volume means deeper bass, right?
I’ve gone down a somewhat similar path before with multiple LF sources: large subs, smaller subs, an MBM, and full-range mains. The biggest lesson I learned is that more boxes can give you more options, but each extra handoff also adds another place where phase, delay, polarity, placement, and room modes can either help or fight each other.

Build it in layers rather than all at once: get the mains and first bass section working, then add the next section, then the true subs. At each handoff, measure sources separately and together, check polarity and delay, and judge success by how each layer sums with the room at the listening position.
 
Another important thing to consider how you'll evaluate seat-to-seat consistency and overall system predictability as the number of bass modules grows. I previously experimented with multiple subs covering different bass regions, and while the end result could be impressive, every additional source added another layer of phase, delay, placement, crossover, and room-interaction variables. At some point the challenge stopped being extension or output and became understanding why a given peak, null, or integration issue existed. With a 20–50 Hz section, a 50–80 Hz section, dedicated MBMs, and the mains all interacting, troubleshooting may become increasingly difficult because a measured behavior could be the result of several subsystems interacting rather than one obvious cause. The concept is interesting, but I'd expect predictability and repeatability to become difficult to manage
 
Yes I can see that. Will start by laying the F-L/R foundation.

There will need to be long pauses between stages just from how slowly the "hobby envelope" fills.

Which time will be well spent climbing that learning curve :-)
 
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The only problem DC fixes for me is there is no AC power when I am off grid, by definition.

My preference is to choose components designed to plug into DC directly, especially thirsty ones, rather than having to run everything through inverters.

I have no need to discuss all these off-topic issues further here, other than answering specific questions y'all may think are relevant to the OP.



...

I will when ready be trying to see if this is a possible way forward
What amplifiers or any audio components give you external access directly to the DC power supply rails? That why I said you are not running on DC power, but you are always running on AC power in practice. This assumes that you are not getting inside each piece of gear to splice in to DC power supplies, which or course is possible! For someone that thinks that DAC oscillators drift enough to mess up the sound, it would not surprise me if you did.

Everyone to their own!
 
What amplifiers or any audio components give you external access directly to the DC power supply rails? That why I said you are not running on DC power, but you are always running on AC power in practice. This assumes that you are not getting inside each piece of gear to splice in to DC power supplies, which or course is possible! For someone that thinks that DAC oscillators drift enough to mess up the sound, it would not surprise me if you did.

Everyone to their own!
In defense of what my be seen as a harsh comment (non intended) you are in a Science Forum, so if you share your ideas you are in effect soliciting science-based feedback. Just sayin.
 
What amplifiers or any audio components give you external access directly to the DC power supply rails?
Exactly why I say I prefer not to do that.

There are literally thousands that simply accept DC power input, designed to run off an external PSU. Mostly 12V with internal boost conversion, but many at 24/36/48Vdc

3e A7 series is OK with even higher, but even 51Vdc is pushing up to the TPA3255 limit, not a great idea, when every last dB is needed best to buy bigger in the first place.


> For someone that thinks that DAC oscillators drift enough to mess up the sound

Simple conversion is required, but not sufficient for the goal here, did you read the use case details here?

Are you asserting that with maybe a dozen independent DSP units, each running crossover and delay filters on their downstream channels, of varying types and tap counts, clock drift will not have audible consequences?

How about when MSO gets involved on the overlapping mono subs?

Or when they are also servicing DRC eventually, not just bass management?

If you know how to manage phase timing issues reliably with a bunch of rPi and/or Pico mcu please link to relevant resources.

Nearly every member I respect thinks it's a fool's errand to even try without using one central DSP, going through on big A/D converter, so all the myriad I/O analog ports are served by a single clock.
 
Here's a 4ch amp runs on 12V, includes some basic bass management, might help reduce my DSP port budget.



Yes designed for auto use but pending @amirm testing, may well turn out to be audibly transparent, and powerful enough for at least some of the boxen...
 
Another important thing to consider how you'll evaluate seat-to-seat consistency and overall system predictability as the number of bass modules grows. I previously experimented with multiple subs covering different bass regions, and while the end result could be impressive, every additional source added another layer of phase, delay, placement, crossover, and room-interaction variables. At some point the challenge stopped being extension or output and became understanding why a given peak, null, or integration issue existed. With a 20–50 Hz section, a 50–80 Hz section, dedicated MBMs, and the mains all interacting, troubleshooting may become increasingly difficult because a measured behavior could be the result of several subsystems interacting rather than one obvious cause. The concept is interesting, but I'd expect predictability and repeatability to become difficult to manage
I think the concept that 2 point sources i.e. full range or co-located subs and mains have a much larger "sweet spot" than multiple distributed subs is underappreciated. Point sources still only have one theoretical sweet spot but assuming they are creating a decent wave front the size of the "almost perfect" and "good enough" area is much bigger than with distributed subs which also only have one sweet spot but the integration falls apart very quickly when moving in any direction away from it.
 
How does 3-channel stereo fit in that continuum?

...

Also, one approach I will test is solid colocated bass on both/all the front channels

testing A. keeping LR(C) separate right down past 30Hz

vs B. breaking <80Hz units out to mono

on the one hand

And then separatelt testing an MSO managed mono collection of <80Hz units out freely placed, specifically trying to widen the sweet spot

then testing in conjunction with B. above together in the MSO group, vs A. keeping LRC out of the MSO group.
 
I think the concept that 2 point sources i.e. full range or co-located subs and mains have a much larger "sweet spot" than multiple distributed subs is underappreciated. Point sources still only have one theoretical sweet spot but assuming they are creating a decent wave front the size of the "almost perfect" and "good enough" area is much bigger than with distributed subs which also only have one sweet spot but the integration falls apart very quickly when moving in any direction away from it.
That was actually my experience when I experimented with multiple bass sources that weren't all identical and weren't all covering the same bandwidth. The result could be impressive at the main listening position, but different seats had different bass balances. One seat might get more deep bass, another more upper bass or impact, simply because different sources were dominating different regions of the room.

I will say it was a more fun system. My MBM was at chest height and I intentionally boosted that region. Great for chest slam. The tradeoff was that different seats could hear noticeably different bass balances depending on which sources were dominating that position.

My wife thought it was ridiculous because I had MBM stacked on top of a 15in sub so the experiment had to end
 
Relevant as a test protocol

 
Exactly why I say I prefer not to do that.

There are literally thousands that simply accept DC power input, designed to run off an external PSU. Mostly 12V with internal boost conversion, but many at 24/36/48Vdc

3e A7 series is OK with even higher, but even 51Vdc is pushing up to the TPA3255 limit, not a great idea, when every last dB is needed best to buy bigger in the first place.


> For someone that thinks that DAC oscillators drift enough to mess up the sound

Simple conversion is required, but not sufficient for the goal here, did you read the use case details here?

Are you asserting that with maybe a dozen independent DSP units, each running crossover and delay filters on their downstream channels, of varying types and tap counts, clock drift will not have audible consequences?

How about when MSO gets involved on the overlapping mono subs?

Or when they are also servicing DRC eventually, not just bass management?

If you know how to manage phase timing issues reliably with a bunch of rPi and/or Pico mcu please link to relevant resources.

Nearly every member I respect thinks it's a fool's errand to even try without using one central DSP, going through on big A/D converter, so all the myriad I/O analog ports are served by a single clock.
Minuscule and non-audible. How would you compare analog crossovers with digital crossovers? Same problem, better or worse?
 
Dunno yet.

My plan is start learning REW, and then try quality passives on the top LS50s since they are HPF only.

Then active LR 24dB/octave but non-DSP to bandpass the MBMs, do the blending best as possible that way.

Then trueSub(s) below also active electronics, if the whole stack seems well blended, sum magnitudes reasonably

Then look at the phase side of things, also if any time of arrival delay issues.

If that stuff actually needs fixing that's where I start with DSP. "Linearizing" doing phase / time correcting the existing xovers IIR or FIR

vs replacing, re-doing with linear-phase from scratch...

Note that there's a hundred action items prior to each of these stages, even if I make fundamental changes to my conative personality / character flaws get executive functions back on track

six month timeframe is conservative to get any real results. Not to mention financial delays if it turns out big port-count single-clock converters are indeed requires.

Sorry to get personal in a public forum, which is the shame / cringe emoji?

I do appreciate the patience and willingness to help from y'all!
 
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