Blog Forums DIY Speakers and Subwoofers DINAS big brother(s)

last updated by AJC 11 months, 1 week ago
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    • #13188
      AJC
      Participant

      So, I’m at a crossroads. I will put the 5.1 system on the back burner. Instead, I’m trying to finalize my driver selection. Since DINAS are a powered speaker, I am planning on doing the same.

      I have two sets of drivers selected for the bookshelves. They will have DSP built in and will have where you can adjust the corrections through the WONDOM app. So, without further ado, the driver lists:

      Budget friendly DINAS “Bro”

      1) Savard RAP 6.5″ (sub)
      2) Dayton Audio RS150P-8A
      3) Peerless by Tymphany NE19VST-04

      DINAS roided out “Bro”

      1) Savard Hi-Q 6.5″ (or Sundown Audio SA6.5 which is currently about the same price as the Hi-Q, give or take $10)
      2) Tang Band W6-1721
      3) Peerless by Tymphany DA25TX00-08

      Now, both versions will be powered by the Wondom JAB5 4x100W amplifier with bluetooth and DSP built in.
      https://store.sure-electronics.com/product/756

      And the Power Supply is the LRS-350-36 (36V)
      https://power-nex.com/product/lrs-350/

      After they release the ISP5 (currently ISP1 and ISP3 are for sale, ISP5 expected end of the month), I plan on putting one per speaker. Why? Because in their infinite wisdom, even though they have bluetooth already on the amplifier, you still need this to program it with the Sigma program or to use their app. Why would you want to use their app when you can calibrate it before calling it good? So that you can do room correction in the future while using REW to measure the reflections, allowing for this to be done IN SPEAKER, meaning no matter which room you put this in, you can customize the EQ on the speaker to sound great in the room, instead of just making a flat response and hoping for the best.

      So instead of having the control knobs on the back with a standard plate amp, you will have more fine control through a phone app (although no automated process for an algorithm to do it for you; sorry).

      It also means you have to setup for two USB cables to be connected on the back. And I still plan on buying a cheap aluminum plate, an aluminum heatsink to screw onto the plate, then soldering small blocks to the aluminum plate over the components that need a heatsink (this is subject to change, but would be to replace the fans on the power supply and the amp; either that or I may do something else; still planning).

      The DSP in the amp will act as an active crossover for the three speakers and two of the 100W channels are able to be bridged, meaning you can feed 100W to the woofer and tweeter each, then 200W to the sub.

      As to price, you would be looking at about $600 for a pair of the budget friendly set (price still in flux during planning), while the pair for the Tang Band roided build would be closer to $850 or so for a pair (I’m building 2 pair, one for my parents and one for myself).

      I would love thoughts on this build.

      Specifically:
      1) Do you think (even with the DSP present) that the sound quality of the Tang Band speakers and the distortion levels are improved enough to justify the additional price?

      2) Is it worth paying the extra for the DA25TX00-08 because they are more sensitive (so you would use less wattage) which also would allow for more attenuation if needed in the high end with the DSP EQ?

      3) Would the extra $30 on the sub give you less distortion or smoother bass (I’m leaning toward the $100 subs on this one, but would like to hear from other’s that know more)?

      4) Is the APP worth the money to buy the programmer for each and every speaker? Or just buy one, calibrate the speaker with the calibrated microphone once (or every so often if you feel the response has changed), and call it good? You could just hook it up to a laptop in the same room, after all (requires mounting a plug to accept the plug between the boards on the backplate, instead of mounting just two USB ports to extend to the plugs for each board; unless using an easily removed backplate to access the connector on the amp), and correct in the final spot it will be in.

      5) is there anything I am not seeing in this build? Any cool ideas to build on from what I just laid out?

      6) Which of these two builds (or a Frankenstein of the suggested drivers, or another driver or tweeter you want me to consider) would you want to own more? Which of these two builds would you buy?

      Remember, these are basically studio monitors on how loud they will be and most of the frequency coverage.

    • #13189
      AJC
      Participant

      I was mistaken. Turns out this is the DSP chip and type to allow for an autoEQ in Sigma software using data from REW. So that means it should be able to help even if you do not care to tweak it by hand.
      https://ez.analog.com/dsp/sigmadsp/f/q-a/64304/adau1701-sigmastudio-rew-auto-eq
      https://ez.analog.com/dsp/sigmadsp/f/q-a/65120/mlssa-and-auto-eq-algorithm

    • #13195
      123toid
      Keymaster

      1) Do you think (even with the DSP present) that the sound quality of the Tang Band speakers and the distortion levels are improved enough to justify the additional price?  

      The tweeter is definitely a step above. I’ve used the other peerless tweeter and I am not a fan.  SO I would either upgrade the tweeter or go with a  different one.  I’m not sure I would choose the W6 as a mid-range by itself.  Although, it is a great speaker and it is an underhung motor, so it will be great.  If I had to choose between the two, I would choose the second set of drivers.

      2) Is it worth paying the extra for the DA25TX00-08 because they are more sensitive (so you would use less wattage) which also would allow for more attenuation if needed in the high end with the DSP EQ?

      THe one real benefit is while DSPing it you have a much less of a chance of accidentally blowing it.  I do love that tweeter though.

      3) Would the extra $30 on the sub give you less distortion or smoother bass (I’m leaning toward the $100 subs on this one, but would like to hear from other’s that know more)?

      I think your sub choice is fine.  Although, it is a little weird that your sub is the same size as your midrange.  That is atypical.  BUt it is your design, so if those are the drivers you like, then go for it.

      4) Is the APP worth the money to buy the programmer for each and every speaker? Or just buy one, calibrate the speaker with the calibrated microphone once (or every so often if you feel the response has changed), and call it good? You could just hook it up to a laptop in the same room, after all (requires mounting a plug to accept the plug between the boards on the backplate, instead of mounting just two USB ports to extend to the plugs for each board; unless using an easily removed backplate to access the connector on the amp), and correct in the final spot it will be in.

      I’m a little confused on this question.  The Wondom boards use Sigma studios, which is free.  You would then just need to buy one ICP-1 from Wondom or from Parts-Express.  Dayton Audio DSP amplifiers are the Sure Amplifiers rebranded.  I would still get a calibrated mic.  WIthout it, you are just guessing witht he DSP.  Let me know if I missed something though. 

      5) is there anything I am not seeing in this build? Any cool ideas to build on from what I just laid out?

      Looks like a good idea to me.  I am interested to see how it turns out.

      6) Which of these two builds (or a Frankenstein of the suggested drivers, or another driver or tweeter you want me to consider) would you want to own more? Which of these two builds would you buy?

      I would consider this peerless tweet over both of them: http://bit.ly/xt25bg60  It would save you some money and is an excellent performer.  It is the same one I chose when I built my studio monitors. 

      For the midrange, this 4″: https://bit.ly/2Qz1Xr9 I used with great success or you could even check out this 5 1/4″ HDS: https://bit.ly/3v4zpVg  Both of these should be easy to work with and should get you really good sound. I would keep the mid a little smaller than the subwoofer.


    • #13196
      Elliott Designs
      Participant

      Hey there guys,

      It’s been a while since I’ve been online but I’ve started freelancing whilst at uni to save up for speaker building. Anyway getting to a few tips I have for this project…

      I love the Wondom DSP amps. The problem with Jab5 (4x100W) is that it doesn’t support 4ohms, only a minimum of 6 if you look in the specifications. It’s really frustrating since I was going to use the board myself and only found out a couple of weeks ago I couldn’t. Instead, I’m opting for 2 Jab3/3+ (haven’t decided which yet). I found it should still be cheaper overall since 2 lots of 24v 5A power supplies are still cheaper than 1 36V 15A power supply needed for the full potential from the 4×100 amp. (well its cheaper over here in the UK anyway, not sure about the price differences for you guys).

      As for DSP, I’d highly recommend using REW as you mentioned, but it’s actually quite a capable board, so what I will be doing is using a free program called rePhase, import the DSP profile from REW and convert it to a linear phase FIR filter to use in the DSP (it’s a great method and I’m currently using my PC to do the signal processing, (equaliser APO)). Normal EQ has always sounded a bit off for me and I found out that it’s because the phase gets shifted over the frequency range depending on what frequencies have what gain etc. You don’t get that with FIR filters but you do get some delay (which I’m alright with). Also, with the DSP boards, I would recommend doing a time offset for the tweeter to get it properly time aligned with the midbass.

      Hope at least some of this helps!

      Elliott


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #13199
      AJC
      Participant

      @123toid – Well, you will be happy to know that before seeing this, I changed and decided on a different sub and mid woofer.

      1) SB Acoustics SB23MFCL45-8 (unless the Dayton RSS210HO-8 would have less distortion, but I think the SB would have the similar performance while less distortion),

      2) SEAS Prestige U18RNX driver.

      The SEAS driver costs the same as the Tang Band, but I think would work out better. That, combined with the SB Acoustics sub performance should be a better selection.

      I had already decided to go to the DA25TX00-08, but will look the one you recommended over.

      For the question on the programming board, when connected to a PC, I’d only have to have one board to program. But, WONDOM also has a phone app which allows for changing the other elements. After finding the autoEQ and reading ElliottDesigns post, I think I’ll just program each speaker to flat separately using that method.

      As to calibrated mics, I have a miniDSP UMIK-1 that has a separate calibration file from Cross-Spectrum Labs.

      @ElliottDesigns – Thank you for the response as well.

      I hear you on the JAB5 being limited to 6 Ohms. I currently was asking Sure Electronics if the two channel bridged to 200W, because that should cause it to be 12 Ohm potentially, could handle an 8 Ohm load.

      That is also why I chose the 8 Ohm versions of the SEAS Prestige U18RNX and the SB Acoustics sub that is 8 Ohm. Also, the DA25TX00-08 is 8 Ohms, so if the AMP bridged channel can handle up to 150W at 8 Ohm (the RMS of the SB sub), then the JAB5 becomes the perfect compliment.

      I will roll around what you mentioned on the JAB3s or what I might do is get the other component for running the channels and get the 2.1 channel amp that has 2×200 and 1x400W. But I’m about to purchase the speakers I mentioned anyways (unless someone can point out why the Dayton sub would be the better buy).

      Also, thank you on the linear phase FIR filter. Definitely worth it. I also loved using equalizer APO. GREAT CONTROL!

      Also definitely planned on the time delay if needed (I am thinking of going open baffle for the tweeter on top to better push it back to align it with the other drivers; but, even then, timing can be an issue and you have to watch out for too far back effecting sound below the top ridge for the cabinet, which means the time delay is a great compromise to accomplish anything I cannot do properly with driver alignment).

      Thank you both!

    • #13201
      Elliott Designs
      Participant

      @ajc9988

      Glad I could help. I’d probably avoid the open baffle tweeter, you are more likely to reach excursion issues and it can cause phase problems making your crossover design (active too) really difficult because you then need to measure the phase and offset it using FIRs. Probably worth just enclosing it since it’s unlikely to be worth the effort to do the phase correction for that.

      In terms of ohms, you will be perfectly fine if the impedances are 8ohms. I assumed you had 4 ohm drivers (as I have). An amplifier can drive anything above its minimum, it’s just less efficient.


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #13202
      AJC
      Participant

      @elliottdesigns – Good to know regarding the tweeter and open baffle.

      Also good to know on the amp. If I may ask, considering the impedance difference (and going 6 Ohm to 8 Ohm usually drops wattage supplied from 100W to 80W, depending on a couple factors and roughly converted on some designs out there), do you think 100W per channel feeding a 100W RMS tweeter and 80W driver would be enough? Or 200W not being too much for the bridged channel (with the channel still only having 150W RMS needed by the sub, but 200W bridged should be 12 Ohm (whole no free energy thing), so it may work the channel harder)?

      Sorry, I am still a noob (although SigmaStudio does look up my alley).

    • #13203
      Elliott Designs
      Participant

      @ajc9988

      Nah, it’s fine, we were all noobs at one point. If the tweeter already has an enclosure, open baffle wouldn’t make a difference and 100W RMS would be fine. If it doesn’t come with its own enclosure (unlikely but can happen) then you need to test it in the enclosure you are building it in winISD and see what wattage it can handle with your crossover filter simulated in the filter tab. You then increase the wattage whilst looking at driver displacement until it reaches xmax, that should be your maximum wattage value. (you can ignore all that if the tweeter is enclosed because the RMS value they provide you would already take the excursion into account).

      You do the same with the woofer and the subwoofer to see what the maximum wattage you should be feeding those is. With the high pass applied to the woofer you are simulating, and the low pass applied on the subwoofer. (This gives you accurate results but you need to make sure the crossovers are applied before you put any power into them otherwise you can go past excursion limits and damage the drivers). Steeper filters allow for less excursion but when using linear phase filters it will keep adding more and more delay for the filter to still be accurate. So try and find a happy medium. I would say setting the subwoofer to woofer crossover at 80hz is usually a safe bet. Depending on how capable the woofer driver is though, this may need to be set higher. Unfortunately when going higher than 80hz you might start ‘hearing’ the subwoofer which you might not like. Anything up to 120hz isn’t too noticeable for most but people more sensitive to directional cues would want to keep it around 80hz. Any lower than 80 there’s really no point unless you are setting your sound system up in a warehouse 😂🤣.

       

      Elliott


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #13204
      AJC
      Participant

      @elliottdesigns – Turns out I will get the SB sub. Same guy that designed the Scan-speak 22W/8851T, so it should be a very good solution. Finding out the same talent did the design makes a difference (skill is skill, so even if making compromises, likely would not compromise in ways that sacrifice performance significantly).

      And I will keep that in mind. The subs are in the same speaker cabinet (but own compartment) in this design. I’m shooting for a 40-20,000 design on axis (although the tweeter starts diverging off axis at around 6,000 or 7,000).

    • #13207
      AJC
      Participant

      Some others have complained about the SB subs. I think they are fine. But I wanted to know if people feel other subs should be considered (including 10 and 12″ subs)

      CSS SDX10 XBL2 (have to save up longer, don’t like the distortion shown in AudioXpress here: https://audioxpress.com/article/test-bench-the-sdx10-home-audio-woofer-from-css-audio; but still considered a decent sub)

      Peerless by Tymphany 830452 10″ XLS Subwoofer
      Peerless by Tymphany XXLS-P835016
      Dayton Audio RSS265HO-44
      Dayton Audio UM10-22 (would have to switch to the 2×200 and 1x400W amp)
      Dayton Audio MX12-22 (same)
      Dayton Audio RSS315HO-4

    • #13230
      123toid
      Keymaster

      @ajc9988

      The RSS subs are pretty good. The CSS SDX12 is unbelievable  Love it! Very low distortion and high linearity.  However, you are paying for the quality and will want to dsp a high pass (I have the passive radiator version).  I have a 15″ Max-x (mx15-22) but haven’t made a box yet.  For what it’s worth, Chris Perez, a engineer for Parts Express seems to really like them.  For the price, it might be your best bet.  I haven’t modeled the MX12-22 yet, so you may want to do that to make sure it’ll do waht you want it to in your sized box. 


    • #13232
      AJC
      Participant

      @123toid – I have modeled a couple so far. Kind of narrowed the field to the following:

      SB Acoustics SB23MFCL45-8 (8″,limited and cannot be put in too large of a box without exceeding X-Max; but if like the other SB subs, low distortion above 50Hz)

      Peerless XLS-P830452 – well regarded, good on distortion, 10″, able to be used in a small vented box (like 1.268ft^3 models to F3 25 and others have achieved F3 of 27 or 28 with this sub); drawback is it needs vented or dsp to unlock, which I plan on using both, so no real drawback

      MX12-22 – This little guy is a beast; models decently well, but needs a box the size of the RSS265HO-44 to really unlock it; it can match the RSS HO performance with fewer watts in an equivalent box, or it can keep up with the SB in a similar size smaller enclosure. Dayton has this going for it: it can do low distortion under 50Hz and can go deep. But, Dayton, in designing for that, has more distortion than others in the 50Hz and above range, which can muddy types of music and other sounds compared to some competitors. That doesn’t mean it is bad, just it has limited use cases. Also, above 100Hz, some models seem to go off the rails, so some daytons have to be crossed there, which is often fine as that is before directionality of the sound really picks up.

      Now, Peerless is the quietest of all three, but it also can be used in the smallest space while still having a flat response. The SB is good, but limits out due to box size. The MX12-22 seems really nice, as it is able to fit in a decently small space, but can hit deep.

      From left to right is the following (all vented (all 120Hz LP filter except U18)):

      Dayton RSS265HO-44 2.1ft^3 23.3Hz tuning freq. 1.5″x12″x38.87″ port 174.06Hz 1st port resonance HP filter 15Hz LR

      SB23MFCL45-8 1.2ft^3 25.8Hz tun. freq. 1.375″x8″x33.98″ vent 199.09Hz 1st port resonance HP filter 22Hz butterworth 2

      MX12-22 1.066ft^3 33.36Hz tun freq 1″x12″x24.29″ vent 278.52Hz 1st port res. HP filter 10Hz LR

      XLS-P830452 0.488ft^3 41.58 tun freq. 0.875″x10″x25.38″ 266.56Hz 1st port HP filter 15Hz LR

      U18RNX-P 0.644ft^3 48Hz Tun. Freq. 1.5″x7″x16.32″ 414.5Hz 1st port res. HP butterworth 2 – 32Hz

      The U18s can already play down pretty low. So even without a sub, these could do well.

      P830452 – 35.9L (1.268ft^3) 26.1Hz
      SB – cone excursion at 3.35ft^3 21.92Hz
      MX12-22 – 2.273ft^3 24.1Hz

      Didn’t change the RSS HO nor the U18. But that extends all of them to the 20s to low 30Hz.

      The CSS 10″ would add $240-280 over the price of these drivers, which is why it is not in the running. If I had the budget for that, I would try to likely stretch it to the SB 12″ sub, if being honest. All of the above are around $140. But unless I build a smaller box, the SB is excluded, and it feels like the peerless or the MX12 may be the better choices. Difficult decision. Really comes down to if I want a 2.5′ box or 3’+ tower.

      As to the drivers that came in:

      SEAS Titans: 2 are like a perfect matched set; 1 is around the same FS as the two in that set, but 1 is around 900Hz FS, over 60Hz higher than its brethren. All pass the rub and buzz.

      SEAS U18: for the two that came in, they passed the rub and buzz.

      Both have the bump in impedence around 1000Hz (this model, in the spec sheet, normally had the bump around 700Hz, which you can see a little bit around, but usually the bump the size of 1KHz was around 700Hz in the spec sheet).

      I ordered 4xJAB5 from Sure Electronics, mainly because I could not find ADAU1467 boards cheep enough (because they are still so new). ADAU1467 development boards are about $170 a piece. But, it can run higher quality filters without an issue.

      I ordered 4x500W@36V power supplies to run those boards (even though they are only 4x100W@6Ohm; that way I will never over run the power supply to make it heat or burn up).

      So literally the sub is the last decision to make. But since the JAB5s are on the slow boat from china….

      But more to come…

    • #13237
      AJC
      Participant

      @123Toid – Here is an article showing the distortion for the CSS sub you love: https://audioxpress.com/article/test-bench-the-sdx10-home-audio-woofer-from-css-audio (Dickason testing)

      It has really low distortion from 20Hz all the way to 100Hz (or 200Hz), giving it a clean sound.

      Granted, this is the 15″ SB Acoustic sub, but it does show where the distortion is and how quickly it dropped to lower than the 10″. Now, I suspect part of what dropped the distortion so quickly is volumetric in nature, so I would suspect, to a degree, that the smaller subs from SB would also have low distortion, but would not reach as low and might take a little longer for the distortion to drop. But, generally speaking (and since this is the company that mfrs the Satori line of speakers), I think their subs should have similarly low distortion, even in the smaller ones, although it must be noted the 8″ uses polypropylene, the 12″ is paper, and the 15″ is honeycomb paper fiber with fiberglass. 

      https://audioxpress.com/article/Test-Bench-Dayton-Audio-LS10-44-and-LS12-44-Shallow-Mount-Subwoofers
      Now, you cannot use the shallow sub to really explore Dayton’s distortion, but it gives a peak at what I was discussing in the distortion from 20-100Hz.

      https://www.diyaudio.com/forums/subwoofers/360855-tale-12-subwoofers-distortion-15-dollars-20.html
      That is where I wound up here, a great thread testing distortion (although he didn’t have the Dayton subs by the time he had his methodology down).

      That then led me to this website:
      https://www.dibirama.altervista.org/le-prove/sub-woofer.html

      There, you can look through the different size daytons and some of their testing, including by size for distortion. If you look at the 8″, the distortion doesn’t settle until 50Hz. If you look at the 12 and 15″, they both settle down around 30-35Hz, which means a little bit of certain types of music can be effected.

      Now, these are not the only factors, but this shows a bit on the distortion testing.

      Also, it is clear some subs are already on a downward slope by 100Hz to 200Hz, whereas some are flat out to like 400Hz. Those that have a high peak between their low and 100Hz, you can use a DSP to tamp that down to flat. But, looking for something more relatively flat throughout (or like the CSS, small bump, but then fairly flat throughout after that, similar to a pop from a tuning port design) can allow for other characteristics to shine in the sub.

      This isn’t to say any sub is bad, designed wrong, etc. This is to point out that some trade-offs can be seen in those charts, while those that know better how to read the T/S factors can divine even more.

      But, with that said, the 12″ Daytons do really have low distortion quickly.

      But that has been my rabbit hole on distortion and subs the past week or so. Fun stuff.

      If I misstated anything or got something wrong, let me know.

    • #13238
      AJC
      Participant

      2060-Titanset2-1.zip
      2061-Titanset2-2.zip
      2062-u18rnx-P-unit1.zip
      2063-u18rnx-P-unit2.zip
      2065-Titanset1-1.zip
      2066-Titanset1-2.zip

      Here is the impedance information from all of the current drivers I have. I have two more u18s ordered, but have to wait for a fresh shipment.

      The Titan set 1 are almost a perfect matched set. The second set has a more significant separation between Hz (898Hz or about and 918Hz or about).

      They all pass rub and buzz testing with it turned all the way up. I also used all the mem slots and ran the impedance over the +10dB to -40dB in increments of 2.5dB while having the settings to make it as difficult as possible.

      Wanted to share in case anyone wanted to take a look.

    • #13432
      AJC
      Participant

      Turns out this has just become a tower speaker.

      Also, finalized the sub today after my paycheck. I’m using the Scan-Speak 30W/4558T00.


      Easier than rewriting the T/S and other parameters. Even though the RMS is 150W, the X-Max is 25mm. The long-term max power is 350W. Meanwhile, this speaker’s response shows 89dB@2.83V, making it extraordinarily sensitive for a subwoofer. The modeling looked phenomenal.

      I currently modeled it’s box volume at 5.5ft^3 (155.7L), about 23-24L larger than the manufacturer says for vented, which they say has an F3 of 20Hz at 132L. And, of course, this is a scan-speak product, so quality is just about assured.

      https://www.scan-speak.dk/datasheet/pdf/30w-4558t00.pdf

      So, this means the SEAS U18RNX-P 6.5″ and the SEAS Titan 27TAC/Gb will be sitting atop a massive box for the 12″ Scan-speak. Maybe call it The SEAS Speak. Don’t know yet.

      But, considering it is a three way, you don’t have to have the subwoofer supporting many smaller speakers (another reason I went this route). That same rationale made me almost get the CSS 12″ subs, as they are great sounding and high detail until they go distorted. At these levels to match the other two drivers, it should never reach the distortion realm, which would have sounded very nice. But, the Scan-speak just edged it out.

    • #13482
      AJC
      Participant

      So, some fit and finish decisions were made and the following have arrived:

      4 packs of these for a total of 16. I am using one pair per driver, then one pair for the plate I’m going to make per loudspeaker as a whole. So if I ever make a passive crossover, I just unplug the amp, then run the wire from the passive to plug into where each driver is. Also looks nice when you open up the back plate (from what I am envisioning).

      4-packs of these for a total of 20, but only 16 will be used. Solid aluminum potentiometer knurled knobs. This goes with the 4 potentiometer pots of the amplifier (which can be reprogrammed, trying to decide if I will and if so to what, but the knobs will be there).

      20 pairs of these plugs. Need 16 to match the speakers. Why these? Well, I bought some 10 AWG solid core to run in these speakers. And since I have a spool of 500ft, wondering if I might do something crazy like soldering two lengths together for each stretch of positive and negative, making it the equivalent of 7AWG, then just have one go a little longer to stick into these 90 deg. banana plugs. Not all play nice with 10AWG, so I went with something I knew would.

      Just a 4-pack of these 15A 125VAC plugs. Has a fuse compartment for the first line of defense for over-current protection.

      We all like a tactile feel when flipping something on. What is better for some tower speakers with an active crossover in the amp than having that hard click when you turn it on (while also having standby possible by shorting a circuit and that button will be on the back also, but this cuts the power between the plug and the power supply, rather than relying on the amp’s circuits to go to standby, which is not off).

      I’m still finalizing rubber feet and wood inserts for the speakers, feet, and plate amp, so that they can be taken off without worrying and you won’t drill all the way into the compartment, leaving one less potential place for an air leak.

      I will also have to get some quick connects from the posts to the drivers, but one more thing to the list.

    • #13483
      AJC
      Participant

      I grabbed 4 of these 500W power supplies. I am designing the speakers for around 350W RMS tops. So, 80% of the power supply being 400W, these should do the job. These are 36V, 13A that were advertised for CCTV, Radio, and LED strips.

      These will be paired with the JAB5 fro Wondom.

      I’m mainly waiting on the subs ordered from a Finnish speaker company because Madisound was out of stock. The Finnish store was even cheaper than Madisound and have free international shipping. But, they only had two unspoken for in their late June shipment, so I do not know if I will get them shipped in June or July (I told them if they wanted, they could wait and ship all four together). But shipping and components, just like wood right now. Man…

      But I do have the U18s and the 27TAC/Gb already. So the project is coming along slowly.

      But feel free to drop some comments on the components, fit and finish, and anything really about the build.

    • #13486
      123toid
      Keymaster

      @ajc9988

      I love the power supplies.  Which brand did you end up getting? Also where do you plan to get those potentiometer knobs from.  Thsoe look really nice.


    • #13488
      AJC
      Participant

      @123toid
      https://www.amazon.com/dp/B08JHJXNPH?psc=1&ref=ppx_yo2_dt_b_product_details
      Those are the knobs, and they also have a light blue (baby or powder blue), a red that comes off with shades of magenta, gold, and silver. Szliyands is the brand, 5-piece.

      There are some cool options if looking through guitar knobs. Main thing is to make sure it matches what you are using, with 6mm being fairly standard.

      As for the power supply, it was Yi Mei Da. Also they are now $50 each, which they were closer to $40 each when I bought them.

      But it did take a lot of searching to find power supplies with 36V with the needed watts. The JAB5 recommends 36V, but the voltage range tops at 39V and minimum of 10V, so picking a 36V so as not to max the voltage seemed about right.

      A lot of power supplies seem to have gone up in price since I bought them, unfortunately.

    • #13514
      AJC
      Participant

      So, for those that are like me and think t-nuts are not as good as wood inserts, or just do not want to deal with t-nuts, here are the wood inserts I got for this project and why:

      E-Z LOK – 901024-20 Threaded Insert, Zinc, Hex-Flanged, #10-24 Internal Threads, 0.787″ Length (Pack of 50)

      https://www.amazon.com/gp/product/B002WC8TTS/ref=ppx_yo_dt_b_asin_title_o00_s00?ie=UTF8&psc=1

      E-Z Lok Threaded Insert, Zinc, Hex-Flush, #8-32 Internal Threads, 10mm Length (Pack of 100)

      MroMax Wood Furniture M6 x 25mm Threaded Insert Nuts Interface Hex Socket Drive Carbon Steel Bronze Tone 50pcs

      The first two are Zinc because they will be used to secure the drivers to the baffle. Specifically, the first one will work with the 5mm holes in most drivers, while the length will work easily with a double baffle (or if you are not doing it to keep the air seal and drilling straight through, just the benefit of NOT having to secure a T-nut that can fall off).

      The second one is for smaller tweeters and such with 4.5mm holes, which have less weight, so the smaller length likely will not matter for securing the driver.

      The last one is 1″ m6 threaded inserts. You can get Zinc instead of carbon steel, but it costs more. I plan on using these to secure the feet (matches common spike sizes, even though I’m using rubber, makes it easy to switch in the future) and for the plate amp. That way to save the others for around the speakers.

      I am likely also going to use waffle speaker grills. With a name like The Seas Speak, the waffle grills are to make people think of port holes and the grid metal grates on ships. Any effect on sound likely can be addressed with the DSP, and with wood inserts for the attachment of the waffle grills, there are no worries about removing them and putting them back on because nothing will wear out with the wood inserts.

      But I mainly find 6.5″ waffle grills and I need to check if that will work or not with the SEAS. I’ve been busy/lazy with research and haven’t just gone and measured.

      But that is a bit more about the aesthetics/fit and finish. And even though they are more expensive now (part on Trump and his trade war which Biden is continuing, part is shipping and lockdowns by Newsome in CA restricting offloading of products at the largest port in the US and other shipping costs), it really is the little things.

      But the first one is harder to find, the 20mm version of the #10-24, so wanted to share.

    • #13522
      AJC
      Participant

      Reminder for people purchasing female quick disconnect terminals for their speakers (which this will help generally for them coming off), ALWAYS use your calipers in order to verify the size of the male spade.

      For example, I’m using 10 AWG Solid CU line. This makes finding appropriate connectors hard.

      My speakers use 0.178″ width terminals with a 0.020″ depth. The power plugs I am using have 0.178″ width and 0.032″ depth. And finally the posts I am using through the wood use 0.205″ width and 0.020″ depth.

      Now, the standard size for 10 AWG is a 0.25″ connector. The gold plated ones from parts express do not give a depth. I have found 2 packages that will work for the 0.187″ terminals, but am still searching for the 0.205″ ones.

      Most people do not consider the depth of the male terminal when buying connectors. This could be why people complain about them slipping off or not being snug enough on the terminal. So always pull those calipers out and make sure you get the right thing for your uses.

    • #13687
      AJC
      Participant

      @123Toid – I don’t know if this is a setting in windows (I run a custom rom and it does interfere with numerous software from various vendors), but the numbers I am getting off this sub is pretty far from the spec sheet and winISD is having the SPL way lower than what it should be when I enter the numbers in.

      Now, I do think the speaker needs broken in could be part of the numbers. But is the winISD having the SPL more like 15-20dB lower than it should be normal?

      This is the scanspeak 30W/4558T00. Granted, this is sub number 1. I have 3 more to check. Just trying to make sure what I am seeing makes sense.

       

      Edit: nevermind. I had the Vas set to liters in winISD, but was putting in cubic feet from DATS. My bad.

    • #13695
      AJC
      Participant

      So, some information from DATS on the 4 subwoofers (30w/4558T00 Discovery line from scan-speak, which although listed in two ways for linear excursion, 12.5mm is the real one way excursion with the limit being 28mm, whereas elsewhere they listed the 2-way, which is 25mm and 56mm; https://www.scan-speak.dk/datasheet/pdf/30w-4558t00.pdf).

      Here is running DATS on each speaker 11 times, with a reboot and retesting of the leads after the 5th run for each speaker:

      When the averages were plugged into winISD, this is the rough look, with recommending 4.65 cu ft and 4.965 cu ft (this one I ran a little while with the generator in DATS at 20 amplitude at the speaker’s resonant frequency, so it being larger by more than a tenth of a foot compared to the next closest, 4.837, may explain why a larger disparity).

      This is with high pass set for 14.5hz and low pass set for 120Hz, with a port resonance above 380 Hz (more than two octaves above).

      I am thinking of doing 5.5 cu. ft. with a 21.65 Hz tuning frequency. These don’t hit as low as the spec sheet suggests, and the resonant frequency on all of them is above that of the spec sheet by 4-4.5Hz. That is fine for my purposes. They also have a sensitivity of about 2dB higher than the spec sheet. Also fine. The Qms was also significantly higher, while the Vas was closer to 133L instead of 197L. The diaphragm diameter was spot on, within 1 to 2mm for the most part, meaning any difference could have been my slight placement of my ruler or deformation with the ruler (I grabbed the blade from my combination square because it was handy, but is more hefty than a lighter steel ruler).


      The packing is awesome. the subs were in plastic bags, then placed in two piece foam formed to hold the subwoofer and support all critical parts, followed by a thin piece of cardboard wrapped one way, then a white strand the other to hold both pieces of the foam together.


      As you can see, that is a pretty solid looking product. With that said, with my altitude, I cannot run this more than 120W or it will exceed the cone excursion of 12.5mm. That is fine, because volume leveling these to the woofer and tweeter will only take around 90W due to the 2dB extra sensitivity. So, it looks like this speaker won’t need to pull as much wattage to hit the SPL I’m targeting for the total FR.

      Now I’ll need to go back and check the specifics for the compartment size of the SEAS U18RNX-P and start figuring out the size. I’m planning on a slot vent at the bottom with two separators, so it is calculated as three separate vents. But this helps to keep the resonant frequency in check, considering I’m using two pieces of plywood together for each separator and for the shelf for the vent. Hence estimating port resonance to be 380Hz and above.

    • #13696
      tvor-ceasar
      Moderator
      Posted by: @ajc9988

      Edit: nevermind. I had the Vas set to liters in winISD, but was putting in cubic feet from DATS. My bad.

      I’m glad you posted this because I was having similar problems with a driver I was putting into WinISD. Sure enough, there it was in Liters, not Cu.Ft.

      👍👍


    • #13697
      AJC
      Participant

      @tvor-ceasar – It’s something so simple and so easy to overlook. In fact, the meters and feet conversion wasn’t done on a rover sent to Mars about a couple decades ago. Guess what happened to it. (hint: it came in too quickly and demolished itself; https://www.latimes.com/archives/la-xpm-1999-oct-01-mn-17288-story.html ).

      But I’m glad that helped others here! Also, WinISD uses 1K on the L(e) value. At least my default under preferences in DATS was at 10K. So check that you have L(e) set to 1K there. Edit>preferences then in the menu, select show Le at 1 kHz. You might also want to set the SPL to 2.83Vrms/1m instead of 1W/m, but that is a personal choice.

      You can also set to use metric if you would like for units and/or volumes. Just so you can speed up your workflow in the future.

       

      Edit: I found the 1kHz thing posted by a Chris Romer in this thread over at TechTalk sponsored by Parts Express: https://techtalk.parts-express.com/forum/tech-talk-forum/55177-ts-parameters-from-dats-insufficient-for-winisd-database?p=820260#post820260

      Edit 2: I also forgot to mention I’ve tested all the drivers for rub and buzz using the highest test levels and the drivers have all passed.

      Although, it is easier to get the pass if you just keep going quickly rather than letting it settle after hearing the tone for the notification that pops up. So just move quickly and it will read well when clicking through. If letting it sit for awhile, it just seems to not track as well and I do not know why. But doing it that way gives a consistent result. And obviously, going to the normal guidelines also makes it pass easily. -40db is really quiet.

    • #13699
      123toid
      Keymaster

      @ajc9988 

      Excellent writeups!  I am a sucker for the packaging.  SO I am glad to see it pacakaged so well.  THis si looking like it is going to be an awesome project!  I can’t wait to see the rest.


    • #13706
      AJC
      Participant

      @123toid – I’ll be coming up with a parts list for the build soon enough, a mandatory tool list, a recommended tool list, etc. I’m really thinking of doing a write up how to build this exact project because, although expensive, it really is shaping up to be an excellent product. Due to the parts selected, there are 8 types of screws (possibly a ninth, unless I can make the one for the USB port also work to bolt in the power plug connector), 4 types of quick connects for 12-10 AWG (yellow), 1 ring connector, the rubber feet, three types of wood inserts (no T-nuts because I’m using a double baffle for the entire front and a triple baffle for the sub, with around 3/4in inserts for the speakers themselves, 1in inserts for the feet and to screw in the plate amp (6mm, so you can switch to screw in spikes if you care to at a later date), the sub and woofer have 5mm holes, the tweeter is 4.5mm, so had to use a different size there, but looking at a 4 bar and 3 bar speaker grill for the sub and woofer, which would have 5mm holes, meaning that there would be a total of around 88 of the inserts and screws for the grills and sub and woofer. I figure just use the 6mm on the plate amp as well since the only thing I needed them for was to strongly secure the feet. So that will go with the amp as well, meaning 40 of those are needed (I’m building 2 pair, so those numbers can be cut in half for a single pair).

      Then I have two different standoffs for the PCBs from the aluminum plate (I bought 4 12x12x1/8in aluminum plates to cut down, then drill holes for the offsets, power switch, knobs, etc., along with cutting out the holes for the USB and power connector which has a fuse in it). I’ll likely find a way to mount the bluetooth antenna (WIP). Then I have to write up about sanding to remove the aluminum oxide, applying the metal primer, then painting, how to make cardboard stencils, using that to paint the names identifying the different knobs, etc.

      Then there is the use of finger joints for the sides, using the dado to create the top and bottom and some of the channels, making the vertical bracing open up so that it holds the vent shelf in place, adding more stabilization for the vent (which the bracing will be two vertical, two horizontal, with the vertical and horizontal slotting into each other, and since it is two plywood sheets thick, the horizontal ones will be done as a two piece, the piece that goes vertically up the back and runs the top and bottom of the compartment with the sub (with the bottom then splitting open with it continuing to the front of the speaker the height of the vent, then above the shelf for the vent coming toward the front) and then the top to bottom bracing in the middle. This way the wholly intact horizontal bracing slides into the vertical bracing at the back, then the cross brace in the middle also slides into the top to bottom bracing, thereby creating a tic tac toe type looking brace in the space of the compartment. It’s easier to slide in, then glue those middle bracing with a section on one of the plywood parts of the brace cut out to create the joint for the middle bracing to the rest of the vertical bracing. This is with dados cut with either the table saw or the router for the bracing to firmly slide into and glue down, creating a strong mechanical connection to the outer frame of the compartment.

      Or at least those are my thoughts so far. Once I have measurements finalized and sketched up, then I will be closer to finished. Also trying to finalize what insulation I will be using for each compartment and the sourcing (most likely Dacron in the sub compartment and lining the entire woofer/tweeter compartment with 2″ rockwool 6 rigid board with dacron covering the wood from the braces, which should deal with reflections and standing waves).

      Then pretty sure I will be using oil based finishes on the plywood, round-overs on the corners (which should look interesting with finger joints), an angle (undecided yet) for the transition from 3 baffles to 2 baffles, and throwing a couple pieces to back the area for the plate amp, which will need routed so the plate amp can be flush with the back, while also using the 1 inch inserts, although those are the only inserts I do not care if they go all the way into the inner cavity, whereas everywhere else on the build, no screws create any holes into the speaker compartments, meaning a good beading of silicon around the inside edges of each compartment should have a relatively sealed compartment for the speakers.

       

    • #13732
      AJC
      Participant

      @123ToidI would like your opinion on whether the sub compartment really needs insulation. Doing the math on 0.5″ insulation, it would take nearly 20% more space over the volume calculated for the box by winISD, while the effect of lining with a 0.5″ insulation does little to effect frequency range below 125Hz (see http://www.bobgolds.com/AbsorptionCoefficients.htm for a cheat sheet of absorption coefficients, with 1 being total absorption of the frequency in that range and 15% (or 0.15) having basically no effect. So Dacron or similar in the sub compartment, unless lining all walls with 2″ 703 or rockwool #6, would basically not do the job (and 3″+ would just balloon the box so large I would not build it for a tower speaker). Reason I ask is you did a kit recently, the CSS box, that resembles somewhat this other box (GSG Audio Designs https://shop.gsgad.com/collections/15-inch-marty-sub-flat-packs/products/marty-15 ) from which I drew inspiration from for this design

      Now, imagine two of those as the vertical bracing which also holds the shelf for the vent in place at the bottom. Then, imagine two of the normal braces you have designed that set back for the speaker going horizontal attaching at those notches in the vertical brace (part of the reason with this design that I would actually have the central bar going to the top likely as a second piece of wood to glue in place so that those horizontal braces can be internal to that space to hook into the vertical braces).

      So, the question is, with that much bracing, is adding a 0.5″ lining of dacron worth it?

      The upper compartment with the woofer will be fully lined on the walls with 2″ Rockwool RXL 60 (except for where the frames connect, as the frames will be 1.406″ thick (2 pieces of plywood) and 2 inches from the wall to the center), with Dacron lining the wood of the bracing. I will likely do 2 window braces in that compartment.

    • #13733
      123toid
      Keymaster

      @ajc9988 

      Typically open cell foam won’t take up volume.  However, I rarely use foam or insulation in the subwoofer compartment. Typically just good bracing is all you need. There’s a lot of people who believe subwoofers don’t need any your of insulation as the frequencies are too low (long) to really have any effect. Although I’ve never tested that.


    • #13742
      chedwin
      Participant

      @ajc9988 i would say there is very little point lining the sub compartment. absorbtion coefficients like those listed on that site are very usefull in the right circumstances such as sound treating a full room but they arent neccessarily hugely applicable to a speaker, especially a subwoofer chamber, due to the way the tests are conducted.

      without going too technical, absorbtion coefficient is based on reverb time of each frequency in a room (sound reflections bouncing equally accross the entire room) whereas adding insulation volume to a speaker helps reduce standing waves and lower the decay time (frequency specific spot reflections based on the geometry of the bounding walls)

      Reverb time and decay time are similar and easily confused but they are importantly different enough to each other the distinction should be made, you can think of it like a light bulb vs a laser pointer reflecting off a mirror.

      when dealing with musical content such as a bass line in a song, or sharp bass hits like gunshots in a film the decay time matters more than the reverb time for percieved sound quality in terms of punchiness and sounding tight/controlled. to adequately control standing waves below 100hz can take from 15cm (6″) upto 40cm (16″) or more depending on the density. for example im using 20cm (8″) in the corners of my 2.5x4m (8x13ft) home studio mixing room

      also in a speaker sized box made of wood the sides of the box will be near enough invisible to low frequency sound waves due their length and how much more energy they carry further reducing the effectiveness of adding insulation


      Josh Evans, Professional Live Sound Engineer, High End Commercial AV Install Technician
    • #13743
      AJC
      Participant

      @chedwin – Thank you for the input. It is appreciated. Nonetheless, and unfortunately, there are few resources that have taken the insulation, placed it within a cabinet with the rest of the system being controlled, and systematically testing the ability to reduce resonances or standing waves within the enclosure. You can see some on the material’s effect in a closed enclosure in the book Loudspeaker Design Cookbook, but there are many more types of insulation out there and very little on testing each type (whereas many do not want to put their whole lineup under testing because it undercuts their marketability of their tested solutions, in part).

      If I am understanding you correctly, then, you would be more ready to recommend, for a subwoofer enclosure part of a cabinet, constrained layer damping (placing a viscoelastic polymer or sheet between two solid materials, such as plywood>viscoelastic material>plywood) which would likely help to reduce box resonances due to the nature of converting the mechanical energy into heat for dissipation. That was something I toyed with the idea of doing in addition to insulation, but decided against for this project.

      I also, in considering the framing for my box, reviewed this from audioholics: https://www.audioholics.com/loudspeaker-design/detailed-look-proper-loudspeaker-cabinet-bracing/finite-element-analysis-part-i

      This used LEAP 4 or 5 to model the placement of braces along a plane and the effect on stiffness and displacement of the plane. It’s an interesting read.

      Other than that, still a noob to speaker design (although I have progressed a lot since I started). So please take the above as speaking from ignorance and feel free to correct me as needed.

      Now, the way in which I used that reference for the insulation is just to estimate possible efficiency to address standing waves in the enclosure. Although I did roughly look at it on the sub part of the cabinet, it also informed which materials would likely deal with the reflections inside the upper compartment with the woofer (I decided on cost, likely just getting Roxul 60 rigid board, which is cheaper than the denim acoustic batts for what I have found so far, mainly at ATS Acoustics, but if you know a better/cheaper source, I’m all ears). I mixed that information with the information found on this cheat sheet of sorts (table of frequencies and wavelengths): https://www.jdbsound.com/art/frequency%20wave%20length%20chart%202013.pdf

      That is more to give an idea of, where a standing wave appears, which frequency to look at and what to try to trap for the frequency (and how the box design itself allows for a wave of a size to exist and perpetuate itself). But, once again, a noob.

      But once again, thank you for the input. I do understand those values are not perfectly translatable for use in this case, but it does help ballpark for when it makes sense to use what (such as knowing which insulation is even capable of reducing well a specific frequency found in a standing wave).

    • #13744
      chedwin
      Participant

      @ajc9988 i should have worded my previous post better to make clearer that when im talking about the data not translating to speaker cabinets i meant for the overall cabinet/main low frequency chamber or a standalone subwoofer box not smaller internal chambers for other drivers. what you have done and they way you have referenced the data to find the best material for mid and high chambers within the main cabinet is absolutely fine. appologies for the oversight in detail on my part there

       

      in terms of the sub chamber you are in fact correct that i would be more likely to reccommend constrained layer damping for a sub chamber or a standalone subwoofer. from a purely scientific point of view contrained layer damping would be much more effective than single layer ply and insulation and in something like a 2 x10″ standalone sub or bigger i would strongly reccomend considering it. however the practicality of it has to be determined on a case by case basis for cost, time, effort and other measures already taken.

      if you have already done FEA calculations for your bracing placement then it sounds like you are already at a good place for the sub chamber, the bracing alone should be plenty to get the best sound out of the speaker. in this instance i would say constrain layer damping isnt needed on top of current measures taken

       

      hope that clears some things up a bit. cant wait to see how these turn out 😀 


      Josh Evans, Professional Live Sound Engineer, High End Commercial AV Install Technician
    • #13746
      AJC
      Participant

      @chedwin – No apologies necessary! Although I found those resources for generally designing the mid-high chamber, I was also trying to figure out whether it could be used efficiently for the sub chamber. But after reviewing the materials, the designs, etc., it just didn’t make sense (unless trying to greatly explode the size of the cabinet in order to fit the volume of thicker insulation needed). Part of this exploration was prodded on by some applying insulation to the design known as The Hammer.

      But considering the box was a kit and others building the kit did not use insulation, there is the chance the design was never meant to have insulation and the dimensions not adjusted to account for and accommodate the insulation.

      As discussed though, the materials just are not really able to handle low frequencies very well without going insanely thick, etc. And, as we both agree, constrained layer damping would likely do much better to control box resonance than insulation in any event.

      I have not yet done FEA calculations for the bracing placement, but do plan on testing my design. The inclusion or exclusion of insulation for the sub chamber was the primary last factor before finalizing the design. I then would like to move that design over to an FEA software that can import the design from sketchup. From there, it is using the plywood’s values in order to roughly test resonances, bracing locations, etc. I was inspired to do FEA from that article, but from the beginning used the findings in that article in order to determine what dimensions should be used for the bracing (like thickness double that of the wall materials, the length away from the wall impacting the influence on the modal frequencies, etc.). It also made me think about what can count and act like a brace within the structure, such as a full shelf to separate the upper chamber, which carries both the chamber for the plate amp and the chamber for the woofer and tweeter (the tweeter is a sealed design, so I didn’t need to design a third chamber to support the tweeter).

      Now, to add constrained layer damping (hereinafter “CLD”), all it would take is adding a line to my spreadsheet with the thickness of the layer, then adding a second layer of plywood thickness to the outside dimensions of the cabinet, widen the front baffle (trying to keep the product to 4′ high and 2′ deep, so widening the bottom outside from about 17″ to accommodate the extra area would be doable; in fact, then you could design the bottom sub cabinet to slide into the overall outer frame for the speaker cabinet, allowing for further compartmentalizing in construction and testing) in order to come up with the design for that. I have created a worksheet to automatically adjust the dimensions by updating values. So I might create two designs, one with CLD and one without. Then just run the FEA on the two designs.

      Either way, you are probably correct, especially since, due to my altitude effecting air pressure effecting the linear excursion point of the sub going in, and thereby the sub being limited to under 108dB for each tower (the target was to make tower speakers which could play up to 105dB, just because, which so far the Scanspeak 30W/4558T00, the SEAS U18RNX/P, and the SEAS Titan 27TAC/GB can all reach, so that should be doable with long term power limits of the speakers, with long term defined by the ISO meaning thereof), which means it isn’t playing as loud as it may play in other scenarios (likely remaining around 70W for 105dB).

      But, I will have more as the design is about to enter into the sketchup phase for what the math says the dimensions should be, which then will lead to the FEA calculations.

      Just wanted to say you were not incorrect in understanding the question. Also thank you for the design input! I really enjoy gaining any and all insight that I can get.

      Edit: What materials or products would you use for a viscoelastic layer in a CLD cabinet?

    • #13747
      chedwin
      Participant

      @ajc9988 Im not really able to give you mid layer material suggestion for CLD cabinet as usally when im applying layer damping principles its in the building construction design phase for bars and clubs using comercially purchased subwoofer(s) not self designed cabinets. For example im currently working on a concrete and gypsum board design with a triple 1200W 2×12″ subwoofer array for a high end bar installation.

      I can share the rough draft drawing for it showing how I normally use layer damping concepts in my work, this one is free layer damping where the gympsum layer is the damping material. Grey is large poured concrete block with a cavity cut out for the sub array to fit in, blue is gypsum board lining top, sides and back, red is wooden battens to isolate subs from floor and yellow is a plywood surround. The cavity in the concrete is about 2m wide and 3m deep.

      I dont think any of the materials I would normally be using in buildings would be suitable to CLD a speaker cabinet. You could try 1/4″ rubber gym matting/flooring. Never tested it but thats the best thing coming to mind right now that seems worth attempting


      Josh Evans, Professional Live Sound Engineer, High End Commercial AV Install Technician
    • #13748
      AJC
      Participant
      Posted by: @chedwin

      @ajc9988 Im not really able to give you mid layer material suggestion for CLD cabinet as usally when im applying layer damping principles its in the building construction design phase for bars and clubs using comercially purchased subwoofer(s) not self designed cabinets. For example im currently working on a concrete and gypsum board design with a triple 1200W 2×12″ subwoofer array for a high end bar installation.

      I can share the rough draft drawing for it showing how I normally use layer damping concepts in my work, this one is free layer damping where the gympsum layer is the damping material. Grey is large poured concrete block with a cavity cut out for the sub array to fit in, blue is gypsum board lining top, sides and back, red is wooden battens to isolate subs from floor and yellow is a plywood surround. The cavity in the concrete is about 2m wide and 3m deep.

      I dont think any of the materials I would normally be using in buildings would be suitable to CLD a speaker cabinet. You could try 1/4″ rubber gym matting/flooring. Never tested it but thats the best thing coming to mind right now that seems worth attempting

      A couple of the materials I have considered to date (some of which agree with your suggestion) include:

      1) isolation material for sub-flooring, like Peacemaker underlayment ( https://www.audimute.com/peacemaker-soundproofing-underlayment );
      2) mass loaded vinyl;
      3) viscoelastic polyurethane (two pail mix type, sometimes used in marine and engine rooms for noise isolation, etc.; not meaning green glue, just am spacing on the name of the company I did think of getting some from); and
      4) bitumen rubber.

      I would look for cheaper construction materials that have similar properties, but many have VOCs that both limit for inside use and also the chemicals must be checked in comparison to the construction materials of the drivers themselves as the VOCs can harm some speaker drivers.

      But I did not think about rubber gym matting rather than underlayment. Might need to price that out to see how well it works.

      Edit: Pyrotek’s Decidamp DC30. That was the poly barrier I was thinking of. https://www.pyroteknc.com/products/decidamp/decidamp-dc30/
      Which I would also take recommendations on market alternatives to decidamp dc30 as well.

    • #13750
      chedwin
      Participant

      @ajc9988 sub flooring isolation material like that one is maybe a bit on the thin side for a CLD speaker, better suited to stopping vibrations from footfall but could be usefull on the bottom of the cabinet to help isolate the cabinet from standing surface though depening what its sitting on. looks like they have separate thin/dense to block sound waves permeating a surface and thick/less dense to stop vibrations travelling along a surface. thick and dense would probably be more ideal in a speaker

      mass loaded vinyl should work, i have 5kg/m2 mass loaded vinyl as an underlay for my LVT floor instead fo regular underlay. although similar to the sub flooring isolation material MLV is often available thicker whilst maintaining density. MLV should also be available cheaper than an underlayment, at least here in the uk it is

      paint on solutions like DC30 should work very well acoustically but are very permenant, only use if you know you never need to separate the surfaces for any reason

       

      of the mentioned options if say rubber gym mats or suitable density/thickness mass loaded vinyl if you want a less permentant more flexible solution or DC30 (or simlar) if your happy to sacrifice the ability to take apart for some improved performance

       


      Josh Evans, Professional Live Sound Engineer, High End Commercial AV Install Technician
    • #13795
      AJC
      Participant

      @elliottdesigns – Update on the ADAU1701 programming. I made my first attempt to do a mono Left channel programming for the upcoming speakers, with a 3-way double precision Linkwitz-Riley 48dB crossover, an infrasonic filter with a 6th order Butterworth (brought down the top dB of the sub, but the sub can now reach deeper with F3 of around 19.48Hz), an FIR filter with 550 points, a limiter, a gain per driver (sub, woofer, tweeter) potentiometer and overall volume knob, a signal detection circuit to turn it off when nothing is coming, and inversion switches in case you plug the drivers in backwards.

      Note, this is before programming in any solutions in the event that time delays or phase are issues.

      Now this is a first attempt, so please do not think this is done. It compiled without an error, but that is not the same as a finished, working product. I also plan to make a version that uses SigmaStudio’s autoEQ to check results.

      This is the output data from the compiler:

      ################## Summary ########################
      (Note: Estimates are based on a 48 kHz sample rate)

      Number of instructions used (out of a possible 1024 ) = 1001

      Data RAM used (out of a possible 2048 ) = 766

      Parameter RAM used (out of a possible 1024 ) = 683

       

      So, as you can see, I’m running out of the finite instructions allowed (the ADAU1701 is limited as it can only process so many instructions in a given period). What I might start looking into is adding an ARM developer board to the mix (analog discusses this as using a microcontroller with the DSP, with a recommendation of an arduino like the Teensy 4.0 available at https://www.pjrc.com/store/teensy40.html for $20). It would take learning more depth on the programming to integrate it on the software side, and adding an additional mount to my home-made plate amp, but that would greatly increase the number of instructions allowed, which also means having an even better FIR filter, thereby getting more comparable to a SHARC (which has a version with a 1GHz ARM processor, or the mini with the 500MHz cores). Granted, the Teensy 4.0 only has a 600MHz frequency, thereby being more comparable to the mini, but this is only the start of my search. I need to see what other boards have been paired with the ADAU1701 (or analog chips, generally) and which people have already poured the hours necessary in to make them work together. If I find that, then adding in a board will happen (if not, but I find enough on adding the Teensy 4.0 or 4.1, then I’ll do that, because $80-120 for the boards and standoffs is not bad).

      2441-Design-try-1.zip

       

    • #13796
      Elliott Designs
      Participant

      @ajc9988 

      Nice, I’ve been working on moving over to raspberry pi 4s, yes they are expensive and very involved, but once I’ve figured out, on a per speaker basis I will have more DSP capability than the 2×4 HD, better sound quality, and the capability to add a music server, WiFi capabilities to allow for Spotify streaming, multiroom playback and even a smart assistant. Once I’ve completed all of this (which will take a long while) I’ll be completing a comprehensive guide on how to do it on this forum.


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #13799
      AJC
      Participant

      @elliottdesigns -https://digital-audio-labs.jimdofree.com/english/raspberry-pi/adau1701-i2s-driver/

      One of the first searches for rpi I did popped that up, which means there is a way to use the i2s to possibly have the rpi as the microcontroller for the 1701 boards, which will be cool to have that with the jab5s I already have.

    • #13802
      Elliott Designs
      Participant

      @ajc9988 Yeah, you could do that, the audio systems for processing the audio pipeline and doing the DSP are in their very early days and are next to impossible to get working at the moment, let alone easily. That’s why I’m working through it at the moment and then I’ll be creating the guide. I’ve been spending 2-4 or more hours a day for the past two weeks on figuring out what I can get working, so I’m trying to put the time in for you all, just saying before you go in and try getting it to work yourself, trying to save you the effort 😅.


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #13805
      AJC
      Participant

      @elliottdesigns – I already have the JAB5s. I have for months. And I moved the signal detection circuit earlier. I stopped the popping sound I was getting with sudden power offs and honored the filters better. The subs have cleaner sound with proper infrasonic filter and crossover applied. But there is definitely more work. Just hoping the price of plywood drops soon…

       

      Edit:
      Also, since I already have the ADAU1701, I’ll be working from that end on posting general sigmastudio builds and learning on using an arm chip tied in potentially. If the RPi by itself or tied with a 1701 work better, then that will be included down the road. But, looks like we all have some really cool projects working!

    • #13860
      AJC
      Participant

      @elliottdesigns – Hey, so how many coefficients can you run on the RPi 4? The arduino solution can do 20,000, but in 0.367 seconds. Trying to check on the RPi 4 and how much memory is needed for the number of coefficients processed. Mainly because I can buy a 2GB RPi 4 now without an issue to start toying with development, but need to know if the NEON and FP are that much better that it can spank the coefficient processing of the arduino solution.

    • #13862
      Elliott Designs
      Participant

      @ajc9988 ah, not sure about latency since I use linear filters and make sure each of my driver’s have the same amount of taps of processing applied to them, so sorry I cant help with that part. I’m using a pi4 4gb, not sure how many the 2gb can handle but I have no where near reached the maximum the pi is capable. I know lesser hardware has been able to run upwards of 1,000,000 taps in total. So 🤷‍♂️. Also what is NEON and FP?


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #13863
      Elliott Designs
      Participant

      Oh, just found out that this figure took place in 400ms. But latency varies widely depending on what audio pipeline you use. For example pipewire is the best by far but it’s still in early-ish development, although perfectly usable for the most part


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #13864
      AJC
      Participant

      @elliottdesigns – FP = Floating Point, which the ability to do floating point calculations can help expedite the execution of the coefficients. Neon is an Arm Neon is an advanced single instruction multiple data (SIMD) architecture extension for ARM processors they started including.

      Neon instructions allow up to:
      • 16×8-bit, 8×16-bit, 4×32-bit, 2×64-bit integer operations 
      • 8×16-bit*, 4×32-bit, 2×64-bit** floating-point operations 

       

      The extra compute is where the A series, at a certain point, overtakes the inherent abilities of the DSP module within the Cortex-M series microcontrollers. It takes more footprint, but the additional compute really can give it run away advantages.

      And that is why I was trying to figure out the coefficients per time period on the A72 architecture (IIRC that is the Arm version for the RPi4, that or A73, but I think it was 72). The main reason is for this to work with a TV as these speakers will not solely have music. The delay doesn’t matter for music so long as all channels are synced, which the same number of taps helps to do. Instead, you can only put a delay on the TV for so long, so being able to finish the FIR filter in a reasonable time is something I’m also considering in the design.

    • #13865
      Elliott Designs
      Participant

      @ajc9988 oh, ok. Raspberry Pis really are designed to run using an operating system, I’m implementing my FIRs with a convolution plugin built into a package called EasyEffects which I use on an operating system called Manjaro, but then again, we have very different use cases. Seems like for your use the Arduino is probably the better solution, especially for the price! I’d love to know how you get on with it!


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #13867
      AJC
      Participant

      @elliottdesigns – I’ve actually found some interesting ways to accomplish using the RPi, although some applications actually have the RPi as a slave due to a specific needed clock generator found on the ADAU1701 but not the RPi, making it easier to make the RPi a slave. Others allow different builds and using drivers to support the ADAU1701. So, if need be, and if the results would be better, I could incorporate the RPi which would have better support for things like ethernet, while still using the i2s to connect the two.

      I was wondering, how does your memory usage look on it while running your FIR?

      I also could do a different layer of abstraction, where I integrate the arduino microcontroller working with the ADAU1701, then setup an RPi for a pre-amp to take in signals and then output them to the speakers.

      This is why I’m trying to research if this can do what I need, as the arduino plus the SRAM and PSRAM chips puts it at $30, and if I grab a couple other things, the price goes up, while I can grab the RPi4 2GB for around $35-42, depending on the website. Prices excluding shipping. So at this point, I am still open for which row to hoe, so to speak.

       

      Edit:

      Explanation of clocks needed to support system timer for RPi:
      http://www.crazy-audio.com/2013/09/raspberry-pi-auda1701-dsp/

      https://ez.analog.com/dsp/sigmadsp/f/q-a/65427/connect-raspberry-pi-to-adau1701-via-i2s-bus

      More on being slave:
      https://community.volumio.org/t/adau1701-dsp-and-slave-i2s-kernel-module/1962

      https://www.raspberrypi.org/forums/viewtopic.php?f=44&t=8496&start=400

      Drivers:
      https://github.com/MKSounds/ADAU1701-I2S-Audio-Driver-for-Raspberry-Pi/blob/master/adau1701-i2s.dts

      Audio Codec:
      https://github.com/raspberrypi/linux/blob/rpi-4.1.y/sound/soc/codecs/adau1701.c

      https://github.com/torvalds/linux/blob/master/sound/soc/codecs/adau1701.c

      These are some of the pages I have looked at, but that I could find again quickly. The one about the clocks on analog is better. 

      Also, I haven’t heard Manjaro in awhile. I never used it, but my friend in undergrad was using it while going to school for comp. science. I still haven’t made the jump to linux fully, but can get around well enough.

    • #13895
      AJC
      Participant

      @elliottdesigns –  So, because the 2GB version is sold out so many places, I sourced a 1GB version to start work to see if it is viable for inclusion as a slave. I did so under the theory this has 1MB of L2 cache and 1GB of memory. If the code is optimized enough, it should not need to hold too much in memory in order to process, especially since the current research suggests for lip sync, you can go -125ms to 45ms and there still be sync fine for people. If true, and looking at 20,000 coefficients taking 0.367 sec., or in your example 1M coefficients in 0.4 sec (400ms), in theory it would only be able to do around 100,000 coefficients or less to return the signal within the time period for lip sync. Considering this is still higher than the theoretical 2,000 for the Teensy 4.1 or the 550 I currently have set on the ADAU1701, so long as 1-2GB RAM is enough for the overhead (which is considerably more than is found needed on the other products, but doesn’t mean the additional overhead related to the differences in products doesn’t necessitate more memory), then there is a chance, even though expensive, that it may have the most robust solution for applying a FIR filter in this case for what I have already purchased.

      Many do not have the 2GB version available and are back ordered until October to December. As such, just starting development now may be the way to go, so that if it is a dead end, I can drop the idea by sometime next month and go with another route.

    • #13930
      Elliott Designs
      Participant

      @ajc9988 Thought just occured to me, how are you using the Arduino for audio processing? I would have thought it was only 8bit capable, no? I’d love to hear more about what you are doing with it!


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #13931
      AJC
      Participant

      @elliottdesigns – I’m starting with trying the RPi4. But this is where talking shop on specs of the Teensy 4.1 come in, discussing arduino code ran on other platforms (you can run it on RPi with modifications, part of the reason I decided to try to make the RPi work first before switching back to the Teensy), and how this would work.

      First, let’s talk the Teensy 4.1 specs. It uses the ARM Cortex-M7, which has the built in DSP dedicated hardware. https://www.pjrc.com/store/teensy41.html

      As you can see, this can do floating point 32 and 64-bit math, 8MB flash memory, 1MB Ram, expansion for ram or flash chips on the developer board. So this isn’t exactly arduino, and that gets to the next point.

      Analog Devices, the maker of SHARC, the ADAU chips, sigmastudios, etc., has already worked with the Teensy 4.1, running arduino code, to run as a slave device to their other products. They have done this because the Arduino Uno or Duo (forgot which ATRM) was once used in conjunction with their projects to increase crunching of numbers, so they have how the code interacts with their software down. You could make it work with other code, but there is something to not recreating the wheel. So when I brought up Arduino, I was meaning the arduino code which can be more readily optimized for use with RPi for my purposes than recreating it in a deeper dive into raspbian optimizations, etc. Instead, it is doing the changes to the arduino code they already have, which they already adapted for the Teensy 4.1, to also work with the RPi 4.

      Analog has an export function of the code created in sigmastudios which can work with Arduino (there is a little more to it than this, but in essence, it is this exporting that is why arduino got brought up). In other words, when you have the devices connected to sigma studios, you can design the flow of processing to include the offloading of the DSP functions that are faster on the modern ARM processor, especially with NEON for floating point calculations and some integer work otherwise, to then increase how much you can do with FIR filtering and any advanced additions you might have to make, such as using all pass filters cascading to address phase shifts if needed, etc. So the arduino code, which already has been worked on, is the important part, not an actual arduino itself, which at this point is slower than the Teensy 4.1 or the RPi 4.

      So, what I am doing is having the signal come into the ADAU1701 JAB5 board, mainly because it has to be the master due to the master clock which is used for the timing signals related to the music processing, with a 12.288MHz crystal, meaning 48,000Hz native. You then use the i2s in and out on the board to send the signal to the developer board. Normally I’d prefer one jump, going from the RPi as master to the slave ADAU1701, but to do so, I’d have to develop the external clock board, then have both point to use that for MCLK, which the JAB5 has more limited pin connections to do this with. If developing the board from scratch, you could throw both chips on the same board and have more direct communication without as much jitter through the connecting of two boards, while optimizing one as the co-processor to the other. But that is not this project.

      After setting up the RPi 4 or Teensy as slave, you take in the signal from the master (I need to play with this to see if I can create a scenario to allow input from the RPi4 to somehow be put through the processing loop, but that is why I wanted to start working on the project now to figure out the limitations of doing it this way, which isn’t as well documented as I would have hoped), then route it over the i2s out to the RPi appropriate GPIO pins you change the arduino code to point to, then have the RPi apply the FIR filtering, returning the signal back to the ADAU1701 over the i2s in, to then have the three channel crossover applied to split it to the individual speaker channels, which then gets sent to the integrated amp and onto the speakers.

      Now, I did mention about other inputs. I know people have gotten the CEC to work with an RPi so that a remote can control outputs on the RPi. But there isn’t much information on if the RPi 4 has the pin for ARC connected or not (on the B+ from around 2014 or 2015, it was not). So determining that is of import, because if the hardware is capable, then it moves to software side, which once again, I’m not sure I can add inputs to the slave device into the chain, even though I know outputs on both master and slave are possible. This goes with also using the internet connection on the RPi as well.

      Regardless of the additional inputs, the goal is to add the superior crunching power of the RPI4 chip into the line in order to apply a better FIR filter to the signal, considering it should greatly exceed the 550 coefficients of the ADAU1701.

      Now, this is to do fully in speaker. Why? To basically make the speaker response as flat as possible and to minimize additional processing, considering you could place the speaker and then calibrate the FIR filter meaning the filter is serving both as the room correction filter and the speaker response filter. That reduces the need to do that using a receiver. And sure, there are questions on number of times through the ADCs and DACs, quality of the components doing the conversion, etc. But what I envision with these speakers, since they are powered, is first getting all that correction dealt with on speaker so that at most you need is a pre-amp, if not simple audio extractors or HDMI audio decoders, etc. to split out the source to go to the speaker to do the rest of the work. Eventual development into further products incorporating better quality components can come later. Proof of concept is first. And that is where using previously tread ground in order to integrate these into a plate amp is my first step.

      Does this make more sense now?

    • #14010
      AJC
      Participant

      @elliottdesigns – so, an update if you are considering using Raspberry Pi with ADI products (Analog Devices Incorporated, which makes the ADAU line of codecs and SigmaDSPs, Blackfin, Sharc and Sharc+, DACs and ADCs, etc.), ADI has their own version of the Linux kernel for Raspberry Pi specifically. Here is the link to that:

      https://wiki.analog.com/resources/tools-software/linux-build/generic/raspberrypi

      The documentation is a little outdated as the most current version of the OS that they have is 5.4.y, not 4.14.y.

      https://github.com/analogdevicesinc/linux/tree/rpi-5.4.y

      This can be used by also creating driver modules from their more detailed drivers on github. I attach the image I copied into /boot, replacing the kernel7l.img file in that folder. Obviously, if you need customization, it is better to just git the list and compile your own kernel, using make menuconfig to customize as needed. I made very few changes from their stock config, so if you need more specificity, something to consider.

      Today, I will be trying to get the programmer to communicate with the RPi over I2C and getting the ADAU1701 to communicate with the RPi over SPI (and figuring out where to connect each of the clock generators to get proper data transport between the two). Then, I will see where to go from there once all the hardware can talk to each other, including, if needed, signal delays put in if needed so that both master and slave don’t collide on trying to command the port on boot, etc.

      How’s your progress going?

      2580-zImage.7z

    • #14012
      Elliott Designs
      Participant

      @ajc9988 Thanks for the update, got the Rona so not too much has changed. Temporarily leaving the smart speaker side of the project alone and going with a simple FIR DSP with the Teensy for now, so I can save money and get at least a 4.0 setup working. Haven’t got myself a Teensy board yet to test it out, but I’m sure it will be fine 👍.


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #14014
      AJC
      Participant

      @elliottdesigns – sorry to hear about that. I had it last year toward the start. Had hot and cold sweats all night, aches all over, doing nebulizer treatments, oral steroids and IV steroids for when I went to the ER, and a prophylactic of broadband antibiotics to prevent secondary infection. It lasted about 10 days for me, was about like catching a bad cold or a flu (for context, I’ve had pneumonia that ate a hole in the left bottom lobe of my left lung filling it with blood where they thought I had a punctured lung; I was life flighted and in the ICU with pneumonia a different time for 13 or 14 hours, then was in an induced coma with pneumonia the next year, being under for 3 days that time; so I’ve had so many seriously life threatening lung ailments along with my asthma that this I place in that category, while also having sought emergent care decently early and was in the ER the following day, then it got better over the rest of the period). So I hope you feel better.

      I can understand that. I’m to the point of trying to get everything to talk, but sigmastudios only has the ability (that I can find so far) of acting like a microprocessor to program or load the program onto the DSP chips they have. I posted on their forum asking if they have a way to offload the SIMD instructions for their DSPs over to the RPi4 (with the NEON SIMD processing), so I will find out more soon and will continue to play.

      If that doesn’t work, for the more expensive route, I could always use a DAC HAT for the RPi4 to take the signal in, do the FIR filter, and then connect the audio out to the audio in on the JAB5, then just fill out the program on the ADAU1701 doing other tasks since that was done. I could look into inputting the signal over i2s, but let me first do more research.

      Either way, step by step.

    • #14016
      Elliott Designs
      Participant

      @ajc9988

      Sounds like I’m quite well off compared to what you went through, hope you recovered well.

      Something you might want to do if you end up using multiple pi’s for processing is to set a constant clock frequency so that all devices have the same latency, otherwise it fluctuates. As for setting up the communication, it certainly sounds difficult.

      If you can find a DAC hat with aux in; I definitely think that would be a wise choice, unfortunately I couldn’t do that since all the board’s you can get in England were output only. You can always attach a USB dac if you can’t find a hat though.

      I did find this: https://github.com/bmillier/FFT-Convolution-Filter-Uniformly-partitioned which is how I am now able to move over to the teensy since it allows for enough taps compared to the standard 200 limit. Not sure if you came across it, it seems a lot easier of a method and might suite what you are trying to do better too, since you are just trying to do processing rather than using the pi as a source for example. Also seems like a dirt cheap DIY alternative to the miniDSP 2x4hd if you use a few of them. Not high bit rate or sample rate though.

      Also, we should probably have a separate section for this. Maybe “Active Crossovers and Manual room correction”? @123Toid


      Elliott Dyson – Owner of Elliott Designs (the YouTube channel) and 3rd year Mechanical Engineering student. -UK
    • #14017
      AJC
      Participant

      I’ll definitely take a look @ElliottDesigns . As to miniDSP, most don’t know but the original 2×4 was an ADAU1701 whereas the 2x4HD is one of ADI’s Sharc chips. If I was buying a Sharc chip, I’d go for a developer board I saw for around $250 over the miniDSP 2x4HD. Not because that product is bad, just because I could get more bang with a newer Sharc or Sharc+ doing it that way. 

      https://www.hifiberry.com/shop/boards/hifiberry-dac-adc/

      That is an example, which is funny because it seems all they did is switch the headphone jack from output to input. But, that still is priced with the RPi fast approaching paying a fair amount (remember, I already have 4 JAB5s, so the solution is beefing up their capabilities, at least for this project). But, if need be.

      I found out instead of using the SPI connections (which JAB5 uses SPI to setup their master and slave), it was recommended to me to use serial to connect the ADAU1701 to the RPi by ADI. I’m supposing use the i2c then, but shot an email to Wondom to see what they recommend. 

      Best way I can think to run the filter (because he didn’t mention that sigmastudios could program for the RPi to do the FIR filter) would be to route the signal to the RPi and create a little program just to grab it and apply the FIR and shoot it back. Not sure yet. Or find a program already made for RPi that could take the serial in and do a FIR and output it over the same. 

      Just trying to figure the best way to do this.

      And an active crossover and manual room correction section sounds awesome. Main reason it is in this thread is I plan to use this on my speakers. But, would possibly be worth having a devoted section or thread.

    • #14091
      AJC
      Participant

      So, part of my problem was the design for the software to run on the JAB5. I have at least created better working files for it. As such, here are the sigmastudio files to have a working FIR and working IIR based autoEQ. There are built in filters to add time shifts if needed, as well as a baffle step compensation filter and an infrasonic filter. The four knobs are for master volume, gain low, gain mid, and gain high. That way you can adjust volume and volume per driver (I have the knob setup after the crossover for the gain, thereby meaning it will be solely on the basis of that driver, not a frequency range being adjusted with gain).

      Benefits of autoEQ: FIR takes processing more coefficients, which means more processor instructions and resources taken to process the filter. IIR takes fewer resources to do the same task. Seems like a no brainer until the drawbacks. IIR also relies on a feedback loop done to create the filter by using samples to optimize the placement of each tap, etc. It then allows to do the same after creation of the filter (which can be done in REW, then you move those settings over to the sigmastudios software).

      Drawbacks of autoEQ: AutoEQ can lead to having a loss of phase coherency (not just time shifting). Polarity is sometimes called phase, and this can flip the signal by 180 degrees. But, an all-pass filter can be used to shift the phase by 90 degrees. 3xall phase is around 270 degrees for phase. But I have not yet learned if there is a way to fine grain tune the phase of the signal. So the risk on phase is there. The filter has settings for alignment and auto align, but this refers to delay and gain. So, just a reminder, autoEQ could have phase considerations. Be mindful of this when using it, but do not be afraid to use it.

       

      P.S. this time the two programs are tested as outputting noise properly. This is only one channel. To do the other channel, create a second save file, then switch the input from 0 and 4 to 1 and 5. That way you get a left and a right speaker. The 4 and 5 are the inputs for Bluetooth. The 0 and 1 are for the input over wires.

      Edit:

      Still working on the RPi part, but was just getting noise back. I’ve seen reports of that both for connecting the wrong things or that the cables are not shielded and having interference for the jumper cables. That is a WIP. But the above files get it closer to a working solution either way (580 coefficients for FIR; no longer getting the -6 dB I was getting from other working files, so proper signaling unlike before; and working).

      2620-JAB5SigmaStudio.zip

    • #14269
      AJC
      Participant





      Bracing for the subwoofer part of the cabinet. 1269.32 cubic inches of bracing (excluding the volume of the vent shelf shown in the final picture tagged, which is 379.22 cubic inches).

      @123Toid @pieter @Chedwin – tagged you guys because it seems you might be interested in the bracing design.

    • #14271
      pieter
      Participant

      @ajc9988 

      That looks great and I think stable.

    • #14272
      chedwin
      Participant

      Looks great but I have to wonder how you will assemble this? how will you attach the horizontal notched pieces? your trying to link 2 closed rings!

       


      Josh Evans, Professional Live Sound Engineer, High End Commercial AV Install Technician
    • #14273
      AJC
      Participant

      @chedwin – Correct. Which is why there will have to be cutouts. Now, each brace is two pieces of plywood glued together. So, by cutting each of the two pieces wide enough to slide the smaller horizontal ones inside, then gluing the two pieces of plywood together, you can avoid a weakness if you had glued the two pieces of plywood together first and then cut a hole for the smaller ones to fit through. Further, by cutting in two different places, the uncut other piece for that spot acts as a scaffolding when you glue the cut piece back in.

      So that is how I plan to do that part of it.

      This is only a brace for one off type manufacturing, as it is not practical for large scale. Bracing done in the way of GSG’s offerings (see below) would be a better way of doing it. Very similar, but slips together to form the bracing. I just have not given it enough thought to fully do it their way. Also, I do not have a CNC router, otherwise doing a design like theirs would be fairly trivial.


    • #14277
      AJC
      Participant


      So, with it mostly the rest of the way modeled.

       

      Edit:

      Fixed the sub hole in the image and added the port slot on the bottom. Wouldn’t want to forget that!

      For the upper chamber, I did 2 window cross going left to right, then vertical up and down the back, with the vertical ones and the window ones interlocking. 4 braces in the upper chamber may be overkill, but it will also help to control anything coming from the sub below.

       

      Edit: that moment when you realize when you make the cut list that your design calls for 14 sheets of plywood to make 4 speakers, each speaker will likely weigh 220-250 lbs, and that you use about 86% of the 14 sheets (excluding further shaping the braces).

    • #14527
      AJC
      Participant

      So, updating the design. Turns out, if you instead reduce the plywood to 0.5″ from 0.75″, add in 1″ of rockwool in both compartments, you wind up with 1/10 the displacement from the pressure (using 120dB, which is 20 Pascals or 0.0029007548 PSI) versus doing the extraordinary bracing. This also reduces greatly the weight.



      First two are the new design, the third one is the old design.

      There are cutouts to allow the braces to slide into each other, then glue into place, including having cutout blocks to hold the vertical left to right brace in after it is slid into its final placement. I have enough room in the upper chamber to cover the exposed bracing with dacron. I probably will do the same for the bottom. I also kept the insulation cut back enough that it will not interfere with the port flow of air.

      @nixem – we should pick up the topic of material use, etc., here, rather than in DSP.

      This takes 14 sheets of 24x48x1in rockwool insulation to cover both chambers. Approx. cost (for what I can find at the moment) including shipping = $210.

      Edit: The current design, the FEA displacement may actually be higher than reality. When clicking to apply the pressure load to the bare portion of the walls at the bottom by the vent, the program applied that much pressure to the entire wall, instead of the portion of the wall, meaning it is overestimating pressure on the parts of the walls covered by the insulation. So, I might run it again in a bit to try without pressure at those points.

      But, the last analysis took up nearly 500GB for the tower alone. So, definitely a space hog. lol.

    • #14528
      nixem
      Participant

      @ajc9988 

      Great work! 👍

    • #14530
      kanaaudio
      Participant

      @ajc9988 Looks amazing!

    • #14531
      123toid
      Keymaster
    • #14532
      AJC
      Participant

      @kanaaudio @123Toid @Nixem – also, to clarify, the $200 in insulation is to build 4 enclosures, not a single enclosure (around $120 for the insulation, then $80 to have it shipped), which then is $50 per. I used a constrained layer with cheap 1/4″ gym rubber mats for the vent shelf and the separator between the upper and lower chambers (partly with the vent to reduce resonance from the air passing through it since you cannot use insulation and need to avoid insulation being too close to the vent; whereas for the separating chambers is to further reduce vibrations not handled by the insulation from either chamber). I figure that would be around $34 to do all 4 with rubber. For Dacron, that could be $40-80, give or take, depending on the quality of polyester/dacron/PET used and local availability and shipping. So I estimate around $300 for all insulation and damping materials for all 4 speakers.

      But, this shows why many speaker companies use 3/8″ to 1/2″ materials for their boxes, because so long as there is enough bracing (which there is less benefit to bracing from my studies than seen on 3/4″, although for constrained layer damping, I saw more benefit from 1/2″ on both sides with a constrained layer than 3/4″ on both sides or 3/8″ on both sides) and there is insulation, you can damp more than thicker material and bracing alone. Constrained layer damping is in between the effectiveness of bracing alone and insulation. As such, insulation should be leaned on the heaviest in designs, with CLD used judiciously in cases where it would do better than insulation at damping or when insulation cannot be used for one reason or another (or when using it in conjunction with insulation makes sense). Using bracing alone can help (for example, non-braced 2’x4’x0.5″ had 0.06in displacement with no bracing, whereas the 2’x4’x0.75″ had 0.02″ displacement without bracing; a simple cross (meaning one vertical brace intersecting a horizontal brace forming a cross shape, except centered) brought the 0.5″ sample down to around 0.016″ displacement, but the 0.75″ sample was brought down with the same type of brace to the 0.006″ displacement. Doubling up the thickness of the brace by gluing two braces together does add more to 0.75″ samples (so 1.5″) than doing so with the 0.5″ samples (1″). As such, the thinner the material being worked with, you have diminishing returns on trying to increase bracing thickness to compensate.

      The insulation also damped the clear ringing (looking like a rising sun shape) on the original design. This would likely have shown up in final testing after built, with a wider (lower) Q effect in that range, rather than the much higher Q, very short duration modes seen in the recent design with insulation. That means it will be much less audible, if at all, in the final design. This almost makes me wonder if an FEA was done on speakers identified to have a port resonance or box resonance (sometimes called ringing), that then the ringing could be identified by modeling speakers for reviews (although the cost and time to run FEAs on items for review does not justify any ROI of doing the analysis, so interesting thought that we could possibly visualize the data that currently is hard to show in testing data for frequency response through looking at displacement vs frequency in FEA, but that any cost to do so means we won’t see it outside of published studies, potentially).

      I just find the data interesting to understand why speaker manufacturers do what they do.

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