Introduction

Hello, i'm Erwin. I joined here to discuss about the various DIY parts i'm also using in my system. At the moment i am working on a pre-amp and converting my system to balanced.

my DIY parts are:
  • Pre-amp
  • LM3875 'gainclone' amps for the horns
  • Speakers, except for the BD-Design horns. They are sort of like ripole bass units, but ported for increased bandwith and efficiency.
  • Various cabling

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Erwin

2SK183 SIT-WICH

Hello DIY friends,

As you may or may not be aware. I recently acquired a pair of Tokin 2SK183 from Ebay.

I have searched everything that I can get my hands on regarding these transistors from this site or on the web.

As far as I am aware the only other design using the 2SK183 I have seen on the web is from this website (http://www.ne.jp/asahi/evo/amp/2sk183cspp/rep.htm).

I have no plans to follow that design, except to copy how the designer managed to build a contraption for the Source and the Drain, as I posted here and here.

He used a thick Square piece of Aluminum, I plan on using the L-brackets instead. I am rather worried a bit about heat dissipation, for obvious reasons. If anything I may have to create a proper babysitter for this amp.

The plan is to create a SIT Sandwich. The Drain and the source are attached to 1mm copper plates, which are then attached to an Aluminum L-Bracket I purchased from Amazon. The Aluminum brackets is then attached to the heatsinks.

The hope is after I finish my SIT sandwich, then I can ship it to a member and he can do some curve tracing for me.

After that, I need to choose a design. Off the top of my head, I prefer to keep this project simple. Either a lightbulb amp (common Source or Drain) or a choke-loaded follower. I have a set of Warbler PCB's that I got from @schultzsch , I am unsure if that will work/be problematic.

Of course, even before I get to that point, I need to have the SITs tested and identify operating points, bias, etc.

I was going to purchase the aluminum and copper plates from onlinemetals.com but right before I was about to purchase, Onlinemetals decided to increase the price of the shipping, which to me was not worth it. So I got the L-Brackets, Copper Plates, and Keratherm thermal pads straight from Bezos Bookstore.


Is there a high chance that I will make a fool out of myself? :nod:

Are the 2SK180 and THF51S easier to work with and have already proven designs on this site? :nod:

:rofl: wish me luck in advance. The only thing missing are tin snips and some long bolts and nuts to secure the SIT-Sandwich in place.

Some porn below....

Best,
Jose

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Data only cable USB-A > USB-C

Hello Gents,

Made a silver wired USB cable for USB Dac connected to my dietpi music server. Unfortunatly my current dac, a smsl sanskrit MK-II, doesn't support a data only cable.
The 3 silver wires (D+, D- and GND) are 0.5 mm² with a ptfe insulation which makes to diameter 1.3mm. Alu foil and a metal braided shield for the protection. The number of twists on 35 cm cable is about 20 turns. It is said the cable must be 90 ohm impedance but I can't measure that as I don't have to tools.......yet.
Whether or not you can "hear" any difference I honestly doubt it but It sure is fun to build one just for the sake of it.🙂
I'm looking at the smsl dl200 dac which has its own AC power supply but whether or not he USB input of the dac expects 5volt I'm not sure...we'll see...

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Criton need help

As a newbie, I started with Criton 2TDX since I got their plans, and their LDW7. I played with Vituixcad and Xsim to end up with this;
1730949948537.png
But, when I tried with the crossover, there is a lack of loudness for both (tweeter and woofers), I could only get 10% of DB expected. I made tests, bypassing the capacitor for the tweeter and the woofers, it's sounds better in terms of DB.
My question; where should I focus modifications or research to find the trouble?
Tkx

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NAD 2400 problem - need help

I am looking to obtain a schematic and possibly some direction as to fix my NAD 2400 amp.

Here's what happened. I loaned my pre-amp and amp to a friend that was wiring his house for in-wall speakers. We had three sets of speakers (8ohm) hooked up in parallel. We were using impedance mathing volume controls (at x4 resistance setting). My friend disconnected one set of speakers (After the volume control) but did not disconnect the wiring to the volume control. He turned on the system to listen to music on the remaining sets. Music played for about two minutes and then the amp blew the main fuse (5mF 125V 7A). After consulting with an electrical engineer at work, he told me that it was a bad thing to leave the one volume control hooked up (the one without speakers attached). The volume controls were Phoenix Gold VMT100

I replaced the main fuse, plugged it in, and nearly instantly blew the fuse. I tried again and blew the second fuse. Obviously I have a problem.

Does anyone have a hypothesis as what is wrong or what is the next diagnosis step? I am thinking it is the transformer. Can someone provide me with a schematic for a 2400?

ADS M12/90 Schematic? Anyone?

Does anyone by chance have the schematic for ADS M12/90 speakers?

I'm picking up a pair of M12s this weekend that had the crossovers removed and I've been unsuccessful in finding a schematic for the upgraded version of the M12/90. As I understand, the woofs were crossed at 200 Hz with a LR 24dB/octave slope using a laminated inductor. The tweets were crossed at 2kHz also employing a LR 24dB/octave slope. Knowing the part number or equivalent for the inductor would be very helpful. The M12 saturated at high levels and the upgrade supposedly fixed the issue. I'd like to avoid choosing the wrong inductor for that reason.

For Sale Klipsch project parts

Hi there, I'm selling some parts that I will never use. more to come after this. The following are available. shipping will be added to the posted price.

Crites CW1526C woofers pair, unused, excellent condition. $300.
B&C DE120=8 compression driver pair, with Crites treble horn. lightly used, excellent condition. $180.
Crites 2.5 mH iron core inductors pair, $40.
Crites 3654 autoformer pair, excellent condition. $100.
ALK cornscala-wall universal crossover pair in excellent condition. $375.

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Kicker ZXS1500.1 power supply

Hi everyone!
Im working on a kicker zxs1500.1 it had blown power supply. I change 3843ic, 2045 and 3 new irfb4110. The ps turns on and raise 85vdc. But it has a noise at the same time the red power led flash, only one time every 3 seconds . The dc bus voltage is stady 85 and the input the same 12.5 volts.
No shorted outputs.
If i conect a speaker it hits at the same time the red light flash.

Any ideas??

Thanks!

I found details for the Dual PA130 XKi cabinet

deep within email

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Reactions: xrk971

Hello

Hi, I'm Ciro from Argentina. I'm currently in 6th grade studying electronichs in a technical high school (Instituto Politécnico)
Last year I was sick for 2 months, and while staying at home for that time, I developed a profound interest in audio electronics, I began making guitar pedals and now I'm starting to mess with amplifiers.
Aside from electronics, my other 2 hobbys are music, although I'm not quite good yet, and speedsolving rubik cubes, in which I perform a bit better.
My favourite music genre is Progressive Rock and I love bands like King Crimson, Emerson Lake and Palmer, Yes, Pink Floyd and Rush.
I hope one day I'm able to be a good bass player and with an entire rig made by myself.
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Hello intro

Hi All,

I am a long time music and sound aficionado. I am handy with somethings. I have restored vintage turntables, built my own step-up transformers, but am looking into upping my technical knowhow. In particular I am looking to build my own speaker crossovers a la Peter Riggle and Werner Jagusch. Looking forward to tapping into the collective brain trust here on diyAudio.

Ozujiro

Mailbag Monday: Midrange Taps on a Unity Horn

Someone emailed me and asked about the various midrange taps I've used in Unity horns.

I've used slots, I've used holes, I've used intricately designed phase plugs, etc

I've generally found that the formula used by Danley in the SH50 Synergy horns works well. They're located 3.5" inches (89mm) from the throat and they're 3/4" in diameters (19mm.)

There are two midrange taps per midrange, and each midrange is 5" in diameter.

These dimensions will get you "in the ballpark" and it's hard to go wrong with them.

If you want to obsess about the geometry:

As the beamwidth increases from 50 degrees, you'll need the move the midranges closer to the compression driver. This is because the main variable in the midrange taps isn't the distance from the throat, it's the distance from the left midrange to the right midrange, and the bottom midrange to the top midrange. That's what determines how high they'll play.

This is basic midrange-tweeter-midrange theory, the further apart the mids are, the more you get off-axis nulls.

The exit of the midrange taps causes a discontinuity in the horn. It's up to you to decide whether you want ONE relatively large hole in the side of the waveguide, or whether you'd prefer a long and narrow slot.

Both options have their merits:

1) a hole only causes a discontinuity at certain angles, but the hole is relatively large.

2) a slot causes discontinuities at nearly all angles, but because the slot is narrow, the wavefront barely "sees" it.

Hello

Been messing around and building various loudspeaker designs for a while., I started this journey before the internet existed using parts from old and damaged speakers and then Radio Shack parts which was at the time the only place to find speaker parts. I am also a wood worker and a Master Electrician.

I also enjoy learning and writing code in HTML, CSS, Python, SQL,
My first webpage was very badly written and had crude HTML which was at LoudSpeakerBuilder.ca which existed online from about 1998 - 2011 when Shaw shut down my free web hosting. Recently I re-wrote the site, fixed the .html and CSS and re-posted it on a domain I own at https://diyaudiospeakers.5150.ca.

Back in 2001 Google had the site at #1for search results on the thiele -small parameters, crossover networks, and loud speaker enclosures.

Parallel SE 6V6 amplifier questions / design

A new project: a parallel 4x 6V6 with triode, pentode, and UL switch. Plus, adjustable global negative feedback from 0 to 9db. back to the cathode of driver tube.

Each 6v6 will be individually cathode biased and have its own coupling cap from the driver tube.

OPT will be a 1.2K. Edcor, Sowter, Lundhall or equivalent.

Power supply... no idea. Just a low noise one.

I am trying to figure out driver tube. Pentode, Triode? has to have enough gain for 9db of feedback plus decent input sensitivity and be able to drive a 100K load and the input capacitance of 4 6V6's. Thoughts are a EL84 in pentode mode. 4-8K resistive load. 150v screen, 300v V+, 150v plate, 37mA plate current.

Thoughts? Or should I go with a triode driver?

probably 6W triode, 16W UL, 20W Pentode .... maybe. lol.

All of this is still in very early design stages. But I am pretty fixed on using the parallel 6V6s.

Thank you sooo much!

Hifonics BXX4000.1D no oscillation / not in protect

Hi all, I've got a Hifonics BXX4000.1D that (mostly) has no output, no oscillation on the output section.

After it's been off for a long time, it'll sometimes work for a little bit, and seems to produce clean output, R/R oscillation waveforms look clean, then the output sections stop oscillating.

Power supply remains on, relays are closed, and it does not go to protect. Voltage going into the output inductors is zero (none of the output devices conducting or leaking). Cycling the remote at this point gives what looks like a normal startup, power supplies start properly but no attempt at all at output oscillation, not so much as a single gate pulse anywhere on the outputs.

Nothing is getting inordinately hot according to my thermal cam, and the +/- 15V and +/-5V supplies going into the modulator board look OK, even when it's not oscillating. The ground-coupling circuit is good.

Anyone seen this and figured out what causes it? Thanks!

Seeking help with reconfiguring jFET differential input for new jFET Parameters

Hello all,

I'm working on rebuilding a Kenwood KA-9100 integrated amp and the need has come to replace the input differential pair JFETS (Qi 21 and 23) in the phono section. The original were 2sk68a L rank parts. The replacement parts I received are N rank parts with different Vgs on / off and IDSS parameters. I thought it would still be possible to use these new parts with some reconfiguration of the source and drain resistors in the circuit. I will present my math and reasoning below and I'm hoping that someone more knowledgeable than I would be willing to check my math and reasoning.

The original Jfet had the parameters Vgs OFF: - .45V Vgs ON: 0.22V IDSS: 2.2 mA

Current through the drain resistors (RI 105, 107) is 16.5V / 22K = 0.75 mA each
voltage drop across source resistor (Ri 109) is 1.5 mA * 10K = 15 V

Current through Ri 113 is 14.1 V / 3.3k =4.27 mA

Current through Ri 111 is approximately 4.27 mA - 1.5 mA = 2.77 mA

The new JFET parameters are Vgs OFF: -.8V Vgs ON: -0.1V IDSS 5.5 mA

For the new circuit, I will maintain the drain voltage of 11.5 V for proper biasing of the downstream stage. I am planning on using a source voltage of 350 mV (about 50% of the difference between Vgs on and off). I am planning to use a drain current of 2.5 mA (nominally around 50% of the IDSS).

New drain resistor values for RI 105 and 107 is 16.5V / 2.5 mA = 6.6 Kohm each

For calculating the source resistor i need to take into account the increased voltage drop through Ri 113 due to the increased current as this will lower the available voltage at the junction of Ri 109, 111, and 113. I used a bit of a guess and check method here instead of setting up a system of linear equations because I am not fresh out of school at this point. I know I need 5 mA through the source resistor Ri 109 so I know that the current through Ri 113 must be greater than that to allow current to flow to ground through Ri 113 (at least I think that is the correct assumption). In this case I assumed a current of 6.5 mA through Ri 113

Thus the voltage at the Ri 109, 111, 113 junction is 6.5 mA * 3.3k = 21.45 V, 29V-21.45 = 7.55 V

Thus the new source resistor voltage drop is 7.55 V + .35V (source voltage) = 7.9 V and thus the new source resistor value is 7.9 V / 5 mA = 1580 ohms

I'm hoping that the voltage divider with ri 111 isn't critical for some reason that Im unaware of and is ok with the reduced current through it compared to the original
design. The math is simple enough I hope but I really need help with checking the assumptions I used. Hopefully the new values I came up with would work better than leaving the stock values!

1730350006311.png

Threshold CAS 1 - Trying to undo a previous “repair”

Happy new years all!

I received this amplifier years ago as a gift and have been enjoying it with a pair of KEF LS50, until the left rca input failed…

Once I opened it up, it was apparent that there were some previous repairs done. There was a hand-written date code on the transformers (98’-99’) and the capacitors were not original (under-spec’d). Now I am on a mission to restore it.

I plan on replacing tantalums with electrolytics and new filter caps. I assumed the 5,400uf caps should be 15,000uf.

Would a Kemet ALS31A153KE063 work?(Digikey link) other suggestions?

Many thanks!

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For Sale UK Based - Amp Camp Amp v1.6 kit, PCB Components and 2 enclosures

I'm selling my ACA 1.6 kit, with PCBs, components and two enclosures. Based in the UK, this is pretty much unopened and in great condition, bought it but just haven't had time to construct it and trying to focus on other projects.

It's currently on eBay here https://www.ebay.co.uk/itm/305890600032 I'm open to offers.

Damping Equivalent to Dacron 9oz

The speaker design I am building calls for a “Dacron Type 9oz” dampening material, which looks to be about 15mm thick on the drawing.

Can anyone advise me what the equivalent would be in the USA for this?

From my Googling and calculations, this is a medium loft Dacron batting with a 1 oz per sq ft density. I.e. 9oz = 305 gsm = 1 oz/sqft.

Is this the correct equivalent to what is spec’d?

For Sale Acoustat 2+2 spandex socks

Hello Acoustat 2+2 owners! If anyone who owns 2+2's or Spectra 44's is looking

for bone or ivory socks, I have 2 brand new spandex socks that have been sewn

and ready to slide over your frames. These also have a 1 inch reinforced hole

to allow the panel wires to fit thru. They measure 108 inches which gives you

about 7 inches extra on each end to make it easy to staple then cut away the

excess. Asking $100 plus shipping. Please add 3.5% if using PayPal unless

using the F&F option.

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Why not IIR filters + a global phase linearization by FIR

Question came up in another thread....thought it best to reply in a new thread....
A big reason that I think NOT to use global FIR on top an existing speaker setup, be it passive or active,
is to avoid pre-ring.

Out-right pre-ring is obvious in electrical IIR xovers summed together and then phase-linearized globally in a subsequent processing stage .

Here are two Linkwitz-Riley 24dB/oct IIR xovers at 200Hz and 1300Hz that are summed together to give mag and phase traces as shown, and are as expected.

24dB IIR summed mag and phase.JPG

Here is after global phase correction is applied, and is as expected. (slight phase drift at 20kHz...not being picky here for this simple demo)
24dB IIR summed mag and phase w global FIR.JPG

Here's the step response to look at. Out-right pre-ring is not pretty.
24dB IIR summed mag and phase w global FIR step response.JPG


Ok, the same two Linkwitz-Riley 24dB/oct xovers at 200Hz and 1300Hz are summed together, but now are linear phase.
Flat mag and phase as expected.
24dB Linear phase summed mag and phase.JPG

With an essentially a step response free of pre-ring.
24dB Linear phase summed mag and phase step response.JPG


Ok, i hope that makes a pretty good case against global FIR phase correction. There's more reasons to avoid global, but that's the easiest one to show some quick meas.

Converting vintage 3-way speakers into an active speaker with Bluetooth

I have a pair of vintage speakers, and I would like to convert them into active Bluetooth speakers. I know that it may be cheaper just to buy a pair of Bluetooth active speakers, but I would like to modify this pair of oldies.

Questions:

Which plate amp as these old acoustic suspense speakers need a bit of power to perform? Most high power plate amp that I found online are for subwoofers, what can I do?

Adding bluetooth, how to do that?

Thanks for helping.

Simaudio’s Power Amp MOON Neo Series "330A" (Advanced Renaissance Circuitry) - Schematic wanted

Simplified schematic is to find under
https://www.studio-22.com/pdf/MOON_330A_Lit.pdf
OWNER's Manual under
https://simaudio.com/wp-content/uploads/2018/10/330A_Manual_EN.pdf
and a review under
https://www.moremusic.nl/reviews/moon/MOON-Neo-330A-SoundStageHifi.pdf

who can upload the schematic resp. a service manual ?

Thank you very much.

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Hello from Adam

Joining the forum. Interested in learning more about how to build my own gear. I've built my own speaker cabinets, made a couple of amplifier kits, dynakit ST-35 and bottlehead SEX 3. My gear at the moment is a Wand 14.4 table and 10.3 masterlite arm with a soundsmith carmen and mmp3 phono into a rogers 65v2 amp into lowther designed accousta 90 cabinets and PMA2 drivers.

Burson discrete opamps

Numerous videos have been posted on YouTube where people describe the impact of discreete opamps in there D-amps like Fosi and Topping etc.
but in the context of Diy amp boards I could not find any reviews.
Knowing first hand what a discreete opamp can do in the output stages of two of my DACs, I was excited when Burson Audio approached me to review there V7-C and V7-V in my Class D amp boards.

Many would never consider putting about $170 worth of opamps into a pair of $120 diy Chinese amp modules, but I find it intriguing to see where this can get me sonic wise.
In the next couple of weeks I will put the Burson opamps into three different Class D amp boards and report my personal impressions.

1.TPA 3255
2. Infineon Merus MA12070
3. Infineon Merus MA5332MS
As the latter two got smd based opamps I will supply them with diy adapters.

All this will take a few weeks or even months depending on my work schedule.
In the spirit of diy I will later on also run the discreete opamps with external power as I think that the onboard regulators on most d-amp boards are of poor quality.
My impressions describe the difference between the original opamps and the Burson Audio V7 and are of course subjective and relate to the context of my system and room.
As a source I am using the Wiim Ultra with it's internal dac to keep things simple. The speakers I choose for these reviews are the Quad 21L as they are able to throw an impressive three dimensional sound field if the recording is done well. A deep soundstage is IMO exactly what most stock d-amps are lacking compared to good class A or AB amps.

Now on to our first unit the Zero Zone TPA 3255 boards. As we diyers know these Chinese amp boards are seldom equipped with genuine passive or even active components. I made it my habit to change all caps including coupling caps but in the case of the Zero Zone boards I cept them stock for the review.

From the first note on after the change to the Burson Audio V7C there was no doubt that the boards reached a new level of performance.
The bass has more authority and control. Nuances come through that I previously did not hear as clearly defined. Also bass drum hits while having more impact are not as overpowering as with the stock opamps.
Instruments in general sound more real and live like.
They are also more carved out and there is more room between them. For the first time I hear adequate depth information on a TPA3255 based board. Not as deep as with my AKSA or EL84 tube amp but to the point that I could hear different planes on well recorded material. Another positive change is how much better cymbals sound. No it's not like you are there but much closer to the real thing than with the stock opamps.
Now to the last observation that in my book is very important: Dynamics. Not being able to play my horn setup at the moment I really appreciate any gains in dynamics. Boy this is the most obvious change to the stock opamps. Everything comes alive big time. Sudden orchestral fortissimos are a joy and mabe this ability to effortlessly deliver dynamic swings is the secret why cymbals sound more real now.
In the past I have done lots of things to get class d amps to sound more real: Transformer coupling, Skipping the whole opamp stage, better coupling caps ect. These Burson discreete opamps transformed my TPA3255 amps like nothing before.
It's to bad I had to do this review in single ended mode as that's the only mode the Wiim Ultras internal dac supports. All the d amps I tested perform much better when driven fully balanced. Most Chinese d amp boards are only pseudo balanced designs and that's why I normally skipped the whole opamp stage.
Fortunately the Zero Zone boards are fully balanced if you bridge two pads.
I will report back on how the Zero Zone boards perform fully balanced once I get my external r2r dac board converted to differencial operation.
Also I still have not tested the V7 vivid in the Zero Zone boards but will report the difference between them and the V7 classic later.
1000202689.jpg


Take care,
Klaus

First subwoofer design

Hello everyone,

A collegue and I are planning to make ourselves subwoofers. I did try to find all the information required on the internet including a lot of information on this forum.
As I think will be the case for most people, I tried to balance the size, cost, volume and lowest frequency to have something we are happy with.

I would love to hear your input on the design. While I do like the idea of it being my own design, if someone has a suggestion for a design that performs better for a similar price and size, I would be open to it.

For the driver I went with the SB Acoustics SB34SWPL76-4
For the amplifier I plan to use the Dayton Audio SPA300-D

Both of us use a receiver with significant EQ options for the subwoofer.

Targets for the design:
Comfortably plays up to a 105 dB or more
Frequency deep in the 20's (preferably sub 25Hz at -6dB)
Total budget max 750 euro per subwoofer. (which leaves about 600 for driver and amplifier)
Play up to 100 Hz for crossover of smaller speakers.

Internal volume is about 40L excluding driver and port.
Port is devided into two by a brace, making it 2x 25x150mm and 710 mm long

I have included a couple of screenshots from WinISD and the drawing of the enclosure.

In addition to apreciating general feedback, there are a few questions/concerns I still have.

At the maximum 300W power, the volume is much higher than required, but the port velocity also goes up 24 m/s. I am most certain this would create a lot of port noise. However, I think at 200W I already have quite the volume for my space. At that power level velocity caps out at 20 m/s. I have seen the guideline 18 m/s max. Is this expected to be a problem if the port end is rounder off? Obviously I only care about port noise in real world scenarios and not test tones.

I found a lot of people finding that for these long port lengths, the required length for the tuning is actually shorter than calculated. I am hoping for this effect, as all desired parameters will improve in this case. I would be more than happy with the calculated results.

At work we have large CNC mill suitable for MDF, so we plan on milling out the shapes of the plates in such a way that they fit perfectly. (See the picture of a side panel)
Does anyone have any experience with this? At work the people that actually use this machine thought it was a good idea, but we don't typically build speakers. They did know the tolerances required for nice fits in MDF, so I am assuming that would work.
All the rounded edges will be done using a rounding bit on a handheld router.

Is my placing of bracing correct? I have about 20 mm distance from the rear of the driver for the cross brace. I feel like the brace in the port is required as one side doubles as internal volume. Is that correct, or can I leave that out?

I see a lot of debate about panel thickness. Lower weight is higher eigen frequency, but thicker is stiffer and higher eigen frequency. I am doubting between 16 and 22 mm MDF, which are both easily available to us. Does it make a significant difference?

Would a high pass filter be required for this subwoofer? I can't imagine the driver being damaged with real world use due to large cone excursion, but maybe someone can tell me otherwise. WinISD calculates Xmax at 19.5 Hz at 300W. I can't find the Xlim of this driver.


Thank you in advance.

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Mark Levinson 585 Noise

Hi,
I'm owner of amplifier Mark Levinson 585.
From the very beginning, it used to happen that when switching on from standby, noise appeared on the left channel. After turning it off/on again the noise would disappear. After a while turning off/on doesn't help and the noise is constantly present.
I'll be grateful for any help, and if any aditional info need please ask.
In attachment is recorded noise on left channel

How to repair damaged heatshrink tube

Sometimes I get audio or power equipment with damaged wiring or wiring with a damaged heat shrink tube around a wire or a socket.
The obvious solution is to put in new wiring.

However sometimes this is very impractical or even impossible. Is there any product that can be used to repair a damaged heat shrink tube?
For large coax cables I have used self-vulcanizing rubber tape in the past, so perhaps something like this exists for thin wires?

Regards, Gerrit

Looking to upgrade from SEAS 27TAC/GB

HiI am currently using this Seas driver in a 12 inch three way system using the SB34NRXL75-08 and Arum Cantus AC130F1, XO at 350hz and 3200 second order. Its a pretty high end system using CJ electronics and Holo dac, etc. I want to play. Im thinking of trying a new tweeter. (I guess if other designers are not enamored with the mid I could do that instead but I wanted to start with the tweeter). My budget is say $500 for the pair. I'd like it to drop in to my current flange opening (104mm or 4.1 inch). Thanks

Switching Alsa Loopback Timer Source to Playback HW Device

The alsa loopback software device is by default clocked by a system timer. Chaining with any hardware soundcard thus introduces two clock domains which must be aligned somehow. Fortunately the loopback offers alsa controls for fine-tuning the internal software clock which is conveniently used in alsaloop or, more importantly, in CamillaDSP. The bridging SW must keep track of clock differences of devices on both sides, and using some controller (basically a PID regulator) fine tune the loopback clock to keep the production/consumption rates on average equal. But this controller has delays, can overshoot, which is fixed by keeping the bridge latency sufficiently long - undesired effect.

I noticed a feature added to the loopback kernel code in 2019 https://patchwork.kernel.org/projec...1120115856.4125-1-andrew_gabbasov@mentor.com/ . The loopback timer can be switched to a virtual timer provided by any running alsa device. Technically it's quite simple - when the master device fires end of its period, the loopback runs its end-of-period code too. As a result the loopback and the master device run absolutely synchronously.

The timer is either configured at module load (param timer_source), or dynamically via /proc/asound/LoopbackX/timer_source RW file. The configured/changed timer is applied when opening the loopback device, that's important to keep in mind.

I tested the feature on a chain with USB gadget as a playback device, with CamillaDSP as the bridge (no rate adjust, empty pipeline). The USB gadget conveniently allows to tweak its samplerate in a large frequency range with its alsa control while playing (works via USB async feedback messages to the host).

I switched the timer source to the USB gadget playback device on the USB host (empty string means the system timer):

Code:
echo 'hw:Gadget' > /proc/asound/Loopback/timer_source

CDSP would not start processing, due to a simple deadlock - the playback device waits for chunks received from the capture side, but the capture device is stalled (as the playback device is not running) and cannot produce any chunk to start with. This was simple to solve by sending two kickstart chunks from the capture side with zero samples before the capture loop (one chunk did not hit the configured playback device buffer threshold for start up).


Diff:
--- a/src/alsadevice.rs    (revision 3981740d8a4f38f00a44de3abee77ffd4342b546)
+++ b/src/alsadevice.rs    (date 1705754133966)
@@ -682,6 +682,14 @@
         peak: vec![0.0; params.channels],
     };
     let mut channel_mask = vec![true; params.channels];
+
+    if !send_zero_chunk(&channels, &params) {
+        return;
+    }
+    if !send_zero_chunk(&channels, &params) {
+        return;
+    }
+
     loop {
         match channels.command.try_recv() {
             Ok(CommandMessage::Exit) => {
@@ -871,6 +879,19 @@
     params.capture_status.write().state = ProcessingState::Inactive;
 }
 
+fn send_zero_chunk(channels: &CaptureChannels, params: &CaptureParams) -> bool {
+    let waveforms = vec![vec![0.0; params.chunksize]; params.channels];
+    let chunk = AudioChunk::new(waveforms, 0.0, 0.0, params.chunksize, params.chunksize);
+    let msg = AudioMessage::Audio(chunk);
+    if channels.audio.send(msg).is_err() {
+        info!("Processing thread has already stopped.");
+        return false;
+    } else {
+        info!("Sent kickstart zeros to playback.");
+    }
+    true
+}
+
 fn update_avail_min(
     pcmdevice: &PCM,
     frames: Frames,

With this simple change CDSP starts running fine, and no matter how much I torture the playback device rate, loopback runs synchronously and CDSP keeps the playback buffer levels basically constant:

Code:
amixer -c UAC2Gadget cset numid=1 1000000
Momentary freq = 48000 Hz (0x6.0000)

2024-01-20 14:44:05.865793 DEBUG [src/alsadevice.rs:573] PB: buffer level: 1584.0, signal rms: Some(HistoryRecord { time: Instant { tv_sec: 14944, tv_nsec: 337173553 }, values: [0.0, 0.0] })
2024-01-20 14:44:08.873720 DEBUG [src/alsadevice.rs:573] PB: buffer level: 1584.0, signal rms: Some(HistoryRecord { time: Instant { tv_sec: 14947, tv_nsec: 345099643 }, values: [0.0, 0.0] })
2024-01-20 14:44:11.881425 DEBUG [src/alsadevice.rs:573] PB: buffer level: 1584.0, signal rms: Some(HistoryRecord { time: Instant { tv_sec: 14950, tv_nsec: 352803979 }, values: [0.0, 0.0] })


amixer -c UAC2Gadget cset numid=1 875000
Momentary freq = 42000 Hz (0x5.4000)

2024-01-20 14:44:35.004469 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 42000.428535622406 Hz
2024-01-20 14:44:35.956677 DEBUG [src/alsadevice.rs:573] PB: buffer level: 1507.9, signal rms: Some(HistoryRecord { time: Instant { tv_sec: 14974, tv_nsec: 428059201 }, values: [0.0, 0.0] })
2024-01-20 14:44:36.028451 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 42000.5191411824 Hz
2024-01-20 14:44:37.052470 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 41999.34199663706 Hz
2024-01-20 14:44:38.076479 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 41999.594277356846 Hz
2024-01-20 14:44:38.979670 DEBUG [src/alsadevice.rs:573] PB: buffer level: 1508.0, signal rms: Some(HistoryRecord { time: Instant { tv_sec: 14977, tv_nsec: 451050650 }, values: [0.0, 0.0] })
2024-01-20 14:44:39.100444 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 42001.36209104719 Hz
2024-01-20 14:44:40.124472 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 41999.031151256 Hz
2024-01-20 14:44:41.148455 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 42000.761181763686 Hz
2024-01-20 14:44:42.002688 DEBUG [src/alsadevice.rs:573] PB: buffer level: 1507.9, signal rms: Some(HistoryRecord { time: Instant { tv_sec: 14980, tv_nsec: 474066920 }, values: [0.0, 0.0] })
2024-01-20 14:44:42.172459 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 41999.57122703359 Hz
2024-01-20 14:44:43.196445 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 42000.87151808399 Hz
2024-01-20 14:44:44.220464 WARN [src/alsadevice.rs:787] sample rate change detected, last rate was 41999.17634037181 Hz


amixer -c UAC2Gadget cset numid=1 1005000
Momentary freq = 48240 Hz (0x6.07ae)

2024-01-20 14:45:48.449535 DEBUG [src/alsadevice.rs:573] PB: buffer level: 1579.6, signal rms: Some(HistoryRecord { time: Instant { tv_sec: 15046, tv_nsec: 920911704 }, values: [0.0, 0.0] })
2024-01-20 14:45:51.463561 DEBUG [src/alsadevice.rs:573] PB: buffer level: 1579.0, signal rms: Some(HistoryRecord { time: Instant { tv_sec: 15049, tv_nsec: 934935325 }, values: [0.0, 0.0] })
2024-01-20 14:45:54.477358 DEBUG [src/alsadevice.rs:573] PB: buffer level: 1578.8, signal rms: Some(HistoryRecord { time: Instant { tv_sec: 15052, tv_nsec: 948735932 }, values: [0.0, 0.0] })

IMO this is a rather useful loopback feature which may allow using lower latencies (shorter chunksize) for CDSP bridges/simple DSP involving alsa loopback.

Cons:
  • The loopback device does not consume samples from the player until the CDSP bridge starts running (i.e until the loopback timer source starts running).
  • The /proc/asound/Loopback/timer_source is writable only by root.
  • The Loopback capture and playback devices must use the same period size (already used by CDSP)

Coax build V2!

Hi everyone, I recently had a thread going around my struggles with the Radian 5210 and trying to use it to make a pair of high sensitivity speakers for use with a flea watt tube amp. As I posted in that thread, I've thrown in the towel on the Radians, and am now looking at the Beyma 12XA30ND, a 12" coax. After looking at the parameters, and modeling some enclosures in WinISD, it looks like a good candidate in an 80L box tuned to 45hz, more or less. I'm still playing with different alignments, but at least I'm in the ballpark now. I'm still considering another driver, the Faital 12HX240, but haven't modeled it yet. I have some questions. Oh boy, I'm nothing BUT questions.

First question is which alignment people usually use in enclosure modeling? I'm using SBB4 right now, but not sure what the differences are, even after reading the chapter on vented enclosures in the LDC four times. This stuff only really makes sense I think after you've designed a bunch of stuff and heard the differences, to me it's not very intuitive at this point. What is the relationship between choosing a modeling alignment and designing the crossover down the line?

Second question is really basic. Since all I've got right now is factory numbers to work with, I'll build an 80L box and do some measurements. I assume the best way to get accurate parameters is in a suitable enclosure, but is this true? I'm talking about both FR stuff in REW, and T/S parameters and impedance curves using my Dayton audio DATS device.

Thanks in advance!
Bryan

Power Distributor DIY

Hi Everyone,

I'm about making my own power distributor for audio and video, I hope someone can help me in some areas.
I need know what best option about connections, starwire or daisy chain (loop), to wire six outlets?

The wires I'm thinking: Neotech SOCT-12AWG Copper. Sockets: Furutech FI-E30 Schuko sockets. for use 240 volt.

1- If I want use Furutech FI-09 IEC as starting point, the six wires will not fit. As each terminal of FI-09 accept maximum gauge only 10mm. There is a solution, attach all wires to Furutech spade 209, which will fit inside FI-09, but I don't want use spades because they degrade sound quality.

2- Alternative, not use IEC connectors, connect power cable strands strands directly with solid core, twisting all wires together, then tighten by metal crimp, or solder.
But not too comfortable about this even if can archivable, because not commonly used, and not sure about safety and reliability after long run.

So, I prefer better idea.

Many thanks to all!

Power Amp burning smell

I have a Counterpoint solid state power amplifier which I have been using since a year with my B&W 805 matrix speakers. This morning I changed my preamp to a Schiit Valhalla 2 Tube preamp and everything played well as usual. When I turned on my system this evening, both speakers were whining like a burglar alarm and I immediately switched off the power amp to be safe. There was a burning smell from the power amp for sure like some thing went wrong and the power amp has a 6 Amp 3Ag fuse for each channel next to the banana terminals. Upon observing, one fuse of the left channel had blown. I am reluctant to switch on the power amp again as I'm afraid if the speakers may get damaged.

Please advise what I can do to fix this and also is it possible for something from the Tube preamp pass through the attached rca interconnects and has caused this?

Thank you

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Hi Folks!

It's a real pleasure to be here.
I'm reading the forum for some time now. I always been curious about the electronics. Started, like many boomers, with dissection of old tube radios.
Unfortunately I left the world of electronics for many, many years but now, in my late fifties, decided to build the Opus Vitae or Opus Magnum... 😉
Wish me luck!

Cheers!
Andy

QUAD 33 capacitors

I have a Quad 33 which was modified probably nearly 30 years ago now (12v battery power supply, new rear panel with phono sockets, new capacitors and many resistors throughout, apart from tone control circuit which I never use) I have noticed that there is a distinct lack of low bass using the disc input now. I am using a Shure V15 III through the disc adapter board, which was modified using Vishay resistors and new film capacitors to provide (apparently) optimal loading. The disc pre amplifier boards use Elna electrolytics which appear to be still within spec according to my DMM tester. Other inputs still sound fine as far as low bass is concerned, and I think that they all pass through the Tape board, so I am thinking that the problem may be in the Disc pre amplifier board. Could it be the electrolytic capacitors? It would be easy and cheap to replace these with modern low ESR types from Cricklewood Electronics, but I would be interested to hear any opinions/advice.

I also still have the original printed guide to modifying the 33 and 303 by Russ Andrews, which I used all those years ago, and I can scan and post them if anyone is interested.

Vocopro FA500 - Member those?

Hi there, hope all is well in audioland...

Have a pair of the Vocopro FA500 with the IcePower modules inside and am currently using them to power some 4ohm bass shakers.

As the manual states, the 1/8 input has a preamp and will get me a higher level, but I've always used to RCA adapters into the back inputs. The preamp was always the weak point in these and know most bypassed it with a Ghent Module I believe.

Assuming I want to run this unbalanced, I have a few questions:

- Does running with the RCA adapters into the 1/4 ins actually bypass the preamp?

- Will the Ghent Audio modules get me maximum output, bypassing the preamp and getting max gain possible, and louder than the 1/4 adapters I'm using now?

Clearly audio quality isn't priority in the bass shaker world, but would like to keep these amps useful as hifi amps too and wasn't sure what I should be modding to get what I want. Thanks

A Kenwood KR6030 anomaly - input selector switch or component on verge of failure?

Thankfully the garage/estate sale season is over.

There are no thrift stores to speak of in the small city where I've lived for three years so my scrounging urges compelled me to attend garage/yard/estate sales in the area. I am also compelled to look for audio equipment and reorded music. Consequently those urges have caused me to bring home a few finds.

The last audio component I obtained in such a way was a Kenwood KR6030 receiver. I knew nothing of it's history other than the deceased owner bought it new and that fellow's next of kin were disposing of his belongings.

Once I had it home, I looked online for info on that model and saw that it had some respect. I also learned about the potential failure of it's power on / power off / speaker selector combo switch and it so happened that this receiver displayed a telltale sign of that failure. After reading that disassembling then cleaning the switch contacts plus installing a triac device was a recommended fix, I did that and it was successful.

There also seemed to be a need to spray contact cleaner inside the toggle switches. That was done and after which the switches were exercised 100 times. I performed that process twice and also included the rotary switches and potentiometers. Another chore I undertook was replacing all of the electrolytic capacitors on the power supply board but did not disturb the two large filter caps.

Afterwards, as far as I could tell, the receiver worked well. That is until I gave it an extended listen while playing LP's during which the volume periodicaly dropped slightly at random intervals. It was akin to turning a 10dB loudness switch off then engaging it again. The sound would recover, play normally for a short time then drop again. That cycle repeated over and over and it seemed to occur on both channels simultaineously. It also occurred with more than one input source.

Not kowing where to start looking, I first considered the input selector switch and took as good a look as I could with an illuminated magnifier. I could be mistaken but I was expecting the contact surfaces to be shiny metal but they are black.

The photo is of the switch in situ.

If someone would care to comment if the section of the switch that I was able to photograph looks normal or not, it would be appreciated. To my eye, it sort of looks like paint.

Is it supposed to be black?

IMG_5482.JPG

Dayton SD315A-88 12" DVC 8+8OHM

Hello guys, I'm sure there are many request for help concerning subwoofer builds, but I'm targeting a specific driver in a specific configuration. Firstly, I am NOT a pro at speaker design thus I asking for help here. Forgive me if i don't immediately understand all of the technical jargon... Give me a minute and I'll catch on fast though.

I got the idea of this build from two members here on DIYAUDIO. @xXBrunoXx and @mayhem13. I do plan to IM them with hopes that they will chime-in and further hone the ideas that they cross-pollinated into another build conversation that utilized a different driver.

My build desires starts with 6 each Dayton Audio SD315A-88 12" DVC 8+8 Ohm drivers. I am looking to integrate these driver into my HiFi listening space that is also in development and utilizing 15" coaxial full-range-loaded Open Baffles with 1" Horn Tweeters. The listening space is: 12' W x 18' L x 8ft ceilings. I'm hoping for results with as low distortion as possible, and as low frequency as possible using the driver selected. Amplifiers are a-plenty so that's not an issue.

My primary question for you guys and specifically the two members mentioned above is: What is the best scenario to use these driver as desired? My first intent is to use 4 of the drivers in front left/right-stacked configuration and placing the remaining two in the rear corners of the room. IDK if this sounds silly or not, that's why I'm hoping for great advice here. BTW, any of you guys are more than welcome to chime in... knowing that I am aware that there are always better parts with better potential. I got these drivers on sale and they have decent sensitivity. I don't listen to music at great SPL, clarity is what I hope from this build. Thanks.

Pioneer Transistor equivalents

Hello! In my attempt in cloning the SA-9800 phono pre, I've ran into problems quite quickly. I've finished the PCB, and started getting the footprints on, and when it came to transistors it started going south. I already know I can replace 2SK131S with thermally coupled matched pairs of 2SK170, but other transistors don't seem to be any more available, and I don't wanna risk it and order from AliExpress. Are there modern replacements for those given parts? And do these thermistors have to be specific, or any 10K thermistor will do? Thanks in advance

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Building a radio receiver: Need help on the audio stages

Howdy! I have a project going on right now, where I am building a radio receiver from old tubes I have(type 76s, type 41s, etc), and other components I find around. I know how to make capacitors, rectifiers, and a lot of other components, but the circuit itself is one issue. The detector is another, because I need a datasheet of the 76 with grid current specs, since I need to bias the grid to about -15 volts for it to work as a detector, and I can't use a battery for that. Any tips?

Choice of Transformer or Choice of Design

I have a tube amplifier that I built with a Hammond 270EX power transformer. I understood most of all of this 40 years ago during school. I want to make an amplifier for a friend to play guitar and my question is two-fold:

1. Is there a more economical transformer than the 270EX?
2. I'm assuming that I really need to understand the power "needs" first?

I'm here for a while, so my first thought is (since I didn't design my amp's schematic) to design new circuits around my 12AX7 (two-preamps) and EL34(power) tubes. I want to design an amp that I could legally sell on order from guitar friends. And, back to the Hammond, are their other (cheaper) alternatives that I could look at even if I have to design based on differing power requirements? I know nothing about power transformers other than the curly lines on schematics and in textbooks.

thanks
David

Putting a bunch of wallworts and bricks into an enclosure

First off apologies if this is an already addressed question.

I'm looking to put all of my systems wall warts and power blocks into a single chassis, purely for aesthetic reasons. I'm not looking to do any filtering, surge protection, etc.

Is it kosher to have a standard iec inlet in the chassis that then feeds multiple standard duplex outlets inside? I've never seem it done, so there must be a good reason not too? Basically, it's a power distribution but with the AC outlets inside.

And then for the low volt outputs, there are a few paths, none too complicated.

Thoughts?

Great Heil, great midrange. Needs (super) tweeter?

I have been playing around with my recently bought factory new ESS heil tweeters. Other than the shipping cost and import tarrifs to my country being more than the tweeters themselves, they are absolutely great. They are airy, detailed, full, non fatiguing........BUT

Am I alone in feeling they need a tweeter? I am missing a certain "bite" in the top octave and feel due to their size they don't image all that well.

I was thinking of running 2 Heils per side with a small (super)tweeter sandwiched between them...crossed over at around 10khz 1st order. Either using the same wings (extending them up or down) or with its own horn(as small as possible)....

Added bonus would be a dipole tweeter but I am not ruling out monopoles.

Any tweeter suggestions? I was thinking of using a small neodymium 3/4 inch silk dome without faceplate (or 2 back to back out of phase)....but maybe there are better options?

ROAR15

Hello fellow diy'ers.

I recently received my ROAR15's from the CNC shop and was in the process of loading them with B&C 15SW115 when I noticed something amiss with one of the drivers, coil scratch, so that driver is going to be returned to the dealer.

The other one worked as intended and despite having only one ROAR15 up and running I still wanted to share some pictures and very early impressions (that may change as I get more time with them and both of them working).


They currently reside in our apartment's living room, so I cannot test them properly (you know, neighbors...) but even so I also do not have the means to properly power them, at least not yet.

Anyway, on with some pictures.

The driver baffle and interface:
ROAR15_build_15SW115_interface.jpg



Next to above we see one of the quarter wave segment inlets into the resonator:
ROAR15_build_lower_qw_slot_and_brace.jpg



The post summation quarter wave resonator:
ROAR15_build_aperture.jpg



The one that worked:
ROAR15_build_loaded.jpg



And finally the state of things as present, I will given time arrange things a little better, what you see blow is a work in haste, perhaps I will be moving them into another room seeing as the person I live with has voiced some opinions on the matter, can't say I blame her, they are actually somewhat noticeable.

set_up.jpg


Preliminary impressions of one ROAR15, corner loaded, in a relatively small room at low levels, is that it sounds massive, in this placement it also digs very deep, and I do notice a lot of bass details both in the lows and higher up in it's range.

It does seem to have the ability to be both punchy and dig deep at the same time, this trait it was also noticeable in the ROAR12 previously evaluated but this is a more of everything sort of experience in comparison.

This is not much to disclose given the conditions described above but it is a start.

I will be putting some more pictures and impressions up once I have the new driver mounted, until then.

signal cable inside amp...shield usage?

With reference to signal cables running inside amplifier/preamps such as from RCAs to boards, volume pots to boards etc. In the past I've used twisted pairs of single strand copper that I've extracted from Cat5/6 cables. I've been careful with wire routing and physical separation of supplies etc. and so far it seems to have worked fine. However, I have so far also avoided putting big linear supplies in the same box.

So, in the context of an amplifier using a big transformer in the same case and using a ground loop breaker / ground lift thingy using shielded cables for small signals seems like a better idea:
It appears that in general people use single core coax cable for their small signal routing on and off boards. They connect the core to the signal and the shield to the return which typically is the `signal/pcb' ground. There is an effective faraday cage around the inner wire, but the signal return/ground (the other half of the circuit), is `exposed' so doesn't this negate the value of using a shield? - I appreciate that the far end of the `signal return'/pcb ground is connected to ground via a ground loop breaking resistor/diode bridge arrangement, but in an ideal world I'd assume you'd want to avoid adding noise anywhere in the signal/return circuit.

Follow up question: Does it make sense to use cable that is two core + shield with the Signal and `return/pcb ground' connected to the two inner cores and the shield connected to the Permanent earth/safety ground/case at one end (...one end only Vasily).
?
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What is this brown stuff, and how to clean?

Hi
I was asked to take a look at a pair of Vincent SP-991 amps. Both are dead. That is, no sound, but relay kicks in, and no DC on the outputs.
After some time fiddling with how to take it apart ( no fun task ), I discovered that one of the pcb's ( the input stage on the bottom ) was totally covered in some
brown goo'ish stuff. It is quite sticky. I first thought it might be a leaky cap on the other pcb on top of it, but no. The top pcb is clean.
It is all of the pcb that looks like this, not just a small spot, as if the pcb itself has been sweating, could that be the case?
The way it is build does not allow for air circulation for this pbc. No ventilation holdes around it or on the pcb on top of it.

Sadly i forgot to take pictures before I started to clean. Here are a few images after cleaning round 1

What could this stuff be? and how to clean it ? The caps dont look bad. The goo isnt concentrated around the caps
The other light brownish goes off when using ordinary pcb cleaner and qtips

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JBL M2 for The Poors

This afternoon I had an idea for a JBL M2 for those that don't want to drop $10,000.

But first, some background on the M2.

An externally hosted image should be here but it was not working when we last tested it.


First off, JBL M2 is a two way monitor which uses a similar format as my reference monitors. (Gedlee Summa.) There's a 15" waveguide atop a 15" woofer.

JBL-D2.jpg

The compression driver in the M2 is very interesting. It's a dual ring radiator, very similar to the BMS compression drivers, except with two diaphragms instead of one. BMS sells some coaxial compression drivers, but the D2430K is different. In the D2430K the diaphragms face each other. (Not 100% sure if it's push-pull or push-push.)

diaphragm.jpg

magnet.jpg

Here's a pic of a BMS compression driver that Paul Spencer posted to his blog. Basically it's a ring radiator, that mylar ring is the diaphragm.

c0143750_17234140.jpg

500x1000px-LL-193b1568_P1020185.jpeg

In this cutaway of another BMS, you can see how the sound radiates, basically the output from the ring radiates toward the throat. One of the reasons that the BMS has such good bandwidth is because every point on the ring is equidistant to the throat. (IE, draw a straight line from the throat to the ring, and that distance will be the same no matter where you draw it.)

So the idea I had was to replace the ring radiator with an array of very small drivers.

An externally hosted image should be here but it was not working when we last tested it.


The 1/2" tweeters used in the Keele CBT array are a great candidate.

More details in the next post...

how do I connect subwoofer cables to a car amplifier?

So I have a Rockford Fosgate T1500-1bd amplifier

Its got 2 channels for speaker cables, channel A and B

On my subwoofer I have 2x 4 subwoofers which lead to a single cable, black and red cable

So how do I connect this speaker cable to the amp?

to channel A plus and negative? that means channel B is empty,

or do I connect the red subwoofer cable to the plus on channel A and connect the black subwoofer cable to negative on channel B? so looking like a bridged connection?

The amp is a constant power amp so its confusing on how to connect this amp to the subwoofer

thnks

Mr Noob

Ikea Unity Horn

I have an extremely difficult time with woodworking. Every time I make a box, it comes out looking wonky.

Ikea now sells MDF cubes measuring 35cm x 35cm x 35cm for $20:

https://www.ikea.com/us/en/p/eket-cabinet-white-80334603/

So I wanted to see how suitable they would be for a Unity horn.

Admittedly, the size isn't ideal; it would be nice if they were bigger. But I can't make a good looking box to save my life, so here we are.

Attached is my first stab at it. It used a waveguide that's 25cm x 25cm. Response is pretty rough because it needs a roundover.

Depth of the waveguide is 2.25" (5.7cm)

Here's the ath model:



Throat.Diameter = 25.4 ; [mm]
Slot.Length = 3 + 9.7*cos(2*p)^2
Throat.Angle = 17.5 ; [deg]
Throat.Profile = 1
Length = 57.15 ; [mm]

Coverage.Angle = 62
;Coverage.Angle = 48.5 - 7*cos(2*p)^5 - 12*sin(p)^12
;Term.s = 0.7 + 0.2*cos(p)^2
Term.s = 0.8
Term.n = 3.7
Term.q = 0.995

Morph.TargetShape = 1
Morph.FixedPart = 0.5
Morph.TargetWidth = 229
Morph.TargetHeight = 229
Morph.Rate = 3
Morph.CornerRadius = 76.2 ; [mm]
Morph.AllowShrinkage = 1

; -------------------------------------------------------
; Mesh Setting
; -------------------------------------------------------

; ABEC
Mesh.AngularSegments = 80
Mesh.LengthSegments = 30
Mesh.CornerSegments = 8

Mesh.Enclosure = {
Spacing = 38.1; 38.1; 38.1 ; 38.1
Depth = 330
;EdgeRadius = 19.05
EdgeType = 1
FrontResolution = 8,8,16,16
BackResolution = 20,20,20,20
}

Mesh.Quadrants = 1

Mesh.DepthSegments = 20
Mesh.AngularSegments = 60

Mesh.ThroatResolution = 6
Mesh.InterfaceResolution = 12.0
Mesh.InterfaceOffset = 5.0

Mesh.ZMapPoints = 0.5,0.2,0.5,0.8

; -------------------------------------------------------
; ABEC Project Setting
; -------------------------------------------------------

ABEC.SimType = 2
ABEC.f1 = 500 ; [Hz]
ABEC.f2 = 16000 ; [Hz]
ABEC.NumFrequencies = 61
ABEC.MeshFrequency = 1000 ; [Hz]

ABEC.Polars:SPL_H = {
MapAngleRange = 0,90,19
Distance = 2 ; [m]
;Offset = 140
}

ABEC.Polars:SPL_V = {
MapAngleRange = 0,90,19
Distance = 2 ; [m]
Offset = 140
Inclination = 90
}

ABEC.Polars:SPL_D = {
MapAngleRange = 0,90,19
Distance = 2 ; [m]
;Offset = 140
Inclination = 42
}

; -------------------------------------------------------

Output.STL = 1
Output.ABECProject = 1

Report = {
Title = "Feb25_2024_waveguide"
PolarData = "SPL_H"
NormAngle = 20
MaxAngle = 90
}





feb25-2024-unity.png
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