Illustrated guide to the 4U deluxe Chassis



This is a visual guide to the assembly of the diyAudio 4U Deluxe Chassis. The “Jack of all Chassis”.

It has a pre-drilled back plate and pre-drilled and tapped heatsinks. The heatsink drill pattern is the diyAudio “Universal Mounting Specification” and the back plate has holes for input RCA jacks, Speaker jacks and a Power Entry Module that contains IEC mains inlet, power switch and fuse holder.

This chassis is ideal for the HoneyBadger, F4, F5, Aleph J, or Burning Amp. (The BA- with just one PCB of output transistors/channel)
It could also be used as the chassis for a Monoblock F5Turbo v2 or a Burning Amp with 2 PCB of output transistors.

Lastly this chassis has the “DIY-friendly baseplate” a sub-floor that has a grid of 10x10mm holes making no or little need for drilling holes to attach most of your components to the chassis. You will find that with just a little planning you will commonly need only to widen one hole to accept the mounting bolt for a toroidal transformer. Everything else can be mounted with the existing holes.

The chassis is also compatibletable with the diyAudio Chassis Riser system, allowing for more area to mount whatever your project may require.

Much more information may be found here -

Deluxe 4U "Jack of all Chassis" (All Aluminum) - Full width with 40mm Heatsinks - Chassis

Let’s look at the contents of the package -


Feet and hardware.


Heatsinks.


Top and bottom plate. These are custom aluminum, with an extra row of vents per side.


Back panel.


10mm custom front panel. Obverse.


Front panel reverse. This is a custom part made for diyAudio with the array of holes shown - these are all blind, but most are tapped. For example you can attach the V3 PSU circuit board vertically to the front panel, or attach a riser plate. The bottom centered offset hole can be drilled out for a power indicator LED. Other holes are there to accommodate handles. I’m sure there are many ways to utilize these attach points that the community will discover.


These rails are the mechanical heart of the chassis. They attach to the heatsinks, and then everything attaches into them.


DIY-friendly perforated baseplate.


Everything.


The logical place to start is here, with the rails and heatsinks.
(sometimes there is streaking from the cleaning process. It makes no effect on the efficiency of the heatsink, and is quite smooth.)


Bolt the rails to the heatsinks, but leave the screws a little loose.


The next pieces will be the back and front panels.


Locate the back panel and this hardware.


Attach both sides as shown.




You should have this assembly when done.


Align the front panel and your assembly as shown.


Attach here with this hardware.


Looking good!


Next the perforated base needs to be installed. This is the hardware provided.


It attaches into these countersunk holes in the bracket.


Using the supplied hardware it is intended to attach with the lip up


Which will give enough clearance in-between the black aluminum baseplate at the perforated base to clear small screwheads.
I like to mount it with the lip down, giving enough clearance to clear the through-bolt of a toroid transformer, but it requires some alternate fasteners. (See below for more information.)


Perforated base installed.
Now is a good time to snug all the screws and square the chassis as best you can.


The top and bottom panels are identical, but there is a front and back - but as the holes will align in only one direction, it is easy to determine.


The top and bottom panels attach with the sheet metal screws provided. It is a much stronger interface that you would think, regardless the strength is all in the brackets.

Good morning everyone.
I have just finished my no compromise subwoofer project composed of Faital pro 18XL 1800 thanks to your precious advice. So now I can do nothing but finish the job by also building the speakers.
I have a lot of work to do. At the moment, as a system, I have a Sabre Topping D90 III dac, a Topping Pre90 pre, an Audiophonics ET400A Purifi power amplifier and a Powersoft power amplifier with Loto DSP and 2 Litemod 2ch modules.
currently the towers (PL200 audio monitors) are moved by the ET400A purifier while the subwoofer by a single BTL powersoft module.
I could use 2 more channels to harmonize the subwoofer with the woofers in the project I will realize.
The project will be in pneumatic suspension. Speakers divided into medium-high and low sections. Woofers (2 per speaker) separated from the medium-high range. I am looking for high efficiency, precise, fast, dynamic components. In short, the best you can have while remaining within human figures. I am not very expert in transducers, so my research was very limited.
For the low end of the project I had considered the FaitalPRO 10FH520 16Ω x4 (16 ohm because they will be connected in parallel, but also because compared to the 8 ohm version these can be put in pneumatic suspension, while the 8 ohm ones unfortunately cannot).
Midrange FaitalPRO 6PR160 8Ω d'appolito configuration x4.
and for the high end the Beyma TPL 200.
The main doubts concern the integration of the Faital midrange with the Beyma AMT. From its MMS of 12.5gr (FaitalPRO) it seems that it can keep up with the Beyma.
But as I said before I am not sure because I am ignorant on the subject.
I chose the rest of the components left in Faital for a familiar sound problem, but I have no problem changing brands if there are better drivers that can work with high efficiency air suspensions.
I had thought of putting a wide band as the medium, limiting the band to the medium one. the fostex fe 168 ez sigma drivers. But I didn't know if it was a good idea.
I also have to choose whether to make a passive crossover filter or buy an active one. They say that the passive one is better.
The type of speaker must be a tower, trying not to exceed 1.30 meters in height. and I would like to stay flush with the transducers in width. While I can have margin in depth.

Hello, I have a relative who entrusted me with a CEC TL3 because the player was having problems recognizing the table of contents with two of these Cds, which are originals (not CD-Rs). After dismantling the unit, I found three cut chemical capacitors (330µf/6.3v) on the power supply board. I also took the opportunity to replace the three 47µf/50v close to the heatsinks. I didn't find any other chemical capacitors out of order on the power supply board. I should point out that the laser lens was cleaned in two passes (isopropyl alcohol). After reassembly and tests with several Cds, including the one that was not recognized, there is some improvement, but it's still not optimal. I'm thinking of checking the servo card settings and readjusting only if necessary. For this, I'll need the CEC TL3 manual service. The servo card has two markings on the PCB (TK-T71 94V-0) and (4B10012700C). I'll post a photo soon. Thanks in advance to anyone who can give me a hand.

I want to build the regulate power supply base on Passlabs X2's power supply.
The original on use two transformer to form a network of four 40volt rails attached in series and feed to the regulator as show in pic#1
For mine, I wanna use only one transformer. So, may I use the schematic as shown in pic#2. the 2 seconday form 40volt and 120volt rails to feed the regulator to get the same regulated voltage.

My quesion :

• Is there any difference between two of this in term of performance? apart of using higher voltage cap.
• Can I use IRF610 mosfet instead of IRFP240 for the plus voltage rail? since the the current draw from the psu is about 60mA. Is there a special reason why IRFP240 is picked for this job?

Thank you.

Anyone try them?

According the attached images there are in use different DC connector versions for the outdoor power supply - not only dependent of the model and the age.
4-pin DIN- and XLR versions I note, so as a blue version from Hirose.

Who can upload the whole overview of all used brands and types ?
Thank you very much.

P.S.: The blue socket on PCB from third and fourth image seems to be a HRS (Hirose) RPC2
https://in.element14.com/hirose-hrs/rpc2-12rb-3p-71/plug-chassis-3pole/dp/1077726
https://in.element14.com/c/connecto...IkPXrOiQPIqa71CYuHfX_wVg3NAgSYUHf8Fm-b6ifV3VY
https://uk.farnell.com/hirose-hrs/r...lMXJkXeNffff-UbqayHKFkQLoXvmtmmmDoFJndGNxKsUf
Unfortunately no longer available

Hi, I’m Phil. I have been into HiFi since about 13yo (1970’s). My uncle was a TV and Audio engineer. He was semi retired and working from a home workshop that seemed like an Aladdin’s cave! He gave me an old valve radio that I set up at home with a long ariel. I think the warm sound, the heat, and the glow of the valves got me hooked. I’m retired now myself and hope to learn enough about electronics to fix up some equipment, and maybe build some kits.

Good evening good people,

I just bought a secondhand GK mb 112 ii. It had the pink/purple led (instead of the normal blue) on the power button, which indicates a fault, but it still worked fine and sounded good, so I took a gamble on it (I got it for around 150$). I'm trying to troubleshoot the amp and hopefully make it as good as new. So far what I know:

- The amp's manual states : "Smart Protection Circuitry: The MB Series protection circuitry constantly monitors for unsafe operating conditions such as short circuits, miswired speaker cables, blown speakers, improper ventilation, and incorrect speaker configurations. In addition, protection is provided during power up/down. If the MB Series senses any unsafe operating conditions, the output signal is immediately muted and the power light changes from Blue to Red. The amplifier will remain muted until the fault is removed. Be sure that you are not exceeding the maximum recommend load described in this manual. If the problem persists, contact GK directly for technical advice."

No info about purple light, or if it can still work if the protection circuit detects a fault.

- When I plug a bass, turn the amp on, I get sound. When no bass is connected, there's a very low hum, that is (very) slightly affected by the gain button and the -10db pad and moderately affected by the treble control.

- I tested the speaker on my guitar amp too and it works perfectly fine.

- The phones output is not working. I wonder if the purple light can be caused by a faulty headphone circuit ? If it’s the case, I can totally live with that.

- The DI output does work, but I can hear some static on the recording.

This is my first attempt at repairing an amp. I think it would be a good learning experience. If all fails, I'm gonna turn it into a cab. Any help would be appreciated.

Thank you

Hi All. I am a new member from Australia and found many links to articles on this site while I have been repairing musical amplifiers in recent times.
My interest in electronics started around 50 years ago when I was building small DC power supplies and music effects devices - all discrete components.My "old" 6w valve amp from 60 years back is now in vogue! But then I got into solid state transistor powered amps - made a couple of 100w kits all good.
I have been a guitar and bass player for even more years. Since retiring from professional work, I have recommenced my interest in the same equipment but with many "new" components. I have had a few repair successes and have a few projects for further work.

This thread has the goal of collecting schematics, pictures and comments of any kind about amplifiers where ALL the signal tubes are directly heated - both input and output tubes. The only indirectly heated tubes allowed will be in the power supply or support positions e.g. rectifiers or active loads. Can be push-pull or single ended, and directly heated pentodes are also allowed, in triode or pentode.

Since the majority of directly heated tubes are fairly low mu (up to 15) it is likely that designs will include 3 stage amplifiers or 2 stage amplifiers with step-up transformers in the input or interstage. Maybe also some other creative solutions.

So please post your ideas, schematics and comments of any kind and let's make this thread an interesting one!

To kick off, here's a design by Ale Moglia titled "300B SE Amp: 47 Driver: Going DHT end to end"

https://www.bartola.co.uk/valves/2020/07/25/300b-se-amp-47-driver/

Even the best loudspeakers sound quite poorly in nowadays squeezing living rooms. The problem of room reverberation can be addressed by extensive soundproofing and/or by using very directional loudspeakers - but covering the walls and ceiling with class A acoustic foam is not cheap & utterly destructive to the peace in the family; commercial controlled directivity loudspeakers cost a fortune while developing your own without an anechoic chamber nearby is unrealistic. However, in some situations it is possible to use the room geometry to reduce room reverberation and improve the listening conditions.

Introduction​

Let’s consider an alternative as sparse (3…6 drivers) speaker arrays positioned in the room corners so that the floor & ceiling of a typical room create a tall virtual column. The driver’s height is defined by


where C2F is ceiling-to-floor (typically 9’, ignoring 1’ of ceiling space for pipes, etc) height, NS is the number of drivers, and H is the ear height for a sitting listener (typically 2’8” … 3’). Drivers shall be focused (arched and angled, as per near-field beamforming) onto the DIYer and his ever-grateful family at 9’…11’. In (a simplified) theory, such array shall have wide horizontal dispersion and its impulse response (on the axis) shall look like


with floor and ceiling reflections interleaving, each having a small (~1/NS) relative amplitude, with loudness decaying as (1/sqrt(distance)), contrary to (1/distance) for point source loudspeakers. However, I could not find any published (by others) real-world measurements, to either confirm or disprove this theory. Thus, I measured what I could myself. Of course, the results are specific to the room & drivers – but something, even little, is better than nothing.

Methods[1]​

This design and measurements are concentrated on midrange applications, 300Hz – 3kHz, of the maximal sensitivity of human ear to the Direction of Arrival (DoA).

The drivers are Dayton Audio Sig150-4 mounted in approx. 3.6L boxes angled 60⁰ inwards, which boxes are mounted on a long plank which is mounted into the corner of a somewhat spartan room 25’ x 14’ x 9’. Multiple drivers were connected in series.




The amplifier output was set to get 80 dB SPL @1m from a single driver. A (pre & post) calibrated 12 dBA cardio microphone on 2i2 Gen 3 USB audio interface was used. Room Impulse Responses (RIR) were obtained by a regularized Exponential Sine Wave (ESS) method Loudspeakers for AEC: Measurement and Linearization - File Exchange - MATLAB Central (mathworks.com) (to be updated).

The RIRs were postprocessed to extract the Direct to Reverberant Ratio (DRR(f)) as the Y-value that the polynomial approximation of the corresponding normalized reverberant slope(f) crosses at X=0:

Obviously, speakers’ arraying does not affect RT60(f).

Measurement distances are 100:50:400 cm, with 300cm as target. The default measurement height is 85cm, to be varied 70:5:100cm. The first 20ms of RIR, DRR, arete (top of the ridge) of wavelet-transformed RIR, and total FR are considered to be more important than other metrics.

Results​

number of drivers [h->1, n->3, m->4, n->5, j->6] summary​

At target distance & nominal ear height. See the details below.

Single Driver​

Red lines are for quasi-anechoic conditions (alas, wrong polarity).

6 drivers​

Variations of Listener’s Distance​

Here and below, red lines are of a single driver at 300cm, as the reference.

Variations of Listener’s Ear Height​

5 drivers​

Variations of Listener’s Distance​

Variations of Listener’s Ear Height​

4 drivers​

Variations of Listener’s Distance​

...oops, only 20 images
Variations of Listener’s Ear Height
3 Drivers
Variations of Listener’s Distance
Variations of Listener’s Ear Height
6 drivers, straight line [e] vs angled only [f] vs arched only vs arched and angled [j]


As usual, [h] stands for a single driver as baseline.

Other Observations​

• The measurements show only 10 dB of improvements in DRR but subjective perceptual improvement is much stronger, far beyond expected.
• Due to 15.5 dB lower SPL at each driver, non-linear distortions and Barkenhouse noise are proportionately lower.
• The dynamic range widens, high SPL are not distorted.
• The (1/sqrt(distance)) effect is realized only partially per measurements but nearly fully by perception.
• Judging by ear at 10’, 6 drivers with 90dB SPL @ 2.83V sensitivity are louder than a 110 dB SPL single driver, ~same Z.

Discussions​

Many more drivers & configurations were built and measured in the last year. None of them contradicts the data above, and these have relatively low additional meaningful information and were omitted to avoid overloading readers.

TBD

Conclusions​

• Sparse speaker arrays are a compromise. Particularly, they are not a good fit if you require that listening experience shall be invariant to ear’s height. Left/right/closer/farther are not a problem.
• You do not need 20 drivers / column to realize line source advantages.
• You do not need near-perfect expensive midrange drivers to get decent sound.
• You do not need 400W amplifiers, for sure. Even 100W is too much.
• An average DIYer can build a sparse speaker array and adjust it to his preferences without getting into much expense or serious technical difficulties.

GNU General Public License v.3+ https://www.gnu.org/licenses/ applies to all information provided.

---

[1] Disclaimer: The objective measurements below are somewhat related to but do not characterize human auditory perception to any highly disputable degree of fullness or precision.

Hi guys, I'm looking for PCB's for the RAW DSD dac 🙂 or meaby soldred one 🙂

Thanks

Not the usual source, an Architectural “magazine”. Looks like a waveguide, which has one asking, “Midrange?” but there is a short c0omment at the very end that suggests it is an MEH, but the second picture (and the makers bumf) suggests something else. Pretty cheap given the amplifiers come with (if pricng is /pr)

https://www.dezeen.com/2024/11/05/aluminium-cast-speaker-tom-fereday/



Pitt & Giblin: https://www.pittandgiblin.com.au/cast

More coverage:
https://www.wallpaper.com/tech/cast-aluminium-speaker-tom-fereday-pitt-giblin
https://leibal.com/products/cast/
https://www.gessato.com/cast-aluminum-speaker/

dave

My Kenwood Basic T1 tuner has gone silent all of a sudden... Any advice, how to test outputs? Any mods out there? Lights all working, auto tunes to a station but nothing. Any ideas? Thanks!!

Anyone have any favorites?

I have many but I just pulled this one out of my collection and am impressed every time I listen to it. I have the CD which sounds excellent but may get the vinyl version as well...

https://ecmrecords.com/product/break-stuff-vijay-iyer-trio/

This work started in the Filler Driver thread. I was studying transient XO's and built a test bed speaker to quickly test out new XO ideas with miniDSP. Here is a photo of my test bed. It is a basic open baffle for ease of construction and has a pair of Eminence Beta 12cx (no CD, with coax horn sealed up), a pair of PRV 5MR450NDY mids, and a PRV D280Ti with a PRV 45x45 deg waveguide. These are all fairly efficient drivers and are what I had on hand. The amp is provided by TDA7498 100w class D amps powered by 19v (mid and tweet) and 24v (woofers) SMPS bricks.



I first started with the B&O XO which requires a wide bandwidth mid that is band pass filtered with 1st order Butterworth and straddled on either side with a LR2 filter for the woofers and tweeter. I was then turned onto the XO topology proposed by Samuel Harsch by member Phase_Accurate. The paper is in French but I figured out that the basics are:
1. Set the low pass filter for the woofer as a 4th order Butterworth at central frequency, fc for the XO centerpoint.
2. Set the high pass filter for the tweeter as a 2nd order Bessel at fc.
3. Set the delay of the tweeter equal to 1/2 of the period of one cycle at fc.
4. Use all positive phase on woofer and tweeter.

Here is an example of the XO plot from the Harsch paper:


Here is the step function from the above XO as predicted by Harsch:


So I tried it out on my 2-way 10F/8424 & RS225-8 FAST Ref Monitors and got this for the XO plot:


For the 10F/RS225 FAST, this was the corresponding measured Step Response:


For the 10F/RS225 FAST, here was the measured phase (notice the little bump at the XO frequency as predicted - but no wrap):


For the 10F/RS2225 FAST, here was the measured group delay, not over 5ms above 40Hz:


What the Harsch XO does nicely is that it allows one to use a steep XO to block out the nasty cone breakup on a woofer with a high order filter. Yet still not have a large phase wrap in order to maintain a time aligned transient perfect response.

So I got to thinking if this can be applied to a 3-way by mirroring the XO topology. That is Butterworth 4th order for the low pass on the woofer, Bessel 2nd order for the mid high pass and Bessel 2nd order for the mid low pass (or Bessel 2nd order bandpass for mid), and a Butterworth 4th order for the tweeter high pass. So what this means is that pretty generic components can be used in that a special wide band mid is not required anymore as the filter is already a fairly steep 2nd order. And for the woofer and tweeter, we can use 4th order filters which are great for controlling distortion, and preventing cone breakup leak-thru from the woofer.

As I had a 3-way XO test bed already handy, it was just a matter of putting the XO settings into miniDSP. So here is the electrical XO I implemented in miniDSP:



And here is the resulting acoustic XO that was measured (yes the SPL levels are fairly low so that I did not wake up the family during sweeps). As you can see, the curves are fairly common and innocuous. The 1st order textbook curves of the B&O were especially hard to achieve in practice. With the Harsch topology 2nd order and 4th order are much easier to get. The tricky thing is what to do for the time delays? Well, it is exactly the same delay assuming driver acoustic centers are already time aligned. The mid needs to be delayed from the woofer by 0.5/fc_low and the tweeter needs to be delayed from the mid by 0.5/fc_high. For my case, I used a 500Hz and 3500Hz XO points:



Here is the resulting Impulse Response:



And here are the resulting Step Responses - I know there is a little pre-rise but overall it is much better than a jumbled up-down-up-down mess you would see with an all LR2 or LR4 XO:



This is still a work in progress. Does anyone see any problems with it? I think this works. I am still having a problem getting the phase to be wrap free - it seems to wrap at the centeral frequency and not at the low and high XO frequencies. Anyhow, still debugging it. This XO may be useful for folks looking for an easy to implement quasi-transient perfect XO using IIR filters in DSP. How does it sound? Fantastic! My speakers are essentially 98dB sensitive and super loud and clean with this XO. I am really enjoying the sound and how the percussive sounds like guitar and drums really have a "live" feel to them.

Attached are some sound clips (stereo recording using Zoom H4 of the mono- Right channel playing). Change file extension from .asc to .mp3 in order to listen.

Alright so we are here again. Originally I was going to run only run two monitors in this new office but I need to go to three. I could move the monitors aparts by maybe 5 inches but that will make working harder. More head movement.

So yeah, once I actually got in here I realized the 3 way I was making is just going to be too large for this work area.

What build would you make for this setup? I really want a three way but I don't think that is possible unless I mount the speakers on the wall above the monitors. I could put them on adjustable arms have them pointed down at me. The window kind of gets in the way of that.

I really don't want to be boring and put a couple 3" Mark Audios in some spherical enclosures underneath the monitors pointed up at me.

Anyways, let me know what you'd do if this were your new space

Are the any people local to Hazel Dell?

Edit: Moved to another part of the forum.


Very best,

I'm announcing the public release of XSim, a "free-form" Windows-based passive crossover design and simulation program intended to be as intuitive and non-restrictive as possible. . The program is ***free*** for personal or commercial use. Any speaker builder with at least a basic understanding of what resistors, capacitors, and inductors do, and with decent response measurements (FRD and ZMA) of their drivers in their boxes, should be able to get useful results right away.

download from:
http://libinst.com/Xsim/XSimSetup.exe

Experimenting with "ideal" driver data (the default for each driver) can also be fun and helpful in understanding the effects of various circuit types. Some built-in multi-part "Circuit Blocks" also provide for basic pre-configured circuits to be tuned per their overall parameters (Q, corner frequency, attenuation, etc). There are "Example" files inlcuded in the XSim download (currently less than 1.5MB) as well.

XSim allows you to see simultaneous multiple graphs of your choice, including impedances, voltages, power consumptions, group delay, and frequency, impulse, step, and square wave responses. You can watch the performance and/or stress on any part vary as you tune structure or component values in real-time. The circuit network is re-solved automatically with every change you make.

The "free-form" aspect of XSim is that crossover designs are unrestricted, no particular circuit structure need be used. Essentially any R-L-C passive network you can think of can be easily entered using graphical schematic entry, with results calculated as you design. A common "ground" node is provided, but its use isn't required.



Circuit designs (incuding all in included driver data sets) can be shared by email or forum post in XSim's ".dxo" format.* Graphic files can also be saved in standard picture file formats.

Stuff yet to come: [edit: the following aren't actually going to happen now, as it is too much work to herd vendors into getting this info together and hosting it on their sites!]
The release version of XSim featiures automatic downloading of part and driver information based on vendor and part number, making it easy to find and order what you need to turn your design into working hardware. However, no vendors are as yet signed up to provide XSim access to part or driver information. XSim will be able to download these lists itself, when available. XSim can also update its own program (with owner permission) so people who use it online can always be up-to-date. ...And so I can add or fix features without asking you to deal with downloading and installing each time.

XSim does not yet include geometric features (such as varying mic position, baffle or boundary-bounce effects) as in Jeff Bagby's very successful "PCD" and other programs derived from it. But that is in the works, with some assistance from Jeff. There are also as yet no box-design features, but this is also planned so that full designs can be worked with using the infinite baffle data provided by vendors. [these might still happen, when/if I get back into programming and study mode 🙂 ]

edit: * as it turns out, this forum doesn't let you attach the .dxo files (or other files it doesn't know about). So you'll have to first zip the file and then post the ".zip" file containing the ".dxo" file. On downloads, do the opposite: unzip and then open the .dxo file into XSim. Maybe someday we can get diyaudio.com to let these files be attached? They are only text, no way for a virus to sneak onboard one of them!

Added Nov 2017: There is a procedure summary for collecting and adjusting driver FRD data for correct relative delays (which you need to accurately simulate crossovers) here -- XSim free crossover designer

Morgan here. Long time car stereo enthusiast. Hoping to learn a lot here.

Hi all,

Audio well where do I start?
Grown up in a music cafe with Saturdays live music created interest at a very young age
Then when 16 became a FOH Sound Engineer in the late 70s so all ANALOG gear 🙂 but of course moved over to Digital

Other hobbies are electronics, audiophile, sea sailing and mostly computing as work next to FOH

My Audio start time was mostly McIntosh, Crown, Altec & JBL and later T+A

Current having Urei speakers (love point source) and Tube Amps with one SE EL84 from the late 60s or may be 50s and a Croft Phone Integrated (8 EL84)

Hi there. I want to buy the 103 power amplifier (2 box unit, separate PS) if anyone has anything spare.

I will be matching this with the matching 101 preamplifier.

Working or non-working 103 unit is fine. Please note that this will need to be sent to me in New Zealand.

Thanks! Simon

Various Tubelab populated PCB Board for sale.

I populated them with quality parts, then never completed the builds.

They should all be working, but you may want to double check them in case I made any mistake on them.

• 1x Tubelab Simple SE (dated 2006, no cathode resistor installed), $100
• SOLD: 1x Tubelab SSE (dated 2012, 680r cathode resistor installed), $100
• 1x Tubelab Simple SE (setup for KT88), $100
• SOLD: 1x Tubelab TSE-II (set up for 300b)
• SOLD: 1x Tubelab TSE-II (set up for 45, photo attached), $100
• SOLD: 1x Tubelab SSE (dated 2012)
• SOLD: 1x Tubelab TSE-II (dated 2020)
• 2x Tubelab Universal Driver Board PCB 2015, new unused $30 each

Need to add shipping from Brooklyn 11211.

Accepted payments:

• Zelle
• Paypal (add 3% fees if "as purchase")
• Venmo (add 3% fees if "as purchase")
• Revolut
• Wise

Thanks

With lots of help from posters on the site, I put together this modular three-way top for small outdoor concerts. It's still a work in progress, but so far I'm thrilled.

It's a 3 inch compression driver, a 10 inch cone and a 15 inch cone. The 10 inch is from an EAWKF 650, and the 15 inch is from a bag and TA 15. I have attached PDFs of what multiple sources have stated are the closest equivalents to those drivers in aftermarket gear.

It occurred to me that the TA 15 I got the 15 from was crossed over at the same point the 3 inch compression driver crossed over, so for smaller shows I could bring a two-way speaker, the main drawback being that a 15 inch will be beaming for the upper frequencies. But it worked pretty well for shows where there was small audiences and limited budget.

The other day I noticed that the specs on the 10" driver, which actually weighs more than the 15 inch driver (it's a beast!), is rated to go down low enough that I could create a two-way system with a top two components crossed over at say 120 Hz.

With some planning and effort, I can pull the equipment out of storage and find somewhere to test it outside of a gig, but I still felt like I wanted to ask advice, in case either it was categorically doomed to fail, or if I'm taking a chance of some sort of long-term problem if I take this route.

it feels very counterintuitive to me, replacing a 15 inch with a 10 inch, but again, this 10 inch weigh more than the 15 inch and has amazing specs for excursion and says it can cover the frequency range.

So please share opinions about whether this is something worth exploring (or avoiding it all costs 🤣).

I have 6 Quad 405 clone boards if anyone is interested, 4 modules have ST branded BUX41 power transistors, and two have 2N3773. All have LM301AH op amps in a can package.

Details on eBay here https://www.ebay.co.uk/itm/305892349275 Open to offers.

Hi, I got hold of a vintage Norwegian Solid state Amp from Electrocompaniet and would like some support to restore the high-end piece with some help from the diyAudio forum members.
Regards
Arne
Løvenstad
Norway

My Velodyne FSR 10B6 sub was working for about six hours and suddenly made a cracking sound then the sound stop but the subwoofer power LED still shows Green. I turn it off from the wall socket and after about 3 mins, turned it back on. Now the Subwoofer power LED is green but when I connect an audio jack using line in, there is no sound from the Subwoofer. Please, I need help. Thanks

My name is Flo and i am looking for help with my speaker-cabinet project...
Already found a lot of information here and hope to get more out of this forum.
Kind regards - Flo

Ages ago I modified my WAD Phono II scratch build. Inspired by a Nick Gorham modification (attached), I added CCS on top of the first stage. I also put a CCS under the third stage CF (not documented here).

After a couple house moves, I lost all documentation of my current build. And I like it so much, I thought I’d document it here.

So I pulled the bottom off my phono stage to take a look at what I had done.

The CCS on top of the 1st stage is classic Gary Pimm, and I documented it on Nick’s modification attachment. I believe I set it for 0.9mA, but it looks to be 0.94mA as measured.

The first stage is the hard part as I understood the mod did change the RIAA equalization (3180+318).

I did the math back then (I think), but I document the ("unchanged") first stage output resistance here.

The WAD ECC83 operating point (schematic attached) is B+ = 250v, Anode load resistor is 100k, 150v on the plate, -1.1v with 1.2k (un-bypassed) on the cathode. See the load lines attached.

I calculated anode resistance as (332v-78v)/(3.5mA-0mA)=72.57k

Since it is not bypassed and u=100 and cathode resister = 1.2k, I get 72.57+[(100+1)*1.2]=194k

That in parallel with the anode load of 100k = 66k

With the CCS load, I used 1M resistance (Nick Gorham's suggestion) in parallel with 72.57k, giving 67.7k

So 66k vs 67.7k per my calculations. Close enough?

Referencing Morgan Jones I can’t prove out the rest of the equalization circuit (oh well). But I sure do like this mod.

I’m thinking one day to figure out how to put a CCS on top of the 2nd stage valve. (Allen Wright's approach - if it is not supposed to move, lock it down tight.)

I attached a copy of the voltages of my modified build too.

Good morning,

I am looking to repair a Yamaha R N500 amplifier.

The amplifiersafe in safe mode. Nothing is displayed on the screen (which doesn't make things easy!). Only the power indicator seem to work.

I forced the amplifier to start using the method described in the service manual by protecting the amplifier with a safety bulb.

The relays have all activated but not the screen!

In terms of voltages it doesn't seem bad, I have a 3.3V, a 5.5 and a 12V.

The temperature sensor is not the cause of the problem either.

Do you have an idea or a path to follow to try to troubleshoot it ?

Thanks in advance

I have for sale some parts ,some used and some new.Plese look on the picture.If there is something you are interested just contact me.If you have question also plese contact me.Payment by paypal.

List!
Clarity ESA 250V caps:
2 x 9.1uf - SOLD
2 x 4.5uf - SOLD
2 x 6.8uf - SOLD
2 x 3.0uf - SOLD
1 x 12.0uf
2 x 14.0uf
2 x 68.0uf
Clarity Csa caps 250V:
4 x 1.0uf - SOLD
Mundorf caps:
2 x 1.5uf Supreme Evo oil - SOLD
2 x 1.0uf Supreme - SOLD
2 x 1.8uf Supreme - SOLD
2 x 0.22uf Supreme - SOLD
1 x 0.47uf Supreme
2 x 2.2uf Evo oil - SOLD
2 x 2,2uf Evo Silver Gold Oil - SOLD
2 x 0.22uf Supreme(red writing)-SOLD
Jantzen caps:
2 x 5.6uf Standard Z
4 x 0.82uf Superior Z
2 x 0.47uf Alumen Z - SOLD
2 x 0.22 Silver Z (totaly new)-SOLD
Duelund caps:
2 x 0.1uf JDM tinned copper- SOLD
Mundorf back lack coils:
2 x 0.68mH - SOLD
2 x 5.6mH - SOLD
2 x 0.33mH
Intertechnik coil back lack:
2 x 0.15mH
Russian pio caps K-75 series:
2 x 0.1uf 500V
Ampohm PF-XTI Alu pio caps:
2 x 0.1uf
Miflex Kp-Al caps:SOLD
2 x 0.022uf
Mundorf M-Resist ultra resistor:
2 x 33.0 ohm
Tube socket:
2 x UX-4 backelit top quality-SOLD

2 X 1,20 mH Jantzen air coil 16awg

Jantzen Superior Z caps:
2 x 4,7uf
2 x 1.0uf
Jantzen Standard Z caps:
2 x 3.3uf
Mundorf CfC16 Copper foil coil:
2 x 1,8mH
Mundorf air back lack coil:
2 x 0.18mH

Good day, Gentlemen.

I am totally not a tube guy, but old pal desperately asked to overlook his Japanese domestic market ATM-2, which got issues. Among some simple-solving things like no voltage at the front end filaments, there is a thing which made me stumble: ATM-2 has indicator to set up bias manually and the circuit with indicator resistors was totally cut-off (modding, he-he). Now I am trying to understand how to measure bias at KT88. I studied some old posts here and found schematic (hand-drawing) which is very close to what I see on my table (attached), apart that there's EL34 instead of KT88 (no big difference). Figured that KT88 bias shall be around 30-50ma per tube. But what I see there is - no either anode or cathode resistor. Both cathodes are directly on ground, anodes directly to the trafo. Do I need to solder-in 10 ohm resistors to cathodes, set bias and then to remove them? I am not plaiing to re-instaill indicator circuit. What I can measure is about 480V anode-to-cathode and 0,3V between anodes of KT88 pair. Please help.

Some guitar speakers have totally straight-sided cones with or without ribs (many of these are seamed, some are pressed pulp), some have slightly curvilinear cones with ribs (most of the 12 inch Celestions are this way), and some have more obviously curvilinear smooth cones (EV, JBL). I originally thought that the classic Celestion guitar speakers had straight-sided cones, but upon closer inspection I realized that there is a slight curve.

My question is this: what types of curves are typically used for the curvilinear cones? Exponential? Bézier curves? Something else? I've found some references to exponential cone profiles, but most just say 'curvilinear'.

As the title suggest..
I built a pair of Pensil 10 cabs for my Mark Audio Alpair 10P.
I built them, over Frugal Horn as they were easier but now I want to know if I'm missing out lol.
My room is 17ft x 13ft. The speakers fire across the shorter side and the staircase is open and directly behind my listening position.
Would Frugal Horns bring much more to the party?

Thanks,

Rik...

Listing link:
https://www.ebay.com/itm/162050795247

Looking at the print out on the transformer, I can't figure out what is SCR (Yellow-Green)?
Is Green EARTH GROUND? What is SCR? And I don't see Yellow. I'd like to know the connections before I buy.
Thanks for your input.

Hello all & happy labor day -

I just posted the design of a new power amp that I've been working on for some time.

This is a push-pull amp pentode using sweep tubes. Not screen driven (sorry SY) but pentode mode, with plate-to-grid feedback. Using 6JN6 or 6JM6/6GV5 tubes it does about 18W. Works with no global feedback, or with 6dB of feedback.

Drivers are also pentodes. The amp is set up to cancel some distortion between the driver and output stages (yes, it really does work!)

My goal was a practical and affordable amp. The total BOM cost is about $325.

Details are here:

http://www.pmillett.com/DCPP.htm

Pete

After a long discussion.. WE have decided last night to offer Mono Block/Power Amp in Fall 2024
I will offer two types of Push Pull amp (Mono Block / Power Amp based on KT170 ( can use other KT tubes))
It is a Push Pull amp. The power is 40~50W per Channel

PCB KIt:
It is a 50W per block with KT170/KT tubes. OPT :TBA

Once I have more info I will post in here..

GOOD and Affordable ....Made in Japan...

Hallo everyone,
I had a 12V PSU got broken after unplug violently from wall socket.
Odd....

The schematic resemble this one, exept for resistors values and different MOSFET and DOUBLE DIODE semicons. I've highlighted some paths.
Red: Vcc
Blue: GND
Orange: B(1)+ // B(2)+
Purple and pink: trigger and sense of MOSFET


The issue is:
The putput voltage is floating around 12v when no load is connected (only DMM).
To ensure that is not output caps I added a 1000u. Nothing changed.

Now, I connected a load (12V 0.3A led lamp) and the voltage now ramp up to 10V, sufficient to trigger LEDS and then drop to 8. It rises again to 9.5-10 and drop. The frequency is 200 flash per minute or 3.3 flash per second.

I'm a newbie in SMPS field. The pcb doesnt show any failure component.

Can you help me indetify the culprit?
I would exclude 12V part of circuit. I think the stage before is causing this issue, that then pass trough TRAFO.
Or the octocoupler part.

When Ive removed brutally the PSU from 230V what could be happened?

Thanks in advance
Gabriel

I just received my "Thomann T.racks 4x4 Mini DSP".
These sell in Germany and most of Europe for €119 inc. P&P.
For those frustrated by ADAU 1701 boards like the Sure/Wondom or miniDSP, this may be the thing to get what they expected from a low cost DSP.
By the way, I have some of the DSP boards that need Sigma Studio.

The T.racks 4x4 Mini DSP comes with a Windows software and plugs into USB. No messing around with Cypress interfaces and Sigma Studio Software (which is great if you like mighty, complicated things). just what you would expect from Plug&Play.

The software has all you may need for an active speaker, 2.1 setup or even a small PA system. For the last one IMO it is a must have.

To show you what you get for such littlel money, I made a bunch of pictures. Quite some value compared to others.

The case is steel and aluminum, seems close to indestructable.

PS I have no connections to "Thomann"

I have made calculations for a 4th order State variable active filter. The filter can be configured either as Linkwitz-Riley or Butterworth.

The nice thing about a State Variable active filter is that the same circuit has complementary outputs for Low pass and High Pass. The cutoff frequency is set by four identical RC pairs. Changing the RC pairs changes the frequency for both Low-pass and High-pass outputs, Gain can be set within a wide range. Unlike Sallen-Key implementations which need separate High-pass and Low-pass circuits with matching RC combinations and constrained gain options.

The attached report describes the analysis, calculation, simulation and measurement on a real circuit.

Hi, I'm new here and planning to build an acrylic column with sub. More to look nice, but also should sound good.
4x PLS-75F25AL02-08-ND 3.5" woofers (120Hz-800Hz)
4x SLS-85S25CP04-04-ND 3" full range (200Hz-5kHz)
2x XT25SC90-04 tweeters for the small size (>5kHz)
12" sub (<120Hz)

1. Do I need just 2 tweeters or 4 in this case, as the tweeters are rated 100W?
2. Do I put the tweeters in the middle or on top, as this will be mounted close to the ceiling?
3. For the 3" full range, should I just go full range 200Hz-20kHz or will there be breakup >10kHz to take note of?
4. Should I xover the tweeters at 3kHz instead of 5kHz?
Also, I'm thinking to use active xovers like ADAU1452 so I could tweak the freq as I go along. Will need 4 amps incl sub.

I've got some blown JBL LE25-2 cone tweeters from a pair of 4312 monitors in need of rebuild / repair. Who makes decent recone kits for these, being JBL no longer sells any parts for these?

Some ebay listings show $150 for single NOS JBL recone kits. Cheapest aftermarket kit I found was $75, which isn't terrible, but still pretty steep IMO for an average performing cone tweeter.

I also see aftermarket LE25s which are somewhat well reviewed, but I don't trust online opinions for this, being the tweeter supposedly replaces all sub-models of LE25 and LE26. If it weren't for the sake of originality, I'd just make an Audax TW025A2 dome tweeter fit, which likely will outdo the average LE25, but the owner is a devoted JBL fan and wants them to be as original as possible.

Any other thoughts or ideas?

Hi,

I'm making a multilayer DAC pcb.

I have a 50 mils slot between Agnd & DGND.

On one of the pcb edge I have a resistor after a 1 mH coil that bridge the power planes of each side. It is for distributing some power to a batch of digital ic, flip flops.... on the digital side of the pcb this distributed power plane has almost no reference use for signal, but as a ground return path for a reference layer where smd caps end.

I am worrying about that slot for the current return path towards the PS.

Should I continue a reference plane on a layer till the plarity of that power supply smoothing cap near the secondary ? just on the half part of the smoothing cap below the power plane of that polarity ?

that reference return path is independant of the others reference layers, so not coupled by via fencing for instance.

Or does the current will jump straigth that 50 mils slot gap ?

Or a stitching cap of say 100 nF across the 50 mils gap near the bride resistor is another feasible way for the return current path , bridging an AGND to a DGND ground layers ?

Thanks

Pick any title!

Navajo
With Diamond Lateral MOSFET output stage.
THD 0.000044% at 1 Watt
16 Watt output with low distortion

Hello, I am a maintenance technician living in Thailand. Now I have received a DENON AVR4311 which has no sound, but as far as I have checked, there is sound only in stereo mode and direct front speakers. I tried adjusting the sound mode, the Center Surround channel also has no sound. I checked the firmware version, everything is normal, but the position, sound and video show the version PLD:00.00. What does it mean and what is PLD? I don't understand.

Equin single rail power amp, with modified Quad303 triple cascade , came in Elektor January 1976
Parts PCB foil were sitting around in some box for almost 40 years.
LTSpice sim comes almost too good to be true. Perfect harmonics spectrum.
Result shown with sim of two way speaker at 200 Hz, the frequency where most music with acoustical instruments has max amplitudes. The speaker has exactly at 200 Hz impedance minimum. comfortable 4 ohms. More simulation required to test emf feed from speaker , and thermal distortion. Harmonics spectrum remains unchanged with frequency like a tube amp. Thd at 42 watts is 0.012905%
Power BJT TIP3055 is same as 2N3055 but higher thermal junction to case res. Single diffused. Faster BJT are not as rugged especially epitaxial base would require SOAR protection. The diodes D2..D6 serving output short protection are possibly not enough to protect 2SC5200
2AC5200 would allow for higher open loop cutoff , with TIP3055 it is about 10 kHz.

Hello everyone,

Finally I decided to build my own speaker, and realised how deep the rabbit hole goes. Hopefully I can find guidance here, as my main expertise is programming, not acoustics 🙂

Hello everyone !

Long story short, I'm starting the design of my future main monitors for my mastering studio.
I currently have PMC which I have totally redone crossover-wise (with a Linea-Research ASC48F), with a 12" Volt, the famous ATC SM75-150S and an old Vifa D27 as the tweeter.
I plan do upgrade to 4x Purifi PTT10 per side as a 2x2 array to control horizontal and mostly vertical coverage a bit lower than usual (targeting floor bounce @ 205 Hz in my studio, 2.7 meters listening distance). To go with that on the mids and highs, I will either go :

1) Bliesma M74B / Bliesma T34B if I can get a properlly integrated system with a minimum of 500 Hz crossover on the mids and recoup the 8x Purifi woofers cost by selling the PMC.
2) ATC SM75-150S / Bliesma T34B if I need that lower 380 Hz crossover, can't sell the PMC and will probably sell a kidney instead 🙂

Both options will go anyway in a big dual waveguide. I started modeling and "Akabeking" but I'm facing a big problem : my current midrange waveguide is 620mm wide right now, and that takes a bit of horsepower to compute a model which is not crap below 4K (doing it with 2 symmetries and only computing a quarter of the waveguide). Problem is, for high frequencies, it gets really really too big and my desktop CPU wants to hara-kiri if I want a good HF prediction.
Adding to that, when I'm merging both waveguides to do a dual one, I have to get rid of the vertical symmetry, and it basically triples the number of triangles in the mesh, making it even more impossible to modelize. And I'm in the infinite baffle scenario, monitors will be flush mounted.

Here are my current best with artifacts on a really big mid waveguide (shallow but very large) and a smaller tweeter waveguide (not merged together, so this is the "target" response).
I still need to continue to optimize the "slopes" both waveguides give.

With 380 Hz crossover, so keeping the ATC mid :


With 500 Hz crossover, so Bliesma is okay now. It has a slight 600 Hz bump in DI, mostly caused by the vertical off-axis response.


Mid/HF crossover was put at 1.2 CtC with the lowest driver distance possible between midrange & tweeter, i.e. 112,5 mm.

To avoid loosing too much time and CPU by experimentation, I feel it's better to develop a smaller, KH420 sized waveguide, learn to optimize the merging of both, before going back to the monster I plan for my speakers 🙂

I tried two ways of merging :
1) Flat "speaker baffle" with single big waveguide around :


2) Generate both single waveguides, intersect them and radius the transition :


Both method keep the horizontal dispersion almost the same. Method 2 seems smoother in the vertical axis.
So before going further, and since it will not really be for me at the end (even though the examples above are 32x32 cm, which is exactly the size of the midrange & tweeter plate on my PMC so I could try it in my studio), I need your input : what would serve the community best ?

1) Considering the pedigree of the drivers, what would be the most useful woofer size goal to match ? single 10", single 12" ?
2) T34B or T25B ?

Are there people here already owning these drivers and access to a CNC router to compare measurements to simulations ? I will buy the drivers at some point, but not right now.

Played, not soldered, electrically and visually in perfect condition.

Seas U22REX-P/SL 220Eur, bought few months ago
Scan Speak 15M/4531 200Eur

Photos: https://pkaudio.webnode.cz/forsale/

EU shipments only, wire or Paypal.

Good day,
Just bought a used Audison BitOne DSP that I want to use it to play around with some speaker prototypes. It has a digital input, but no internal ASRC, so the input signal must be 48kHz (and 24bits). Audison sells the SFC as well, a sampling frequency converter that takes anything from 32 to 192kHz and converts it to 48kHz https://audison.com/product/sfc/

My question. Do you know about similar products? I would rather have something with a USB input that converts all the different formats I listen to from Qobuz to the desired 48kHz / 24bit. I can get a USB to SPDIF, and then the Audison SFC, but maybe there are (cheaper) alternatives?

Many thanks for your input!
Erik

NOTE - It is recommended to read this thread for basic construction of the unit. Gary B has established a new thread that deals with adjustments and alternate parts selection that may influence your choices. It can be found HERE

This is a revised version of the initial posting on this thread. I contains a collection of general information, recommendations and methods leading to a successful base/stock build. Changes, clarifications and additions may occur over time. As with most projects, it is recommended members read through the entire thread before starting the project. Please PM me or post any significant information that would be helpful as project start-up items. Some photos/illustrations will be added for further clarification.

Suggested equipment:

0.7 mm or smaller 60/40 good quality tin/lead solder.
A small chisel tip around 2 mm on a good quality soldering iron.
Liquid, tacky or soft paste flux (ChipQuick recommended).
De-soldering braid.
Lighted magnifying lens/lamp for assembly and inspection. Example – Goose Neck Lamp

Small point tweezer, preferaby a type to which solder does not stick !

Suggested techniques:

Open SMD containers over the PCB and away from the edge of the work area.
Double check orientation/polarity (where it applies) before applying heat.
Tinning just one pad prior to attaching components appears best for adding minimal solder.
Solder just one tab/pin and confirm alignment on components with multiple leads.
With components that have multiple pins/leads please solder the diagonal pin secondly. Now you know for sure that the chip is aligned correctly and can't change it's position anymore.

Some helpful videos on surface mount soldering -

Professional techniques

diyAudio Project example




There is an error on the orientation of C7 on the PCB. The “+” is on the
wrong side.Our Jean-Paul and Subbu missed that single error and apologize for letting it pass undetected. (We say that's fine as long as everyone is aware of it 😉)

[/url][/IMG]

L1 has no polarity but since it has pin numbers we used a footprint
symbol to solder it correctly. You can solder it the other way around
though.

C-4 is a a dual use pad that accepts SMD or through hole. Refer to thread postings for more information.

On the orientation of Q-2 - the dot on the cover should be at the thick line.

Suggested part placement sequence: WM8804, ES9023, MIC regs, resistor array and then the 100 nF and 20 pF caps …..

Work ESD safe ! The chips are ESD sensitive devices. If you have no ESD strap make sure you discharge yourself by touching a PE connected device with both hands.
In Europe that will be for instance the heating radiator or the metal workbench in your kitchen (check local regulations on that). Anything that has a good connection to earth ground will do.
Don't wear any clothing with fabric that builds up static electricity. Wearing polyester containing clothing while wearing sneakers (isolation from ground while building up static electricity!) will present the possibility that chips will be damaged when the static electricity finds a path to discharge to.

**Try to not overheat parts.**

Solder everything without L3,4,5,6 in place. Check the board for solder blobs and shorts.
After all voltages after the regulators are confirmed, also place and solder the inductors.

Clean the PCB with isopropyl alcohol 99%, let it dry thoroughly - and then check carefully (again)for solder blobs and/or shorts.
Close your eyes, keep your fingers crossed and switch it on.

Confirm correct voltages again at L3,4,5 and 6.

[/url][/IMG]

When those voltages are confirmed still correct, then measure output DC voltage which should be only a few mV DC offset.

Now connect an SPDIF source (cdplayer, mediaplayer etc.). Also connect L + R outputs to your amp.

Open volume very slightly and press the play button on your SPDIF device. Slowly turn volume up and listen carefully if you hear non distorted audio.
Take a deep breath and open volume a little more. When you hear good sound please relax and experience the feeling of a winner.

Give the DAC a few hours to break in. You will notice sound quality improves over time.

There is a public Google doc (with pinouts) that can be helpful in reading and understanding important voltages along the circuit.

Subbu/JP DAC read sheet.

This composite has the correct information and should be used for proper orientation of Q2. Please see post #575 for more detail.





For builders who prefer screw terminals, the following are suggested as a quality option to the crimped style. See post #643 for details

2 pin Mouser
571-282834-2

4 pin Mouser
571-282834-4


2 pin Digi-Key
A98333-ND

4 pin Digi-Key
A98335-ND

*****
korbin69 has contributed these images (Post #911) that should be considered a standard for DAC board construction and correct voltages. Printing the photo and the read sheet from the Google docs can serve well as trouble shooting aids. The top-side photo shows how a properly stuffed and soldered build should appear. Please note the minimal amounts of solder requited.

Thanks Phil - a major assist highly appreciated by all.

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

Erwin

Joined the forum to learn as much as i can to build some OB speakers. Thanks in advance!!!

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?

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

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

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...

Hi thank you for the approval.I am into guitar amps and hi fi amps and would love to learn about repair

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; 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

Anyone have any idea what these are ?

Hi everyone

Has anyone played with the DEH610?

https://patric-sokoll.de/Röhrenmuseum/Datenbank/PDF/Datenblatt DEL 610-1 Marconi.pdf

https://www.radiomuseum.org/tubes/tube_deh610.html

may sound good?

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?

Nm

Looks like a fine triode...

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.

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.

I have a Beyma 12MI 100 as mid but I sommetimes find i hard to find focus inbetween in the middle of the speakers.
Would a 10MI100 bet better suitet as the mid?



The crossover is:110Hz and 1200Hz

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!

Hi thank you for the approval. Im Poppix from the Philippines. I am into guitar amps and would love to learn more and dive into diy sound systems

deep within email

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.

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

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.

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.

Hello all,

I am Xerxes. I am interested in designs by Nelson Pass and other designs that are super simple and sound great.

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!

Luxman M6000 (sold as non working)

Interesting. This 50 year old amplifier is still worth $$$$ in working condition?

Regards,
Dan

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!

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!

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!