Hello all, having trouble finding this information. I recently recapped my HK 12. I am struggling to find the correct bias settings for it? I see the test points, but... Anyone point me in the right direction? TY!

Hi guys I just picked up a Starkit 12-22a tube tester and it needs a meter , does anyone know of a company that sell something that may work . (500au ) it's a real nice tube tester and would hate to scrap it for a simple meter .I tried using it with a volt meter attached to the meter outputs and it seems to be working .. thanks for any help .. cheers Roger

Does anyone have a component map for the phono boards in a 1000A? I need something that has the circuit boards printed out with designators on it for the parts. Also the 7 or 8 IO tabs on the bottom of the boards. What each leg is and where it goes. I could map it out the hard way but was hoping someone may have already done the hard work already. Also I am looking for a small circuit I can replace the stock circuit with. Preferably with improved performance.
Thanks for the help. It would be nice if Sansui had maps of the boards like most of the other brands do.

Hello,
I am trying to build a cabinet for my guitar amp. I have build other cabinets but this time is different and i need some help. I need to build a ported cabinet, like the tl806 but the speaker i have is not the famous electrovoice el12.
I m going to use a jensen tornado classic 100 8 ohm istead.

I have used an online calculator to get me in the ballpark, here the results


Now, my doubts, the cabinet volume suggested is huge, 126 liters it is going to be oversized for guitar amp 1x12.
Do you think these results can be trusted or should i use a better source?
I know the tl806 cab design has slots and not pipes for the bass reflex but i don t care, i m not trying to clone the design, just the concept.

Just thought I'd toss the question out there... about to populate a bunch of boards with resistors and caps from Mouser, and I'm holding off until my DER EE DE-5000 arrives so I can check each one before populating. Doing that would also make sure none of the parts are faulty.

I don't think I'm going to change my mind about that but I'd like to know what other people do/think.

thread index at the end of this post!

When planning, simulating, building and measuring bass reflex enclosures for 2-way speakers one difficulty is dealing with port resonances in the midwoofer’s passband.

I thought it should be possible to absorb resonances with Helmholtz resonators in/at the bass reflex port.
Obviously I am not the first to come up with this idea, but I also didn’t find very much about it on the internet.
Thus I decided to make my own measurements, see the following posts!

some web references:​

Short thread about this issue on diyaudio. note that the thread starter wrongly refers to the "1/4 wavelength resonance" of a port. This is not correct! The fundamental resonance of an open tube happens at the frequency that has a wavelength equal double the tube length.

A german thread about port tube resonators.


various web references for resonance of open tubes and helmholtz resonators:

https://en.wikipedia.org/wiki/Helmholtz_resonance

https://en.wikipedia.org/wiki/Acoustic_resonance#Open_at_both_ends

https://www.physicsclassroom.com/class/sound/Lesson-5/Open-End-Air-Columns

https://en.wikipedia.org/wiki/End_correction

http://troelsgravesen.dk/vent_tuning.htm (including end correction values!)

calculating resonance of bass-reflex port (open tube resonance):​

occurs at frequency that has half wavelength equal to end corrected reflex port length and respective frequency multiples.
Oscillation node (pressure maximum) is at center of port (or further divisions for multiples).
pressure absorber works best at the respective node.

Option 1: tube absorber​

Simple tube, one end closed, it should thus have roughly half the length of port for absorbing resonance of port fundamental; can either be parallel (and inside) the port or 90° perpendicularly mounted at port. Open end should be located at half port length.

Option 2: Helmholtz absorber​

Mounted at half bass reflex port length, connected by its own small Helmholtz port to the bass reflex port

-------------------------------------------------------------------------------------------------

Thread index​

#3 - first resonance absorber tests with tube test setup
#5 - decay plots of test bass reflex box
#8 - resonance absorber results by @augerpro
#11 - link to a relating message by @rdf including link to roozen/philips-paper
#19 - new parallel 6th order bandpass test box for further investigations
#22 - dealing with enclosure resonances
#25 - chuffing audibility test
#42 - particle photos (using water sprayer)
#48 - port 2 variants and measurement results
#50 - port stalling test
#54+55 - influence of smoothed port flange
#56 - port 2 impedance results
#58 - port resonance absorber test
#81 - link to bryce doll paper by @Hearinspace
#103 - port 2 variants resulting in the same tuning frequency, with geometry drawing
#104 - photo of port 1 variants
#141 - port wall stiffening influence
#157 - port 1 variants response measurements
#159 - chuffing RTA measurement of straight port 1
#161 - chuffing RTA measurement of rectangular curved wall port and big 3d printed port
#165 - chuffing RTA measurement of flat port and small straight port
#166 - resonance absorber setup
#167 - measurment of resonance absorber lenght variations
#202 - resonance absorber filled with melamine foam
#206 - chuffing RTA measurement of 3d printed port with resonance absorber
#214 - influence of port diameter/dimension for enclosure resonance transmission
#228 - small 3d printed port drawing and measurments
#230 - testing noise of roughened port surface
#236 - port with resonance absorber documentation by @Kwesi
#249 - chuffing RTA measurements in singe dB steps (small 3d printed port)
#251 - separating resonance absorber from the port with latex membrane
#288 - link to salvatti devatier button paper
#302 - explanation of boundary layer thickness by @andy19191
#310 - impedance measurements for port 1 variants at different input levels
#320 - "progressive port geometry" concept
#324 - progressive geometry port measurements
#328 - progressive geometry port length correction
#329 - progressive geometry port impedance measurements at different levels
#330 - progressive geometry port response
#331 - progressive geometry port chuffing RTA measurement at 100 Hz
#341 - output level measurement / comparison
#349 - is airspeed the main chuffing factor? progressive port chuffing RTA measurement at 50 Hz
#350 - new particle photos
#352 - explanation of turbulent air motion by @andy19191
#363 - relating chuffing to particle displacement
#367 - first GIF-excepts of 120 fps particle videos
#390 - port variants with sound recordings by @Tenson
#391 - correct port particle displacement calculation formula by @David McBean
#395 - (1) first slow motion video: hard edge port
#407 - (2) second slow motion video: flared port
#434 - flange variant video
#438 - (3) third slow motion video: progressive geometry port
#448 - influence of air particle displacement for the excitement of low frequency noise
#449 - differences of a big and small (optimized) port for a small 2-way midwoofer
#460 - (4) fourth slow motion video: small straight hard-edge port
#468 - small very flat port - checking for boundary layer flow resistance
#503 - how to calculate the strouhal number using max port air velocity data, CORRECTION see #591
#504 - water tank simulation
#522 - @Dmitrij_S Karlson couple port
#526 - flared port geometry definition
#529 - 2 way speaeker test ports to be measured ...
#531 - ... response measurements ...
#546 - ... chuffing measurements (RTA) ...
#591 - correct calculation of strouhal number, related to peak-displacement instead of p-p- displacement (thanks to by david mc bean)
#595 - how to find a suitable port exit diameter using the Strouhal number (corrected version!)
#616 - first beta version of a simple excel sheet for tuning calculation of NFR=0.5 ports
#621 - relating the strouhal number to compression and distorsion
#624 - parametric fusion 360 model for a NFR=0.5 port by @augerpro
#627 - augerport with resonance absorber by @augerpro
#646 - parametric NFR=0.5 port model for freeCAD, including STEP and STL model
#652 - announcement by @David McBean: stouhal curve included in the port exit air velcity graph.
#654 - how I make my 3d printed ports
#680 - chuffing/noise audibility comparison: straight and flared port
#704 - big collection and evaluation of 3 speaker and 14 port variants - proposed MID PORT STROUHAL NUMBER (MPSN) parameter
#747 - updated excel calculation tool with correction and variable NFR ...
#748 - ... and the imperial version of the same calculator
#753 - comparison ports vs passive radiators
#811 - compressing the optimum geometry definition to two relevant parameters
#812 - The breakthrough: Finding compression behaviour patterns
#847 - Iterative port geometry optimizer tool


(will be updated)

This Error-Correction/Composite Combo amp is based on Mr Evil "Unnamed feedback method explored" and Nickolay Shvydky "My ZD-50 ultralow distortion chipamp"

Output power at +/-29.5V: 8 ohms clip at 42W, 4 ohm clip at 65W. It is very stable in both EC and Comp mode. Ans it behaves very well after clipping.

I could only measure its THD+N very roughly using Focusrite Solo 3rd, which has a minimal line-out and line-in THD+N of 0.002%.

U1 is OPA1655 and U2 is OPA828. For error-correction(EC) mode, RV1 was set at 330ohm. For composite (Comp) mode, RV1 was set at 680 ohms.


Final Remarks

Today marks the completion of my 6xLM3886 PBTL (Parallel Bridge-Tied Load) amplifier, now fully assembled and installed in its case. Hearing music flow from the amp for the first time was an incredibly rewarding moment.

This has been the most time- and resource-intensive DIY amplifier project I've undertaken. It all began with a spark of curiosity after discovering HAYK’s thread, which then led me to explore designs by Nickolay Shvydky (ZD-50), Mr. Evil’s EC-correction, and Tom’s Modulus series.

I initially started with the ZD-50, but its complex compensation network proved daunting. I decided to pursue a simplified approach. After many hours with LTSpice, I developed a streamlined version of the design, which is shared below.


The first prototype used through-hole components and served as both a learning and testing platform. Getting the amplifier stable took significant effort, with a variety of issues encountered and resolved along the way.


Next came the 2xLM3886 parallel version, which helped me better understand the parallel operation of the LM3886. It also served as a base for further schematic refinement and testing.


Throughout development, I upgraded my test equipment for greater precision, allowing for more accurate measurements and analysis.


The final goal was this 6xLM3886 PBTL amplifier. With the knowledge gained from the earlier versions, I was able to optimize the circuit design, select key component values carefully, and fine-tune the PCB layout. The result is a high-performance amplifier with impressive specifications:

• Power Supply: Hammond 1182T24 625VA transformer (24VAC/13A dual secondary), DIYAudio Store Universal Power Supply board, and 140,000µF of filter capacitance
• Output Power:
  • 175 W into 8 ohms @ THD+N 0.00023%
  • 240 W into 4 ohms @ THD+N 0.00042%
• Residual hum/noise floor: ~10 µV

(Note: These measurements were taken using amateur-grade equipment under informal conditions. They are provided for reference only and should not be interpreted as certified or industry-standard specifications.)


I’ll now spend some time listening to the amplifier and enjoying the results. Meanwhile, I plan to revisit the earlier versions and design optimized PCBs for them as well.

(I made 5 sets PCB, minimal number of ordered sets, of the amp, there are 1 sets left for sale, see swap meat)

2xLM3886 EC/Comp
+/-32V 80W 4ohm, THD+N<0.0006%, Noise 10uV.

Hello, All!
Just joined today as I found a very useful guide here on using Hypex Filter Design, thanks to DannerD3H.

I have always been a music and audio fan and find acoustics and speakers particularly fascinating. I always wanted to design and build my own speakers but never got to it, mainly due to lack of space to use for woodworking (I live in a small flat). But still decided to go for it. I will work around the difficulties somehow. If there is a will, there is a way.

I have reasonable understanding of audio, electronics and design process and ok to use my hands, but I have almost no experience, so face a steep learning curve.

First thing I did is buy the Loudspeaker Design Cookbook. Reading it now. Learning curve.

I have a pair of old slim floorstaders (description to follow) that I intend to convert to actives. This will help to initially avoid much woodwork, while still learning about drivers, crossovers and amps and box interaction. This will be done in two steps:

Step 1 - add Hypex Fusion amp and use existing drivers and enclosure as is

Step 2 - replace the drivers for better specced ones, modify the box

This will hopefully take me to a future point of fully building larger speakers and maybe a sub. One day.

So far I have bought a Dayton UMM6 mic, a Fosi V3 amp and just about managed a few measurements in REW of the speakers in passive form and of individual drivers. I managed to install on old version of HFD software on an old Win XP (!) laptop and hope to use that to program the FA253s when they arrive later this week.

I do stumble a lot and have lots of questions and definitely need some help. I will be asking here and would appreciate any help from the more experienced members.

I will make a post with the existing speaker details and maybe a few other things I am struggling with.

FWIW I do have some forum history on a couple of HiFi forums: Hifiwigwam and Audiosciencereview.

So, wish me luck.

Hello. I'm using a Jab5 4x100 dsp amp board on a project. The SQ is acceptable for the use even if there is some background tweeter hiss. The fan noise is the real issue. I'm wondering if anyone has had any long term experience running the board with the fan disconnected and/or removed. Has it lasted without the additional heat dissipation? Also contemplating adding a resistor to reduce the fan output.

In the fall of 2023 my family relocated to North Western Connecticut which is an area that is a bit more rural than we had experienced previously.

One of the daily joys of living in this new area has been the almost daily sitings of wildlife…everything seemingly wanders into our yard when we least expect it…deer, bears, coyotes, hawks, eagles.

One of our favorite almost daily sitings is the Fantastic Mr. Fox who has set up residence on the edge of our property the past two years with Mrs. Fox and produced a brood of 4 pups both years.

The pups usually start venturing out some time in April. Most likely mom is trying to give them some practice at hunting mice for themselves. They are seen playing in and around the pool and rocky bank towards the bottom of our property.

I’m lousy on the draw with my phone camera. I’m sure some of you guys are much better.

Post a pic of some of your furry wild friends that you caught a glimpse of.

Looks like it's That Time Again, where I make a semi-educated guess about a new Danley invention.

Been at this for a while, here are my previous threads:

1) This one didn't get a lot of attention, but my thread on the Genesis horns provided a TON of information on the Jericho Horns, the Paraline, and the midranges used in both Danley Sound Labs and Sound Physics Labs speakers: https://www.diyaudio.com/community/threads/i-dont-understand.133745/

2) Here's the OG thread on making Unity Horns: https://www.diyaudio.com/community/...ge-cone-for-bandpass-mid-in-unity-horn.88237/ thanks to @Puggie and @GM for getting that one going.

3) Here's the "main" Paraline thread: https://www.diyaudio.com/community/threads/square-pegs.217298/ And the secondary one: https://www.diyaudio.com/community/threads/stargate.225832/

4) Here's my reverse engineering of the Danley Matterhorn: https://www.diyaudio.com/community/threads/old-school-horn-vs-modern-vented-box.250032/post-3794298

Friday evening chez Weldon:

Start with a warm winter evening and a bit of a sunset
Add a glass of red, a shot of hooch
and a couple Dungeness

I would like the part numbers of blown resistors I think they are 4.7ohms but diodes D8 & D10. IC9 & IC14

Hello,
In a world of engineers i'm a tad shocked to don't see any good decoupling, the mecanical engineers are not all dead since i've recentely be teached by one of them and my last school teacher was also an engenneer (with a PhD in mecanics)... he was young 25 years ago.
People are using CAD and FDM printers here and no one seems able to decouple convenientely a loudspeaker, could you say what i don't understand ?

Aliexpress is full of strange spikes and ridiculous spring systems, are we all idiots ?

Here is a SMR for a microphone :

When I was young, we spent hours on end playing what we called word game. Same rules but it was always 4 letter words.
I was introduced recently to this new sensation and tried it today. Now I know it's beginners luck, but first time out I got it in 3 words.
I think I'll try again and let it bring me back to earth.
Who else is doing this?

This is a pair of Apex XRK Audio modified FH9 boards, Dual mono blocks that run in AB
50 watts 8 ohms they are dead quiet and sound fantastic
The amp uses 2 of the Prasi CRC power supply units,
Includes both boards the Heatsink 2 Antek 300va 25v Transformers
Prasi PSU CRC Boards
Which can also be used for Class A amp boards as well.
Use them as is on Baltic Birch Board,
Or put them in a nice enclosure and you have a fantastic sounding amp ready to go.
$260 for all local pickup only.

Hello EL34 Baby Huey builders,

As suggested by SCD in the following thread : http://www.diyaudio.com/forums/tubes-valves/72536-el84-amp-baby-huey-217.html#post5533779 I am starting a new thread specially dedicated to the building of Baby Huey amplifier based on my PCB which has been produced in more than 330 pieces and is still requested by many tube amplifier fans in a possible future Group Buy 4 : http://www.diyaudio.com/forums/group-buys/312869-gb-baby-huey-pcb-42.html 🙂

I will put in this thread all the information available about the assembly of the EL34 Baby Huey amplifier and some of the auxiliary board that I have used or designed to add some features to the amplifier. I must tell you that I am not in professional amplifier business, this is just a time consuming hobby for me and, as you will see, English is not my first language and I have sometimes difficulties to explain things correctly, sorry for that but at least you will understand easily schematics, pictures, B.O.M., etc... 😀

History

All that started about two years ago when I read the http://www.diyaudio.com/forums/tubes-valves/72536-el84-amp-baby-huey-217.html#post5533779 post from "gingertube" ! As I was interested by the shunt feedback concept that I didn't know I wanted to test it and decided to build the simplest version with two PCL86 tubes. Since I am very bad in point to point wiring, I made a small PCB. This amplifier was so good that I decided to continue my quest and, since the ECL86 or PCL86 were unfortunately no more produced, I made a second PCB for the EL84 version which needed a third tube, a 12AX7 (ECC83) for the input stage and added MOSFET with current source to drive the output tubes. This amplifier was very good and I was very satisfied by the MOSFET driver stage, but I was also a little bit frustrated by the limitation in power, about 12 W, and the fact that I could not use different power tubes...

To solve this problem, I made an identical version but with octal socket to accept many compatible tubes from the 6V6 to the KT88 and the EL34 Baby Huey was born ! Following the presentation of this solution on the diyaudio forum some members asked me if I could sell the PCB, I had never done this before but since the moderator move this discussion to the Group Buy forum I have received several request and started a GB1 with a target of 50 boards but finally closed it with 80 PCB ordered 🙂 I had to send them Worldwide and to learn how to make PayPal invoice... Later others readers asked me for more board and I offered a second and "last" GB2 which finished with 120 boards ordered !!!

Later, since there was still a strong demand, I asked Prasi, to whom I have bought a CRC power supply PCB, if he would accept to manage a third GB and he kindly accepted 🙂 He also suggested to make 2.4 mm boards for better rigidity. Now 120 more boards have been ordered and are in production and there is already 20 PCB's in waiting list for a possible GB4 ?

Documentation

In this first post I will enclose some of the latest documents available for building the EL34 Baby Huey and I will add more in next posts depending of your questions.

You will find the updated BOM in the GB letter, the latest schematics, some building recommendations, how to connect and to adjust the amplifier and several pictures showing the PCB assembled on top side (big components go on bottom side), a 6CA7 board fully assembled in test and finally the EL34 Baby Huey in its enclosure nearly finished (two vu-meters are still missing).

More pictures and data will come from all the people who have built this nice small amplifier. It is a very nice project for those who have never made and listen a tube amplifier, but remember there is some lethal high voltage (up to 500 V) and you should be very careful when you work on it

Have a nice time building the EL34 Baby Huey...

Best regards,
Marc

Hi all, here in Europe there is at least one shop where i could pull the spec sheet

https://doc.soundimports.nl/pdf/brands/purifi/PTT1.3T04-HAG-01/PTT13T-HAG-01-Datasheet.pdf

The application information is good, never seen something so good explained from other vendors



So far - so good, Stefano

P.S.: this is the hint that VituixCAD is becoming a quasi standard even in the industry

I desperately need the schematic for PS Audio PS IV pre amp.

I already called PS Audio tech support and was told that they no longer have any info on discontinued products since the company was sold. The owner's manual is available on line.

The regulator is blown. I need to know the rail voltages.

Any help would be greatly appreciated.

Art

Sorry -- another basic question...

I was adjusting a pair of these for 24V/1.2A when I heard a tiny "tick" somewhere. After the "tick" I've now got no voltage at the "1/2 voltage (KT1 adjusted)" transistor.

I assume I blew something. Nothing visually changed anywhere (no scorching or blown caps). Replacing both trimpots on one of the boards didn't solve the problem.

Any ideas of what happened? Can I test items installed in the circuit with a multimeter? If I can, any suggestions what to check?

I vaguely recall a few years ago reading something that I found on the web that discussed placing the dipole woofers in the corner. I'd like to read it again but can't remember what it was called or where it was. I recall thinking it was a decent paper and had outlined diagrams showing the woofers placed near the corner such that from the listening position you're looking at the side of the woofer.

I tried placing my slob in the corner this evening and the bass seems easier on the ear drums but as expected needs much more gain; digital xo 100Hz.

I've been working on this for about a year and still incomplete but almost done. I couldn't hold myself to post some first pics.



Actually, it has been quite some time I'm fooling around low cost soundcards trying to modify them for improved performance. What strikes me is that although manufacturers don't hesitate to use top class chips, the overall performance is compromised by design. For example the power supply even if it's not the USB Vbus, it will be just a small external wall wart. Another problem is the necessity for a user friendly front panel which leads to a messy PCB layout. And then it comes the copyright war forcing companies to adopt proprietary solutions especially in the software domain, causing even instability issues in some cases. Most of the time I came to think what if app notes were followed to the letter but this would take a complete redesign from scratch. Eventually it happened.🙂



I rush to catch up any queries. This is not meant to be a giant killer. In fact, I deliberately stayed within limited bandwidth and dynamic range to avoid meeting spooky shadows. The target was the low to mid cost commercial soundcards. That said, it costs more than a Focusrite Scarlet to build 😱 and one might think that it doesn't manage any better. But all these products need some kind of input protection and usually adding that will also add THD+N. So, my soundcard is ready to accept an amplifiers output, the loop measurement are taken on these inputs and the total cost is much less than a soundcard-input interface combo. And it's mine, which is priceless...


About specifications: Two channels -stereo- full balanced I/O. Input impedance 100k. Input sensitivity for 0dBFS 1,5Vrms differential. Input protection for 400VDC. Also it should be OK for 100Vp AC but I haven't tested that. Output can drive most usual loads with a max 2,6Vrms differential. Full duplex up to 192kHz/24bit, but usable only at 48kHz with ARTA, REW and the alike.


Still on the bench, I expect a little more improvement when I put it in the box. I verified that at least for the 50Hz noise. Lock down delays but it shouldn't take long more. I'll share all schematics and gerber files for anyone who needs something like that and can build it.

Hi all!

I'm working on a "stomp box" for microphone, not guitar, so the input is a balanced XLR (dynamic mic) and the output should be suitable for a mixer/PA (balanced output).

The circuit is based around a PT2399, so I need to take out a single-ended signal using an INA333. My design relies on op amps (TLV2462) to amplify the signal etc. All these ICs run on 5 V single power supply, so I need to add a 2.5 V DC offset to signal path.

I am currently focusing on the input and output stages, how to hook up to XLR contacts and feed into the INA333. Similarily, on the output, how to connect the balanced signal (which I create using op amps). Please review my design as shown in the images.

Are the coupling caps on the input stage oriented the right way, with cathode towards the mic? Also, I believe I should have a 48K to ground on the output (just before the XLR contacts) but I should not have this ground on the input stage. The 48K to ground on the output is to stabilize the potential at the output, right? On the input, the potential will be stabilized by the "bias" resistors which add the 2.5 V offset to the signal, so I don't need the ground connection (also, it would create an unintended voltage divider) so these should be omitted.

Input stage


Output stage


I guess I also wonder if it makes any difference if the op amps on the output (which create the balanced signal) are in series, or in parallel as in my design?

Hi, I am hoping someone can help me with this amp, I have spent a lot of time looking for the problem.
-I started this project and when I needed to buy the mosfot, (Hitachi 2SJ50, 2SK135). I found they were off sale. Then I found that Profusion sold replacement models, (ECF10N20 and ECF10P20), I bought them and mounted them but at the output it had a noise as if it was AC. And that is that it was oscillating. The oscilloscope shows a frequency of 8-9Mhz, with nothing at the input.
-I have been reading in this forum that these mosfets have been mounted as Hitachi and other replacements, and I have seen that the schematic may need some modification, like adding some capacitor or changing some resistor. But my knowledge does not reach this.
- With help, I changed the capacitors C6 and C7, from 33pf to 180pf, but it was not enough, the AC noise at the output is attenuated, but it is still oscillating and it is impossible to regulate the Offset.
-This is why I am asking if someone can help me to get it to work with these Exicon mosfets, because with Hitachi mosfets this schematic work well.


Also, I am looking on this website for alternatives in case I don't find a solution to my project, and use part of the main components, such as Mosfets and power supply.Only I found this proyect with ECF10N20 & 10P20 mosfet mounted. https://www.diyaudio.com/community/threads/elektor-axl-v2-with-lateral-mosfets.332535/
But I don't know if this project could work with double pair of mosfet and power supply could be 50V.

Thank you so much.

Hi All,

I have a stock of new transistores for sale bought from Mouser and Farnell some years ago:

108 -> 50 x KSA1220AY (Fairchild/On Semi)
108 -> 50 x KSC2690AY (Fairchild/On Semi)

160 V / 1.2 A / 20 W / TO-126 / (highest grade hFE = 160-360).

Price per pair with a minimum qty of 20 pieces: 1.5 Euros/pair + Shipping.

Matching available. Please PM.

Regards,
VS

From a thread I've been following, it appears there may be interest in discussing all things whisky (or whiskey for American and Irish variations). What's your favourite, how do you take it, where and when, why even? Excursions into other liquors are tolerated.

Another wonderful design from Mr Fujita - TU-8550​

You can view and listen TU-8550 PREAMPLIFER in the coming - ​

Tube Preamp Kit TU-8550
PRICE : $1,175.00
JJ 12AX7 X 4
JJ 12AU7 X 2








[Circuit configurations, stock tubes]
Flat amp (line amp) : 2x 12AX7, 1x 12AU7, SE input/output
Phono equalizer amp : 2x 12AX7, 1x 12AU7, CR type, RIAA equalizer amp
A-power : DC-power (all tubes)
B-power : MOSFET, slow start, constant voltage power
* Line AMP *
Unbalanced output, continuously variable gain
[Max. output voltage] (1kHz)
Over 30V rms (Volume at the max.)
[Max. gain] (Volume at the max.)
Approx. 17dB (approx. 7 times)
[Frequency response] (-3dB, 47kΩ load) 5Hz to 200kHz
[Residual noise] (IHF-A)
15μV (Volume at the max.)
6.5μV (Volume at the middle)
4.3μV (Volume at the min.)
[SN ratio (IHF-A)]
130dB (Volume at the middle)
126dB (Volume at the max.)
* Equalizer amp *
[Max. output voltage] (1kHz) 20V rms (1kHz)
[Gain (1kHz)] 29dB
[SN ratio(IHF-A)] 112dB
[RIAA equalization] Within ±0.5dB (20Hz to 20kHz)
[Input resistance] 47kΩ
* General *
[Input/output terminal / Input and output impedance]
PHONO IN：RCA jack, for MM cartridge, 47kΩ
LINE IN： 4 lines
LINE 1, 2, 3 : RCA jack, over 50kΩ)
LINE 4 : 3.5mm stereo phone jack (front), approx.12kΩ)
PRE OUT：RCA jack, 2 lines (parallel output), approx.1.1kΩ
REC OUT：RCA jack, 1 line, approx.2.4kΩ for PHONO
Output impedance of the sound source + 1kΩfor LINE IN
[Power voltage] AC 100V 50/60Hz
(For overseas, 115V(110 –120V) or 230V(220 –240V), 50/60Hz)
IEC standard, 3P inlet type
[Rated power consumption] 23W
[Dimensions] W352 x H66 x D298 mm (including projections)
[Weight] Approx.4.0kg (Assembled, excluding AC cord)







4X VCAP 0.47 CuTF (option )

Wanted To Buy

Signalyst DSC3​

I need try it..

So I kick off early today like many are doing during this C19 crisis and I thought I'd do a little housework before the Mrs. came home. I started making the bed and I noticed a pillow was missing. I look around and see it on the floor at the end of the bed. Only problem was, it was moving. It seems that little Bingo was playing hide and seek with Daddy and had now trapped himself inside. Lifted him onto the bed, freed my little prisoner and went onto finish my work. It was funny enough that I thought I'd share because I think we can all use a little something to help us smile these days.
Cheers.

Hi, All

Announce some SMPS+Class D amplifier integrated modules here:

FFA001V3
Output: 2x150W @ 4 / 8ohm
Input: 85--265Vac, constant power SMPS
Size: 90x150mm
Price: $49.9/Unit, w/o shipping (Inc. I/O buffer and mating cables)

FFA001V3-HB
Note: Configured as a high-bass mode. CH1=300W@4OHM and CH2=120W@8OHM, similar to the Pascal U-PRO2
Price: $54.9/Unit, w/o shipping (Inc. I/O buffer and mating cables)

FFA002V2 (1U height)
Output: 2x300W @ 4 / 8ohm
Input: 85--265Vac, constant power SMPS
Size: 95x200mm
Price: $99.9/Unit, w/o shipping (Inc. I/O buffer and mating cables)

FFA003V1
Output: 1x150W @8Ohm / 1x300W @4ohm
Input: 85--265Vac, constant power SMPS, heavy-duty design
Size: 90x245mm
Price: $64.9/Unit (Inc. AL base plate and Volume IO buffer) w/o shipping


PM me if you need more information.
Thanks~

Pics of FFA001V3












Pics of FFA002V2

Pics of FFA003V1



Misc
--- FFA001_V3 spec sheet & Test data;

--- FFA001_V3 & I/O_V1&I/O_V2 dxf file;

--- FFA001_V3 RearPanel_XLR 3D drawings;

--- FFA001_V3 RearPanel_RCA drawing and PDF;

I was told this design could cause scratchiness in the pot because of loading and after about 4 months of use it seems that may be the case.
I'm seeking advice on two changes I was considering.
Replace the conductive plastic Alps RK27 (from Ebay so probably a knockoff) with this Vishay cermet, and swapping out the LM4562 with a FET input OPA2134 to reduce the pot load. I know an OPA1642 would be a better choice but it's not available in DIP8.
Output stage shown.
VR4 is 50K so R112/212 is 6k8 and R113/213 is 15k, per Rod's instructions.

Hello everyone Can you tell me what material the Mark levinson 30.6 assembly board is made of? Arlon?

After months of research on Multiple Entry Horns, I’ve decided to build my own. While there are some excellent designs available, I’ve decided to design my own due to size constraints and budget limitations. My goal is to build a complete system for under €2000, including drivers, amplifiers, DSP, materials etc.

I’ll be documenting the journey and sharing the lessons I learn along the way. Since I’m new to this, I’d greatly appreciate any feedback or suggestions for improvement.

For those unfamiliar with Multiple Entry Horns (also known as Synergy or Unity Horns), I’ll include some resources below. In this post, I’ll refer to them as MEHs.

Why Synergy horns?
Synergy Calc V5
Synergy Patent
SynTripP: 2-way 2-part Virtual Single Point Source Horn | diyAudio
Scott Hinson’s MEH

The Design
Designing this project involved plenty of trial and error with drivers, designs, and Hornresp parameters. First I was going to build a 3-way MEH using four 4” drivers and two 10” drivers per horn + big subs using 15” drivers. However, when I discussed this with my better half, she wasn’t happy with the big size. So I changed it to a more compact design: 30x30cm 2-way MEHs paired with subwoofers.

For the 3-way MEH I got ideas and insights from Syn 9/Syn 10 and Cosynes, Scott Hinson’s MEH and SynTripPs. After deciding to go with a smaller horn, I began exploring more compact designs, such as: Two way synergy Horn, Portable Battery Powered MEH Build and others.

Size
Using Synergy Calc , I calculated the size and parameters for Hornresp. According to Bill Waslo: “with horns, bigger is better!”. While I would have preferred larger dimensions, I had to consider WAF, so I settled on a 30x30cm horn with an 80° coverage angle and horizontal pattern control to 850 Hz. After reading this discussion I changed the S1 value to 5.06 to match the CDs opening area to get a more precise Hornresp Simulation..



Drivers
Hornresp is a fantastic tool for modeling speakers, big thanks to David McBean for his contributions! After countless evenings testing various drivers and parameters, I settled on the following components:

Compression Driver - FaitalPRO HF108 (8Ω) I chose it because it works well in short horns and wide dispersion waveguides and people seem to like it a lot. The recommended crossover is 1300 Hz, which is what I’ll go with in my design.
Midrange - 2x FaitalPRO 4FE35 (8Ω) in parallel - they simulated well in Hornresp and have been used in some MEH builds before.

Hornresp parameters



3d printed horn
The inner section of the horn (shown in light grey in the 3D rendering) will be 3D-printed, while the outer flare will be constructed from 2 sheets of 19mm MDF with a dispersion angle of 130 degrees

The HF108 compression driver has a 31-degree exit angle, so the horn starts with a matching 31-degree angle and gradually transitions to an 80-degree angle at the midrange entry point.

To minimize unexpected cancellation notches, the inner corners of the horn are smoothed with 2mm rounded edges.





Mid driver Port location:
From Why Synergy horns?:

In a synergy horn we put the tweeter at the horn apex, easy enough, but how do we know where to tap in the midrange and bass drivers. We use the rule of 1/4 wavelength. So if we want to cross over from the midrange to the tweeter at 1200 Hz, we would have to tap in at 340/1200/4= 7 cm (2,8"). At the same time the cross sectional area (CSA) at the tap in point with in the horn. should be no bigger than in circumference, than the highest frequency being used in the bandpass. So at 1200 Hz the wavelength is 28,3 cm, so CSA can be no bigger than 28,3 cm, otherwise the hornwalls will no support the frequency. Same thing applies for the bass drivers. If we want to x-over from bass to midrange at 400 Hz, then the axial distance from the apex to tap in is 340/400/4= 21,2 cm (8,3"). This 1/4 wave rule makes sure that the drivers bandpass, is cut off and basically acoustically self terminate. What happens is that the frequency at play, for instance at 1200 Hz, travel toward the apex of the horn and then back again, but this time, 180 degrees out of phase, so a cancellation notch occur. In this way we can acoustically short circuit the bandpass, smart. What this cancellation notch will also do, is to acoustically lower the harmonic distortion above the cancellation notch. This effect can NOT be done electrically ONLY acoustically. This cancellation notch can be as big as -30 dB, so harmonic distortion is also lowered -30 dB. The result is a much cleaner sound, compared to other speaker designs.

Click to expand...

I’m crossing between the mid and the compression driver (CD) at 1300 Hz, which means the port holes for the mid drivers should be within 6.596 cm (calculated as 343/1300/4) of the CD’s exit.

I’ve been wondering if the critical distance is measured from the mid ports to the CD’s acoustic diaphragm or its exit. The CD’s exit seems to work well in previous designs posted here, so I decided to go with that. This approach also simplifies placement by allowing the ports to be positioned farther out.

To ensure I stayed within the critical distance, I decided to shorten L12 slightly (the distance between the CD’s exit and the plane of the mid-entry ports). L12 is 4.28 cm, meaning the centre of the mid-entry port is 5.84 cm from the centre of the CD’s exit, so it’s well within the required 6.596 cm for the crossover.

The circumference of the cross-sectional area at the tap-in point is calculated as 6.531*4=26.124, which is close to the wavelength of 1300 Hz.

However, when I model it in Hornresp, the cancellation notch appears well above 1300 Hz, and the mids start to roll off around 2000 Hz. I can increase the distances well above 6.6 cm before I get the cancelation notch near 1300 Hz. I’m not sure if I’ve missed something in my calculations and the measurements may differ when I measure the build speakers.

Mid driver Vrc:
I made the Vrc (closed rear chamber volume) fairly small 2 L in total (1L per driver). I still haven’t figured out how I will limit the chamber volume. Using tubes like those in the Cosynes would be challenging due to the distance between the mids and the compression driver.

Mid driver Vtc:
I didn’t feel like I needed to minimize the Vtc (throat chamber volume) as the mids can play high enough frequency already. Making it smaller using cone plugs seems more beneficial with bigger woofers (like on the SynTripPs) and also the B&C 4NDF34 which have been used in some designs. For now I’ve estimated 50cc per driver, but will measure the volume when I receive them. I might adjust the sizing after doing the final calculations with the correct volume.

Mid driver port size:
There are 2 entry ports per driver, 4 in total. The entry points are frustrumised, 2.76 cm in diameter closest to the driver and 1.95 on the inside of the horn, which translate to Ap1 of 24 cm² and Ap2 of 12 cm².

The ports are relatively small, with port velocity exceeding 17 m/s when played above 105 dB below 200 Hz. These are meant for home hi-fi use, so I don’t plan to play them too loud. I also don’t want to make them too big, so they affect the CD’s output. The ports are placed next to the corners to minimise the effect they have on the CD.

Vertical drivers:
This is something I haven’t seen mentioned often, and it seems to be overlooked in many MEH designs. I will place all the drivers vertically, on the side walls of the horn, to avoid driver sag.

2. All drivers needed to be nearly vertical. The suspensions of a lot of home drivers will take a set over time if stored horizontal...I've seen it way too often. Since I'm using relatively inexpensive drivers I wanted to avoid this since I don't plan on building myself new speakers every 5 years or so. Source

Click to expand...

Ported or sealed box:
While I was designing the 3-way MEH with 2x 10” woofers I tried modeling the low drivers in both ported and sealed box. I ended up choosing sealed after reading Arts comment from the SynTripP thread.

In retrospect, the 3dB gain the ports provide around Fb are probably not worth the -3dB 350 to 475 Hz and-10dB cancellation at 700 Hz. That upper cancellation requires more output from the HF driver to “fill the hole” in response they cause. Most 3" diaphragm drivers are already excursion challenged in the acoustical crossover range.

To sum up, don’t bother with cutting the port holes, leave the cabinet sealed 😉

Click to expand...

Subwoofers:
I haven’t decided on the subwoofers yet.
I’m considering adding two subs within the same enclosure as side-firing woofers in a push-push alignment. For example, two GRS 8SW-4HE drivers in a sealed configuration with a ca 42L volume. Here is a promising recommendation for these woofers.

If I go with this approach, I might decrease the vertical coverage angle of the MEH horn to 60 degrees to make space for the woofers below the horn. I guess it would be good idea to have the woofers close to the horn, within ¼ wavelength of the crossover frequency, but have also read that it doesn’t matter very much for frequencies below 100 Hz.

Adding the subs in the same enclosure would significantly increase the height of the enclosure, which may not pass the WAF. Alternatively, I could build a separate subwoofer placed approximately 1.5 meters away

Is it better to integrate side-firing subwoofers within the same enclosure or should I make a separate enclosure, which would need to be ca 1.5m away? I’d appreciate any feedback or suggestions on driver selection!



Amplifier and DSP:
As I don’t own any of the parts needed I have the opportunity to build the whole system from scratch.

The TPA3255/51 amps seem to offer great value for the money.
I’m looking into: Fosi ZA3, AIYIMA A70, Topping PA5 II and the upcoming 3e audio amps. I haven’t decided yet and would appreciate any recommendations.

Finding a DSP solution that fits the budget has been tricky. High-quality DSPs can be expensive, while the cheaper options often come with limitations.

I’ve decided to go with a Raspberry Pi 5 running CamillaDSP. For the DAC, I’ll start with the cheapest option: the AliExpress cards mentioned in this guide. RPi 5 Quad Stereo Sound with PCM5102A – Simple DIY Electronic Music Projects. User dptucunduva has had good results with it.

The reasons I chose it are:

It’s cheap and has good potential for upgrades. I can switch out the DAC module for a proper sound card like the Motu Ultralite mk5.
It’s a very flexible system, the Raspberry Pi can act as a streamer and allows for lots of add-ons like a remote controller, Those who use CamillaDSP seem very happy with it.
HiFiBerry DAC, which uses the same DAC chip seems to be decent according to the measurements here. I’m not expecting any fantastic results but see it as a cheap solution that has all the active crossover functions needed.

Next steps:
The CD and mid-drivers are on their way, and I plan to begin 3D printing the horn in early January. In the meantime I’ll focus on refining the horn design, adding mounting holes, figure out how to do the right vrc size. I also need to decide which amplifiers to use and start building the Raspberry Pi DSP. Additionally, I’ll work on designing the subwoofers and determining whether side-firing woofers are the best option.

Any suggestions or input would be greatly appreciated!

Hi Everyone,

Just letting you know that Hornresp Version 18.00 has now been released.

The Tapped Horn Wizard tool has been significantly enhanced. For a given overall horn length, flare rate and driver specification, the optimum driver position can be determined more easily than before.

Note that there are three ‘Interaction’ options. The operation of the first two should be pretty obvious from the titles. The third ‘L12 & L34 Linked’ setting enables the user to “slide” the driver along the mounting baffle / separating panel towards or away from the 180 degree bend (similar to changing the acoustic path length of a slide trombone).

Hopefully the Tapped Horn Wizard is now a far more useful tool. Enjoy 🙂.

Kind regards,

David

Hi!

Stumbled upon these today and tried searching to see of it has been discussed already but could not find anything?

5,4inch/137mm woofers from Bliesma in 2 variants, textreme and paper.

I am a total noob when it comes to these things but will be building a pair of speakers when my CNC kit is here and assembled.

Maybe a 3-way with the t25 + t74 domes and this woofer all with the textreme diaphragm? Don't know if this would work but would make for a cool looking compact 3-way maybe? Supported by some subwoofers?

Link to shop where i found them: https://audio-hi.fi/en/woofers-c-1_5.html?filter_id=33&sort=0a

A bit pricey maybe but still half the price compared to their 7 inch domes...

ELEKIT TU-8888 Monoblock (available now)​

Another wonderful products from Mr Fujita...
PCB type DIY KIT...


















Here below are the spec of the monoblock amp.
Tube usable : KT series from 88 to 170 (for output tubes) KT170 recommended
EF86 x 1 and ECC82 x 1 (driver tubes)
Max. output : Over 50W/unit (with KT170)
Manual bias adjustment - A meter at the top panel
No headphone feature
SP impedance : 4ohm and 8ohm
Input x 1
Balance adjustment volume (with 41-point click) on the rear side
Coupling capacitors : 3pcs (100n 450V x 1, 470n 630V x 2)
Max size for the coupling cap :
100n 450V ... D22 x L44 mm
470n 630V ... D20xL57 mm
* as of now and could change the spec a bit.
Tubes are not included.
*The amp is designed to use custom made Lundahl LL3733"
PCB for OPT and harnesses are included in the kit.

There are only 30 pairs or 60 pieces for USA/Canada in December​

​

Price after Dec 15, 2024 $1,475.00 (will ship from Vancouver Canada)​

Using the BA-3 gain stage (“Front-end”) as a line stage preamp. A mini build guide. 🙂

PCB - P-BAGSN-1V20 - Burning Amplifier Gain Stage for BA-3 (Requires bias boards and output stage; Makes 2 channels; Rev 2.0) - Circuit Boards

BA-3 as preamp thread - http://www.diyaudio.com/forums/pass-labs/234641-ba-3-preamp.html

A club in Minnesota that used this preamp as their club-wide reference! https://sites.google.com/site/audiosocietyofminnesota/Home/diy-projects/pass-ba-3-preamplifier

The BA-3 front end is able to act as a wonderful preamp, with plenty of gain that is changeable with just a few resistors.

You will need a power supply, selector/attenuator, and chassis.

The chassis used in this build is the Galaxy 388 from the diyAudio store. Galaxy 388 (3mm Front) - Compact with Quasi Heatsinks - Chassis



Schematic of signal circuit.


Any project needs to start with a good PSU - this is an Antek 20+20V shielded transformer, and 317/337 linear regulator PSU from Tubecad.com. (Model PS-12) Of course you could use the Super Regulator V2.2 from the diyAudio store. Super Regulator V2.2 - Power Supplies and Accessories - Circuit Boards Look to the support thread to see the changes for 24V operation. (I didn’t use the Super Reg as that particular PCB wasn’t available when I ordered all my stuff…)


The back panel is very straightforward, IEC, and RCA jacks.


The inside of the front panel shows the three-input selector switch (Tubecad.com Select-2) the stepped attenuator (This is a 50K Goldpoint) and the AC switch (Tubecad.com AC switch)


Transformer wiring shown here - IEC (which holds the fuse) to the AC switch, then to the transformer primary. Transformer secondary to the PSU board. (configured as a center-tap) The purple shield lead connects to the same point as AC mains safety earth.


I used small coax cable for signal wiring. (actually stripped out of a long S-video cable) Each ground and signal is switched. Another very good choice for signal wiring is twisted pair from a cat-5 cable.


Switch wiring.



At this point the input wiring, selector switch, attenuator, AC, transformer and PSU input are all complete.


PSU output wiring.
Also (not shown) I took a LED and resistor from V+ to gnd to act as a power indicator.



BA-3 gain stage (Front-end) PCB. Note that there are PSU connections (V+, V-, GND) for each channel.
Also raise resistors R10 and R11 as shown, you will need to clip test lead there to set this stage’s bias and DC offset.


Completed FE board.




If you connect the BA-3 power wires to the back of the PCB they will be out of the way and closer to the chassis - two birds, one stone. Also in most of these photos, R2 is not installed. I later put it in and the attenuator switching pops went away.


Again, please install R2. In this photo you can see where I attached the signal ground. The ‘G’ pad has the PSU ground attached from the bottom.


Completed preamp back.


Front. (The power indicator LED is not shown.)

This amp is in for repair, in circuit test are blown output, amplifier has power but going on protect. My problem is I’ll like to know the part number original or solid substitute for the output mosfets.

I have been using LifePO4 batteries for my FIFO and DAC project for a long time. And also ultra capacitor power supply recently. I'm very happy with their performance and the sound quality.

http://www.diyaudio.com/forums/digi...mate-weapon-fight-jitter-353.html#post4239435

http://www.diyaudio.com/forums/pc-b...i2s-dac-hats-raspberry-pi-57.html#post5478596

Both ultra capacitor power supply and LiFePO4 power supply are passive power source. They are totally different to all kinds of regulators which are based on feedback. The most important thing is the perfect load response, much better than any regulator.

Both of them are ideal solution for DAC projects with amazing sound quality. Ulara capacitor power supply is a little bit better but the difference is minor. However the LiFePO4 power supply would be cheaper and easy to integrate into a all in one solution.

I'll work on both for sure. But I'd like to start from LifePO4 power supply.


LiFePO4PowerSupply_1 by Ian, on Flickr


UcPureProInatall7 by Ian

Ian

I would be grateful if anyone can explain why the DF1704/6 filter chips need to have the bit depth defined for left-justified input formats. The options available are 24 bit left-justified, 16 bit I2s and 24 bit I2s. Does truncation take place if 24bit data are input to a 16bit I2s setting?

I'm in search of a decent 8" midbass driver. Needs to do mid-90s dB/2.83v and handle the 300 - 900 hz range with ease, preferably 4 ohm impedance.

The main issue I'm running into looking for a suitable driver is sufficient xmax along with efficiency. The driver will need to handle some decent power without turning into mush with loud transients. Obviously it will need to be some sort of pro audio driver given the application. Low distortion is a must.

Drivers like the Audax PR170MO don't have enough xmax to play louder without using steeper HP filter slopes. Levels I expect are 110 dB peaks in the specified range.

My other favorite B&C 8NDL51 isn't quite efficient enough and isn't available in 4 ohms. The 8PS21 has the same issue.

The 8PE21 is efficient enough as an 8 ohm driver, but not enough xmax to deal with a 300 hz 2nd order HP at higher output levels.

I've toyed around with the idea of using 2 drivers, but this will result in weird vertical dispersion issues.

Any suggestions?

I'm building some linear PSUs where I'd like to be able to use pluggable terminal blocks such as Wago etc to switch transformers.

Unfortunately one of the boards has a 6.35mm pin spacing for which I couldn't find compatible pluggable terminal blocks on Octopart, Mouser or DigiKey.

Standard screw down blocks from Phoenix etc will do the job, but I'd prefer plugs and sockets to avoid mistakes.

So I'm looking for plugs and sockets on flying leads, 2-4 poles. I'm familiar with the Molex type, don't much care for them. I've also used Anderson which are better.

Are there others, or is there a better way of doing this?

TIA 🙏

https://www.zmescience.com/science/news-science/meet-superwood-a-miracle-material/

Looks promising for cabinets if they create larger boards or sheets. Has anyone here already tried the 3M glass fibre reinforced polyurethane boards? Particularly interested of you got them in the UK?

Alex

As shown here: https://www.diyaudio.com/community/...nd-balanced-preamps-2025-presentation.428773/


Iron Pumpkin SE and Iron Pumpkin kits and finished

Both kits are comprised:

• Motherboard with all SMD work done
• assembled rotary switches for both Volume and Selector, for cheaper option, or
• TFT display/logic module fully assembled, fancy remote (CR2032 battery you need to buy, can't send battery in package), encoder assembled (pcb, cable), protection perspex screen for display; that for more costlier option
• assembled flat cables
• pcb IDC connectors
• all necessary CAD files for front and back plate, sent by e-mail ...... Modushop already did several Iron Pumpkin cases
• pack of schematic and graphic files, also sent by e-mail
• all resistors (metal film 0207 size), including NTCs
• all solid caps
• all electrolytic caps
• all heatsinks
• all necessary isolators ( mica, plastic bushings)
• all diodes and diode bridges
• all active parts - semis, ICs
• jumper pins and jumpers
• all Fujitsu micro relays
• assembled Iron Turtle AVC module
• 10VA mains Relay/Logic Donut, custom made; static shield, magnetic shield, primary for your neck of Wood; regular fixing hardware included
• 20VA mains Audio Donut, custom made; static shield, magnetic shield, separate secondaries for each channel, primary for your neck of Wood; regular fixing hardware included

**************************
WHAT YOU DON'T GET IN KIT:

- small screws, nuts and washers; you're Diyer, having them all

******************************


Kit prices (all in Euro - €) ............. first take a look at available pictures and graphical files, then on list, and just then look at numbers

-SE, Metglas, rotary switches - 1100€
-SE, Finemet, rotary switches - 1200€
-SE, Metglas, TFT + logic control - 1350€
-SE, Finemet, TFT + logic control - 1450€
-Bal,Metglas, rotary switches - 1350€
-Bal,Finemet, rotary switches - 1450€
-Bal,Metglas, TFT + logic control - 1600€
-Bal,Finemet, TFT + logic control - 1700€

That includes shipping wherever you live, but not Paypal fee; if Paypal fee (starting with 30€) is too much for your stomack, possible payment to my Biz Bank Account, usually it costs less.

Shipping time frame mostly dependant of custom Donuts production; they usually deliver in two weeks time.... so there it is.


Finished preamp prices (Modushop 3U/280 case, Neutrik connectors, front black or silver, mains voltage according to your needs):

-SE, Metglas, rotary switches - 2750€
-SE, Finemet, rotary switches - 2850€
-SE, Metglas, TFT + logic control - 3000€
-SE, Finemet, TFT + logic control - 3100€
-Bal,Metglas, rotary switches - 3000€
-Bal,Finemet, rotary switches - 3100€
-Bal,Metglas, TFT + logic control - 3250€
-Bal,Finemet, TFT + logic control - 3350€

........plus legal export/shipping worldwide (100€ is covering that); Shipping goes through "Post Export" service, declared value of 500€, to save you of bigger Custom expenses

Hello.

I have 4 sealed Rythmik subs connected to a miniDSP 4x10HD. I implemented MSO and the improvement in sound is very noticeable, however the measured in-room response is quite different from the simulation on MSO.

Here's the MSO screen: 3 measurements at the main, left and right listening positions. Displayed are the before MSO optimization and predicted optimized response. My subs xo at 60Hz with steep xo such that they drop to no amplitude by 80Hz. The predicted optimization is very flat from 20 thru 80Hz.


I installed the biquads on the miniDSP, and while the response is better than the baseline responses displayed above, they are far from being as flat as MSO predicted. This is a measurement of the full system (showing 15 thru 100Hz only), measured at the main listening position:


I'm hoping experienced users might be able to provide tips to figuring this out. Or can point me to a better forum to ask for this? I've noticed surprisingly few mentions to MSO here so might not be the best place to ask?

Thanks in advance!

There is a fella selling a ton of test equipment on the "Vintage Electronics Test Equipment" page much of it 1980's 90's vintage and seemingly in good shape. I certainly don't need another 576, but ones in good shape are rare. Also Tektronix, Fluke, HP, Keysight etc. I have no connection to the seller.

Calling all Calgarians!

It seems we have a bit of a diyAudio continent in Cow Town. Notably, @bxt2008, @leadbelly, @kirks, and likely others. In a recent thread I floated the idea of a get-together over a few pints at some point and there seemed to be some interest.

So how about this: We meet at Two Pillars Brewery (910 Centre ST N) at 8pm on Wednesday January 15, 2025. The idea is just to put a face to the user ID and to chat/BS about all things audio.

Please respond below by January 7 if you plan to make it. If it turns out we'll be more than Two Pillars can accommodate I'll find a larger venue that's reasonably central.

Hope to see you at Two Pillars!

Tom

I know very little at the moment about the ability of AI programmes to generate sensible, or good data for typical 2 - 4 way loudspeaker design, hence my ramble questions below.

The candidate AI programme I am asking about here is Claude.ai. I assume the free version or do I need to go to the subscription version?

Kindly, can somebody provide a brief overview of what the free version would allow me to achieve, is it viable for loudspeaker design ?

Forum member mechano23 using XMachina has already shown that you can make a well received design, if the correct parameters are applied. I definitely do not want to have to spend three times the amount on crossover components compared to the £80 driver I wanted to used. Cost are one constraint or parameter to be entered?
Do they accept FRD and ZMA file data, do they need real data or can they work from manufacturers data sheet data and apply diffraction loss other cabinet effects, once cabinet data is entered?

Within the questions I am not sure how many microseconds these things take to produce an output, or does it take the course of an evening whizzing around searchable data on the internet before it provides an answer? As a preference I like graphical display in the time and frequency domains as they allow me to have a indication of what is going on

In terms of model training for a loudspeaker design, are you allowed to describe the parameters that would allow you to measure pre defined standard say IEC 60268-21, the Harman preferred curve, BBC, Stereophile reviewers listening room, or audio express recommendations etc.

If not part of the AI and it is up to the designer to provide parameters, maybe we should start listing what parameters we think are relevant.?.
Does somebody want to share their design an parameters to enlighten me. How would it handle a remake of a Spica TC50 using Purifi drivers for one example and currently available Peerless drivers for the other.

Finally I assume as the models learn over the course of time one user in Shanghai will end up with exactly the same Xover design for a certain set of drivers and optimum box size, as somebody in Jakarta, if they use exactly the same choice of drivers?

Any help, thoughts or guidance much appreciated.

Hello,

as I promissed just to share with you concept from YU3MA forum, this is pulse densinity modulated H bridge power dac driven by dsd64, in short it's an full bridge class D direct digital power amplifier pulse densinity modulated, contain no opamps... etc just directly flow DSD bits to the full bridge, more comparable to an water flow boiler with diference boiler flow hot water but this one flow dsd bits. : ) Tested up to dsd256, it might work on dsd512 too? Concept contain variable power supply as a volume control, 1.25V to 5V range. You need Amanero usb combo and you need DSD tracks in order to play sound trought this dac. You will be surprised by the huge dynamics and holographic unchanged and detailed sound! The power of this dev concept is in the range of 3 to 5 watts and I do not recommend more, you simply need to first adjust the current with CC pot by shorting +5V at the one of the 8 capacitors for example C165 to gnd with an ameter until your ameter display value close to 1A, than you are ready for sound testing. Here is soome pictures and videos and more details about further concept . Its fully diferential! Optimised for 8 ohm speakers! Not measured but I have bought Cosmos ADC and when I finish some concepts and when I get ready for measurement I will do it, hope somebody do it allready. Gerber, bom and schematic is in atachment. Please pay atention to R72 resistor, change it for 2mA, ohm law yuor input dc voltage and calculate resistor value for 2mA! Enjoy!



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Not really audio related but I figured I will ask here. I have an AC motor that runs on 115V and draws 1.3A. I need to drop the voltage going to the motor (to slow it down) by using 10 Ohm wire wound resistor. What wattage of the resistor should I use so it is safe for a continuous use?

Not much discussions here lately.

What is the current status of the website?

I found a post on StereoNet referring to Melbourne Audio Club meeting end of May and that Hugh would be presenting a new amp? Newer than Maya v4 and Titan?

To keep things accurate what are the current prices for the range of amp kits?

Hope you are doing well, Hugh.

This product is a series noise filter for power mains for your audio system and started asa P2P on the bench and has been under development for some time now. I am pleased to say that the production prototype has been completed and passed CE and FCC testing and certification.

It is a 5th order Chebyshev filter built with premium name brand components including two massive 19.5mH 10A 1000v PCB mounted CMC chokes. It should provide about -50dB of noise attenuation at 100kHz and more at higher frequencies. This should significantly reduce SMPS noise in power lines. Good news is that it is rated for 10A and works from 85VAC to 230VAC and has combo NEMA/Schuko plug receptacle on the front.

Here is a render of the final product. I don’t have the final in my hands yet.



But here is a photo of the inside from the test lab:


This was my bench testing of an earlier prototype with 2.5mH inductors. We later moved to larger inductors and smaller Y2 capacitors to reduce parasitic dissipation to comply with CE requirements. This had a huge impact on the cost of components as you can imagine going from a 2.5mH to Kemet 19.5mH 10A CMC - but makes for a well engineered, safe, energy efficient, and robust electrical appliance that will clean up noise like a black hole.

Testing below verified -50dB at 100kHz noise removal.

Hi, I've been reading the "dome midrange thread" and keep getting that itch to try something new.
My current speakers are an active 3 way that I've made myself using using drivers that have been recommended from this awesome forum. They consist of a T25b tweeter(5in wave guide), NE149W-08 mid and an sb34nrxl75-8 woofer in a 65L ported box tuned at 29hz, Everything has been impedance checked for resonances and standing waves and rectified where needed.
I made these speakers modular so I could try different things.
They currently sound amazing and I'm blown away every time I turn them on, they have a large 3d/deep sound stage with great detail and an amazing easy listing experience but you know how it goes with us type of people!
These are run from a Minidsp flex ht, 6 channel home assembled Icepower amp(1200as2/400A2/125asx2 new conductor version for the 125asx2). Crossovers are currently 250hz/2700hz all LR2. I have a good bit of experience with DSP from my car audio setup also with a mindsp 8x12dl.


I have a spare set of NE149W-08 drivers and can sort out another two channels of amp. What i'm thinking is to make another box section that sits between the current woofer box and the midrange pod. Then put two NE149W's side by side in that box(that would be the same style as the woofer box) and then a scan speak D7608 3in dome in the mid pod(with the correct enclosure size of it's own) or make a new mid pod that gets the ctc spacing a little closer.
I did also think about using one NE180W for the midbass but if the two 149's per side would work then that would be better as I already have them.
Then I would aim for crossovers somewhere around 150-200ish/6-700/2700 or whatever works best.
Trying to manage tweeter height would also be of concern.
My current desire is lots of detail and good dynamics. I'm also not really wanting to spend Bliesma M74 type money on this experiment.

Any thoughts or problems you can see?

Thanks.

In 1977 I built a pair of transmission line loudspeakers, similar to the famous Webb design, but somewhat larger. When I went to graduate school in 1980, they were too bulky to bring along, so I sold them. Now, both myself and a friend from those days want to build a couple of pairs. So I'd like to locate a copy of the plans.

With that as an intro, does anyone here have a catalog or plans from the company that produced the kit? It was called Accurate Sound, and was at 1213 M Street, Lincoln NE 68508. Below, I've reproduced a copy of an advertisement of theirs, from the January 1977 issue of Audio Amateur, page 49.




Does anyone here recognize this, or have an old catalog or plans?

Cheers,
Bruce

I am able to use the analogue outputs on the UMC1820 with camilladsp (on linux), but nothing seems to get through on either the electrical spdif or the optical toslink. Using an alsa loopback works fine to get audio to the UMC1820, but I really want to drive camilladsp with input from another computer, not the one running camilladsp.
The number of channels for analogue output also seems to vary, having worked with 8 channels, 10 channels and even 12 channels . Can't remember what the circumstances were at the time.
My current conclusion on this issue is that UMC1820 simply does not work properly with camilladsp / linux. Prior to this, I was using brutefir, mainly with other cards, but also with the UMC1820 - it fairs no better.
If someone could show me a config.yaml that will work to access the UMC1820 I would be very grateful. I have wasted literally weeks on trial and error tests.
None of these issues are helped by the complete lack of any detailed tutorials or instructions concerning any linux software. The instructions, such as they are, are always incomplete and seem to assume you are an expert in everything that the instructions do not cover.

Many thanks,

There is still some interest after our last group buy for these modules - so I will open another thread here for a new group buy.

What is offered:

4x pieces of a speaker protection module as shown in the thread here:
https://www.diyaudio.com/community/...-universal-speaker-protection-circuit.415149/
enough for 2 stereo amplifiers.

Since the MOSFETs used are tricky to solder effectively at home, the boards will be provided with the MOSFETs already soldered. The rest of the parts will need to be supplied by the subscriber. They are easily available and in stock at Mouser, DK etc. The modules use SMD parts with the smallest at 0603, but most are larger than this - I'm only saying this as some people do not like using SMD parts. Full BoM and build/test instructions will be provided on paper as is usual.


Technical details:

The built units can be used for amps with total voltage from 24V to 100V, either single or dual rails. It must be run from the existing amplifier rails and you do not need an additional auxiliary supply. Current draw is less than 15mA per rail. The boards are designed to be mounted on the amplifier heatsink - one TO-126 transistor is configured to be bolted to the heatsink - it will dissipate up to 1W.

If using BoM values the units provide a start up delay to avoid turn-on thumps of about 3s and will trigger after the DC level exceeds the threshold for around 2s. The trigger voltage is around 1.2-2V depending on components used.

Board dimensions are 30x58mm

The boards are designed so that a mirrored pair can be made using the same PCB, making input and output wiring simpler and neater for each channel in a stereo amp.
By design, the protection is latching - i.e. it does not automatically reset for a long period. You will want to figure out why there is DC on the output before resetting the latch! It is also possible to manually reset the latch for testing etc.

Costs:

€60 for a set of 4x boards, with 2x MOSFETs per board pre-soldered. Tracked and insured postage anywhere in the world is €15. The rest of the parts will cost about €30 per stereo pair without any volume discount, depending on where you shop for them.

If you would like to take part, please add your DIYaudio name below here along with the number of sets you would like to sign up for. 1 set = 4 channel protection PCBs (2x stereo amps).

Example:

woodturner-fran - 1 set

The group buy will require a minimum of 20 participants/sets. When the group buy closes, it is expected that ordering and delivery will take about 3 weeks, and once the units are here and verified, I will send invoices to each participant. As usual, no payment until the boards are here and ready to distribute. Previous experience shows it will take an average of 2 weeks to arrive anywhere else in the world.

Thanks for your interest!

Does anyone have experience with KEF 105 series 2 speakers? Wondering if anyone has updated or upgraded and how it turned out?

It all started in the mid 1970s, when I worked as a mechanical designer for Scully Recording Instruments Co. in Bridgeport, Connecticut. They manufactured professional tape recorders and of course the Scully Lathe. They were located in the old Underwood Building on Bunnell Street. The Engineering department and the machine shop were upstairs and on the downstairs floor there was a demo room with a Scully 100, 2 inch, 24 track tape recorder and two JBL wooden horn speakers.

During my lunch break, when there were no customers, I would go down to the demo room, eat my sandwich, and listen to whatever was on the 2 inch tape on that particular day. The experience of listening to that professional sound equipment was overwhelming. It started me on the path of designing my geometrically perfect tone arm. My co-workers all had an interest in music in one way or another. Some played musical instruments and one was a part-time recording engineer. I played acoustic guitar and sang folk and country music.

Upon the advice of my co-workers I purchased a Thorens TD-124 turntable and a separate SME 3009/S2 improved tone arm. I started to purchase LPs and soon realized that a pivoting tone arm could not faithfully reproduce music the way it was recorded.

Being fascinated by all things mechanical, I set out to design my own pivoting tangentially tracking tone arm. Although my hat is off to the designers of tone arms like the Rabco and the Goldmund, I did not like the fact that those tone arms occupied so much space on a turntable. I also did not like the various pivoting tangential tone arms like the Garrand Zero 100 because they still skated and had tracking errors. Being a perfectionist, I could not understand why all the talented tone arm designers would stop short of zero tracking error and zero skating force designs.

This thread does not concern air-bearing and Souther style tone arms which are mostly perfect.

I always liked the classical appearance of pivoting tone arms, so I concentrated on them. I designed, built and patented my first tone arm in the early 1980s, and I would have continued if it were not for the fact that CDs came on the market. It wasn’t until 2009 that I found out that vinyl LPs had made a come-back. By this time, however, I owned a CAD program, had built a shop and purchased an almost new Bridgeport vertical milling machine.

By 2010 I had redesigned my original 1980s tone arm by moving its tracks and the carriage into the base underneath the tone arm. It required an active servo and was featured in a thread on the DIY website. In 2012 I improved the tone arm by inventing the “FLOATING HEAD-SHELL” which is shown as the third item in the group picture. Like the 2010 tone arm, it also required an active servo. Not knowing electronic design, I listened to LPs without the necessary servo by nudging the tone arm’s carriage along its track every 30 seconds or so. That became possible, because I separated the head-shell from the tone arm proper. Not knowing anyone who could design the servo circuitry for me, I kept making mechanical improvements until in 2015 I invented the “OFFSET HEAD-SHELL CRADLE”.

Then in 2016 I changed my design from a carriage rolling on a track to a swing arm to support the tone arm. Of course many inventors before me discovered that a swinging support arm is far superior to a rolling carriage. Then, early in 2017 a kind gentleman whom I met on the internet, designed the servo circuitry for me. I transferred that circuit to a printed circuit board and showed it to the public at the Rocky Mountain Audio Fest (RMAF) in October 2017, in Denver, Colorado. The transition to the swing arm reduced friction to the point where the tone arm started to float across the LP purely from the drag force between the LP and the stylus. The servo has become a “passive” servo and only controls the variations in the drag force. Some of the servo components were visible when looking down on the tone arm, which I did not like, and in late 2017 I moved those components to the underside of what I call the tone arm shelf. That way the tone arm would retain its classic appearance and it would make it harder for intellectual property thieves to steal my invention. That being said, anyone who wants to build a copy of my tone arm is free to do so, as long as it’s not for sale to others.

My original 2017 servo’s printed circuit board (PCB) utilized “through-hole” components. In 2018 I redesigned the PCB to utilize “surface mount devices”.

To summarize, I have invented a pivoting, tangentially tracking tone arm with a “FLOATING HEAD-SHELL” carried by an “OFFSET TONE ARM CRADLE”. Please note, that the HEAD-SHELL and CRADLE are NOT offset to reduce tracking error, as in a conventional pivoting tone arm. The offset cradle serves a new and different purpose.

I have a complete set of 3D solid model CAD files and paper drawings for my tone arm. I also have a number of simple but precise aluminum fixtures that I have made.

Now, after several years of machining, testing and experimenting, I am finally able to listen to my favorite music played back with a tone arm of my own design. That experience cannot be described and those of you who have designed and built your own audio equipment know what I am talking about. I have kept my modest collection of 250 LPs, which I purchased in the 1970s and 1980s in almost new condition and recently, a lady friend of mine gave me around 35 boxed sets of LPs that had belonged to her deceased son. There is enough music there to last for the rest of my life.

I hope that the members of the DIY website will forgive me for not making public all the design features of my tone arm, because I want to make it difficult for the copy-cats to profit from my invention.


And, last but not least, I would like to find a manufacturer to make and sell my tone arm.



Please watch for 14 photographs in my next post. I promise, you will be amazed.


Sincerely,


Ralf

I bought the flagship DAC project from Ian Canada. Gabster's YouTube videos were very helpful putting it together but I need help connecting power. I'm using two Hammond transformers and I've confirmed with a DVM that I'm getting 3.3 volts from one LinearPi Pro, +/- 5 volts from a dual LinearPi Pro and I have another LinearPi Pro that I'll use to power a Raspberry Pi with +5 volts once I get the DAC working.

For the default setup of this project, where do I need to connect the different power wires? Should I use the output from the separate LinearPi Pro for +5 volts somewhere or do the +/- 5 volts wires provide that voltage?

Thank you,

-Robert

Hello!

I'm going to pick up some Acoustat M3s (I think they are M3s, not 100% sure of the model) from someone's garage tomorrow. Don't know much about the condition except I was told they work. Looking forward to checking them out. I've always wanted to play with some electrocstatics. Do folks think these are a good speaker, and what do I need to know about them?

How should they be transported? I was planning to remove the assembly from the back and stack them flat on a piece of plywood.

I am planning to power them from a McIntosh MC2105, does this seem like a good choice? It has a 4 ohm output stage.

Thanks for any advice!

Edit Nov 5, 2022: nice analysis and summary of DML materials and response with python script by @homeswinghome

https://www.diyaudio.com/community/...s-as-a-full-range-speaker.272576/post-7159983

Edit oct 27, 2021: a nice Google drive library of DML publications by vdljev:

DML - Google Drive

Edit April 1, 2020: nice summary of DML patents by Burntcoil

A Study of DML's as a Full Range Speaker

Edit Aug 12, 2019: Nice summary here by BurntCoil on how to maximize performance.



I had some cheap exciters that I got from PE a while back and tested them out a while ago with a full 20x30in FC panel here:

Foam Core Board Speaker Enclosures? - Page 225 - diyAudio

I found that a large 20in x 30in panel can sound quite good with nice bass extension and a snappy transient response:



Impedance:



Frequency Response & Harmonic Distortion:



Impulse Response:



The results were interesting in that it sounded nice - with surprising bass and good midrange. I thought nothing more about it until recently prodded by the master of DML, CLS. He has worked on this extensively and is a treasure trove of good info. I since have discovered that you can really do some cool things with them by playing with driver placement, cutouts, adding mass, adding felt, adding ribs, making them huge, making them multi-way FAST, etc. the options seemed almost limitless. Take for example, a large center channel and a super 40Hz capable multiway that CLS built here: PIEZO NXT type panel - Page 60 - diyAudio and PIEZO NXT type panel - Page 61 - diyAudio

I feel like this technology just isn't getting enough attention. There are several very large threads on this on the web. It might be tough for a newbie to comb through. I am a newbie at this so thought I would document my journey for the Full Range forum to follow. I think it really could be a great full range speaker with some careful experiments. This will be really useful with some modeling using CAD and FEA modal analysis - such as available in many CAD packages like SolidWorks. One can play with shapes, cutouts, mass loading, variation in thickness and materials, boundary clamping conditions, etc. The idea is to spread the modes evenly with not any one mode dominating and causing a spike.

First thing is to play with it to get a feel of what we are dealing with. What is nice is that it is relatively cheap to play with. Exciters cost $3 to $20 ea and can be made by removing or cutting out large holes in the cone from an old driver and leaving the spider and VC to attach to the panel. Foam core or corrugated cardboard doped with shellac or PVA seems to be the materials of choice.

Some interesting facts:

1. Although it has no baffle, it is not an open baffle (OB) dipole in behavior - that is, there is no huge bass rolloff and it hits surprisingly low (40 to 50Hz is easy) for a zero baffle driver.

2. It is not a dipole but behaves more like a bipole or an omni.

3. It has very quick snappy transient response - nice drum sounds.

4. It is sensitive to how you mount it or frame it or hold it.

5. You are building a driver in reality - a driver and zero enclosure.

6. You want to avoid symmetric shapes and symmetric exciter placement in order to reduce the effects of the main symmetric transverse drum head eignmodes. Think reflection anti-symmetric shapes like uneven trapezoids, pentagons, blobs, etc.

7. It operates more by having high velocities and large areas for good efficiency vs large displacements - thus small drivers and large panels can be surprisingly loud.

8. The impedance is essentially "flat" relative to normal drivers in that there are modal peaks (many of them) but they range from nominal Re value (say 6 ohms to maybe 8 or 9 ohms throughout the 40Hz to 20kHz range - this presents a very flat load to an amplifier.

Here is a photo of a basic panel I was testing (right before I cut off some edges to form a trapezoid) - panels is about 1 square ft in size and made of standard dollar store foam core board. I suspended it with two pieces of twine from the top corners between a ladder to reduce the effects of edge boundary clamping:



Then I started to play with mass loading by adding blobs of modeling clay (8, 5, 4, 2, 1, 0) so you can see the effect on the resonance modes and the harmonic distortion. What is neat with this test is you can do it live while playing music and immediately hear what sounds more pleasing to the ear. Note the 50Hz bass extension. It is also surprisingly efficient with 85dB and a puny little voice coil. These are 0.5m and 0.71v for equivalent SPL at 2.83v and 1m.

Frequency Response and Harmonic Distortion for...

No added mass:



8 pieces of added mass:



5 pieces of added mass:



4 pieces of added mass:



2 pieces of added mass:



1 piece of added mass:



Impulse Response of 1 piece of added mass, note the sharp transient capability with relatively low after pulse or ringing:



I liked the sound of the 5 and 1 mass the best and for a FAST with a 200Hz XO, the 5 mass might actually do fairly well. Of course, sound clips to follow once I have more time to work on this. Just to demonstrate how quick and easy this is, I did all these experiments in less than an hour including making the DML.

Esse circuito de amplificador super simétrico aplica boas técnicas discutidas aqui no forum.
Essa topologia de circuito é a mesma aplicada na tese de mestrado do Professor Leach.
McIntoch, Rotel e outros aplicam a mesma topologia de circuito em seus amplificadores de estado sólido.
Agradeço por comentários relacionados com melhorias nesse amplificador. Corrente de bias é um parametro que pode ser escolhido pelo montador.
(Comentários em ingles pelas regras do forum).


This amplifier idea uses good technical ideas discussed here.
The circuit toppology is the same used by Leach in this master tesis.
McIntoch, Rotel and others makes uses the same circuito topologies in his solid state amplifiers.
I appreciate comments concerning improvements of this amplifier. Bias current is a parameter each one can choose and it wil not improve this amplifier as it is a option of the amplifier maker.
Regards

I am new to this forum, so apologies if the answer to this question has been worn out, but I am interested in building an F2 clone. Dick Olsher is mandating use of the F2 to power the Eminence woofer in his Basszilla speaker and I can't afford the store-bought version.

Has the Reverend Pass released the necessary circuit information to now make a DIY clone possible? If so, where does one find it? If not, is there some timetable?

last weekend I have heard two different power amplifiers by direct comparsion:
1) Pass "X 600" vs.
2) Yamaha "MX 10.000" (MX10000, MX-10000)
I was surpriced about the low sonic differences between this two devices and I guess, that the HCA technology is also a good solution, especially if the loss power must be lower by the same output power.

Are there HCA diy projects respective other commercial amplifier brands, where is HCA technology inside?

Hello everybody,

The last months I read through this threads to help my in my repair of a NAD C372.
Since september last year I decided I wanted to learn to repair on a deeper level and so I use this case as a learning study.

What was happened?
The amp went into protection and the left channel was toast

What did I already do?
I went through the entire left amp channel pcb and tested every transistor out of circuit (with a Peak DCA75)
The following were gone and were replaced:
Q316, Q313, Q318, Q338, Q317, Q337 and Q328, the one that caused all this.
Q315 is also replaced.

3IC1 en 3IC2 were also replaced, as were R349 and R329 who are fusible resistors for the 64V line.
VR302 and VR301 were also changed.
At last IC301 was also toast, which I replaced.

All diodes were tested. All resistors were measured, some out of the board to make sure they value is correct.
All capacitors in the entire amp I changed to new ones, including the PSU rail.

So what is the problem now?
I can regulate the DC offset voltage between 0 and 3V als described in the manual.
But the big problem is the bias doesn’t go higher then 3mV.

The weird thing is that on the other channel I can’t even go beyond 2mV.
Here the DC offset opamp is also changed and also the 18V zener diode who was defective.

Since I checked both board completely as described above I can’t find the cause of this.
The only thing that stands out is that the supply voltage is 73V instead of 64V.
Is this the problem?

The amps plays music and works already but this bias problem indicates there is something wrong…

Can you guys help me?
Service manual is in the attachment

me.

I'm working on building an Ian Canada DAC. Other (prebuilt) DACs I have or have used: Emotiva Xda-3, Schiit Yggdrasil and the Oppo BDP-103. I returned the Yggdrasil since I couldn't hear an obvious difference when I compared it to the BDP-103. The BDP-103 is getting long in the tooth and can no longer access my NAS.

-Robert

hi there,

I know that some of our users here have pre-ordered some of those new Bliesna T34 tweeters which are shipping now.

It's a brand new company, the owner and inventor was former developer at Morel and later product-engineer at Accuton.

Now he came up with his very own product. The T34 tweeter.
A 34mm (1.3") aluminium/magnesium alloy dome tweeter with varaible thickness.

Have a look at the specs yourself:
http://www.bliesma.de/product.html

In this thread I'd like to hear experience with this tweeter and discuss this product.
The price is 580€ (ex. VAT) for a pair.

Hello All,

Fun to be back on a forum like this. I used to be way into audio gear, then kind of fell off for a bit. Got into in in high school when I bought a nigh club sound system, 2000 watt 4-way McIntosh & JBL custom setup. Also build a plasma tweeter back in those days (anyone remember Ulrich Haumann?) Wish I still had some of that stuff! Have been a nomad for a while but setting up a new system now that I am a homeowner. Looking forward to being part of the community and getting advice from the real experts...

Hello

Just to note ... My only phono preamp that I use every day is the JLH shunt feedback design
so decided to build another project as a direct comparison to see if something is better

Here is a PDF of of the Pearl 3 from Burning amp 2023 for those of you whom already have circuit boards.





EDIT: Video presentation on Pearl 3 for BAF '23 can be found here - https://burningampfestival.com/videos/


EDIT: Build documentation "Pearl 3 Phono Build Doc-1.0d" is attached below. Version 1.0d is the most current. Download and print.


EDIT: Interview with Wayne about Pearl 3 - #881

EDIT: Modushop/ HiFi200 Chassis set available here - https://www.diyaudio.com/community/threads/pearl-3-burning-amp-2023.404054/post-7538807 https://modushop.biz/site/index.php?route=product/product&product_id=971

EDIT: Online build guide - https://guides.diyaudio.com/Guide/Pearl+3/28

EDIT: SMD soldering tutorial using Pearl 3 - Login to view embedded media

CNC phonostage : One of the best phonostage designed by Hypnotoad, very easy to build and sounds great. Prices for each PCB $8 + shipping $6 by Registered post. This is a very highend phoostage a very long thread here Another Super High End Phono Stage! No expense spared... | Audiokarma Home Audio Stereo Discussion Forums
I can also arrange a kit. It includes all onboard parts minus any hardware like RCA, case etc,
Price $65 + $10 shipping
Fully soldered boards , Price $80+$10 shipping.



OPS_Rev1, Dual power supply is designed by Omishra. Its based on LM317/337. The output voltage is adjustable by using two onboard trimpots.
PCB $8
Module $25




Regards
Sachin

Hi
I have 6X DCB1 PCBs layout by my friend Omishra. This version can be built very cheap. There are very few parts on board. Here is a build thread DCB1 Build | HiFiVision.com

Price $8 each, shipping flat $6 by Registered post




Regards
Sachin

Let's return to the discussion of the wonderful amplifier.

I tried to improve it using a simulator.

I used field effect transistors in a voltage amplifier. I equalized the current of the upper transistor of the voltage amplifier by adding an additional current source, and disconnected the quiescent current control circuit from the voltage amplifier. The coil in the differential cascade increases the gain without feedback by 20 db

Hi all,

While thinking about ways to speed up the settling of a single-supply single-op-amp RIAA amplifier, see https://www.diyaudio.com/community/...upply-phono-preamp-design.413571/post-7702435 , I found a way to include a second- or third-order Butterworth high-pass filter. As it may be useful outside the context of single-supply circuits, I give it a separate thread.

When you just look at the topology and ignore the component values, this is a rather conventional RIAA amplifier (you could make it even more conventional by connecting R₇ in parallel with C₅, that doesn't matter much for the principle):



Normally, C₈ is used to cause roll-off in the subsonic region and the network R₇...R₉, C₅, C₆ realizes the RIAA poles and zero. In this case, however, I use C₈ to realize the lowest RIAA pole at -1/(3.18 ms) and R₇ to get subsonic roll-off.

Note that C₈/C₅ = 1000, meaning that without the subsonic roll-off, the DC gain would be 1001, a very ordinary value for a moving-magnet amplifier (1 kHz gain roughly 40 dB).

With everything ideal, at the value of the Laplace variable s where the impedance of C₈ cancels the impedance of R₁₂, the feedback disappears and the gain goes to infinity. This means that there is a pole at exactly -1/(R₁₂C₈), so if this has to be the first RIAA pole, one needs R₁₂C₈ = 3.18 ms. It's actually 3.196 ms in the schematic, pretty close.

The disadvantage of using C₈ for the first RIAA pole is that C₈, which has a relatively large value, needs to be accurate to get an accurate first RIAA pole. (C₈ has practically no effect on the gain at frequencies much greater than 50 Hz, so its tolerance affects deep bass, but not channel balance.) The advantage is that you can include better subsonic filtering in the loop by adding two more resistors and a capacitor.

As an intermediate step, suppose you could add an ideal inductor with a huge value between the output and the negative input of the op-amp, chosen such that it resonates with C₅ at the desired subsonic roll-off frequency, and that you chose R₇ such that it damps the LC circuit to a quality factor of 1/2 √2. The subsonic response would then be very close to second-order Butterworth. That's because the gain of the RIAA correction amplifier is one plus the ratio of the feedback impedance to the impedance from the negative op-amp input to ground, and that "one" is quite negligible at low frequencies. Mind you, R₈ and R₉ contribute to the damping of the LC circuit, but not by much. You could also choose a quality factor of 1 and design the AC coupling at the input for the same cut-off frequency. The combined response is then third-order Butterworth.

Such an ideal inductor is totally impractical, but it can be approximated with a T network consisting of two resistors with values much smaller than R₇ and a capacitor to ground at the point where they are connected, see this figure:



The transfer from the voltage going into R₁₁ to the current coming out of R₁₀ rolls off at a first-order rate from some very low frequency onwards, like would be the case with an inductor.

I haven't found any simple exact equations for any of the values except R₁₂C₈ = 3.18 ms. In fact, I've been very lazy and just calculated approximate values for the other components, and then used a pole-zero extraction program to fine-tune the values.

Regarding those approximate calculations:
R₁₂C₈ = 3.18 ms to get the first RIAA pole at the right spot.

The DC gain would be 1 + C₈/C₅ without subsonic roll-off, so C₈/C₅ = 1000 gives you a midband gain of roughly 40 dB.

At s = -1/((R₈ + R₉)(C₅ + C₆)), the impedance of the network R₈, R₉, C₅, C₆ goes to zero and the gain of the circuit becomes 1. As a gain of 1 is pretty close to 0, this must be close to the location of the RIAA zero. That is,

(R₈ + R₉)(C₅ + C₆) ≈ 318 μs

At s = -1/((R₈ + R₉)C₆), the impedance of the parallel connection of C₆ and R₈ & R₉ goes to infinity. The impedance of the whole feedback network remains finite due to the other branches R₇ and R₁₁, C₇, R₁₀, but it does get pretty large. That means the second RIAA pole must be close, so we get the extra criterion

(R₈ + R₉)C₆ ≈ 75 μs

The (theoretical) inductance L is chosen to resonate with C₅ at the required subsonic roll-off frequency and R₇ is chosen to get the desired quality factor. R₁₀ and R₁₁ get convenient values much smaller than R₇ with R₁₀ also much greater than R₁₂. We then have

C₇ = L/(R₁₀R₁₁)

Best regards,
Marcel

Edits:
Input RC coupling
The RCR T-network that approximates an inductor actually approximates an inductor with inductance L = R₁₀ R₁₁ C₇ and a series resistance of R₁₀ + R₁₁. At low frequencies, it stops behaving inductively, it just turns into the series connection of the two resistors.

As a result, one of the zeros of the high-pass filter that are supposed to lie at s = 0 actually lies somewhere around s = -(R₁₀ + R₁₁)/(R₁₀ R₁₁ C₇). For the second-order cases, I have used the first-order high-pass at the input to cover this zero by making the input RC time constant approximately equal to R₁₀ R₁₁ C₇/(R₁₀ + R₁₁), or actually to a more accurate value for the displaced zero found by the LINDA pole-zero extraction program.

For the third-order case, I have used the input RC coupling to make the real pole of the third-order Butterworth response, so I couldn't use it to cover the displaced zero. I used the output RC circuit in that case, or simply did not cover the zero. The effect of the zero not being in the origin is typically only seen below 1.something Hz anyway.

There is another zero not exactly in the origin, this is related to the + 1 term in the gain expression of a non-inverting op-amp amplifier. It is so close to 0 that I decided not to bother correcting for it.

Regarding the single-supply versions, the filter R₁, C₁ is meant to provide some power supply ripple and noise rejection. (Even though the single supply has to be regulated, some extra rejection is very useful at the input of an amplifier that amplifies hum frequencies hundreds of times.) R₂ and C₂ in series with the cartridge impedance also help to suppress supply ripple. The second-order versions have greater values of C₂ than the third-order versions and therefore have better ripple rejection at the lower audio frequencies (such as 100 Hz or 120 Hz).

16 Hz split-supply versions
This is a version for split supply and 16 Hz cut-off frequency, see post #58, https://www.diyaudio.com/community/...rworth-high-pass-included.413649/post-7927611



For the second-order version (values in parenthesis), the input coupling capacitor C₂ can be replaced with a short circuit if you don't mind when the roll-off reduces to first order below 1.3 Hz.

This is a variant with 46 dB rather than 40 dB midband gain:


Thanks to having R₆ split into R₆ and R₀, the time constant of the input RC coupling network can be set more accurately without needing awkward values for C₂. This was implicitly suggested by hbtaudio on another thread. Because of the high midband gain, the op-amp needs to have a fairly high gain-bandwidth product to get accurate RIAA correction (16 MHz gain-bandwidth product will give about -2 % error of the location of the second RIAA pole).

Finite gain-bandwidth product
See post #100, https://www.diyaudio.com/community/...rworth-high-pass-included.413649/post-7964481 , for some rough calculations on the effect of finite gain-bandwidth product of the op-amp.

From post #101 onward, Nick Sukhov points out that an amplifier with a high open-loop output impedance would result in a loop gain that depends much less on the RIAA correction circuit impedance. That's something to keep in mind when designing a discrete amplifier, you don't have the ability to choose a high open-loop output impedance when using op-amps.

Applying the subsonic filter to a discrete preamplifier based on the Hoeffelman and Meys configuration
The discussion with Nick and Chris about open-loop output impedances made me realize that the subsonic filter of this thread could be combined with a low-noise ("electrically cold") input termination resistance realized with a special feedback configuration that Dual already used in the late 1960's (CV40 phono section, see https://www.diyaudio.com/community/...o-input-load-modification.424717/post-7947176 ) and that was advocated by Hoeffelman and Meys in a 1978 AES article (Jean M. Hoeffelman and René P. Meys, "Improvements of the noise characteristics of amplifiers for magnetic transducers", Journal of the Audio Engineering Society, vol. 26, no. 12, December 1978, pages 935...939, see also Ernst H. Nordholt, "Comments on "Improvement of the noise characteristics of amplifiers for magnetic transducers"", Journal of the Audio Engineering Society, vol. 27, no. 9, September 1979, pages 680...681). The very first electrically cold resistance was made by William Spencer Percival and W. L. Horwood in 1939 as far as I know, but they used a different configuration and did not apply it to phono preamplifiers. See W. S. Percival, "An electrically "cold" resistance", The wireless engineer, vol. 16, May 1939, pages 237...240.

The schematics below show the resulting configurations. They are identical, but the left schematic is for people familiar with nullators and norators, the right schematic for people who feel more comfortable with high-gain twoports and op-amps. At frequencies well above 50 Hz, the input impedance approaches (R₁₃ + R₁₄)/(1 + R₁₃/R₁₂ + R₁₃/R₁₀). You can make this equal to 47 kΩ while using an R₁₄ that is much greater than 47 kΩ, thereby reducing the thermal noise current √(4kT∆f/R) that gets injected into the input.



You can't do this with op-amps (not without floating supplies anyway) because op-amps lack the negative output that conducts a (signal) current equal but opposite to the current through the positive output. That is, you can make electrically "cold" resistances with op-amps, but not as shown here.

Document about dimensioning the circuit
The attached zip file contains a pdf document that explains step-by-step how the component values were found (section 2) and that presents a more accurate method than I have used (one that doesn't need fine-tuning with a pole-zero extraction program, section 3). It also contains a spreadsheet for the more accurate way to calculate the component values.

Deriving the expressions was a nice exercise, but I'm not at all convinced that my more accurate calculation is of any practical use. It can very easily lead to negative or complex resistances.

I want to make an input selector, using Raspberry Zero and phono pre-amp, both jammed on a single board. Since I am IT guy, not an electrical engineer, can you PLEASE review my PCB design and let me know is this going to make it really bad and what should I change to make it work nicely?

This is a RIAA pre-amp part:


This one is an input selector, controlled with Raspberry, with connectors for switch and display:



Here is the entire board, top side first:


...and bottom side


THANK YOU IN ADVANCE!!!!

Marin

The past two weeks have been a black hole of DIY subwoofer obsession. With high-end audio gear costing more than a used car these days, I’ve started thinking—why not build my own

I’m toying with the idea of upgrading to a set of identical DIY subs in near future may be in a year. Is it a great idea or a potential money pit? No clue. But after walking through endless forum posts and YouTube rabbit holes (mostly featuring folks who think Dayton is the only driver on Earth), I’m officially ready to consult the real gurus—you guys.

The bass battleground:
13.5 ft wide x 21 ft long x 10 ft high
Its a Acoustically Treated Room (Thanks to Anthony Grimani Videos), I have M&K S150 LCRs, SVS Bookshelf Surround and Subwoofer i have two PB 2000 (Not Pro) at rear and single Rythmik FVX 15 in front

Here’s my current state of mind:

1. DIY subs: brilliant or bonkers?
2. My woodworking skills are… let’s just say Sharpening the pencil .
3. What good drivers are available in India? Dayton is everywhere, but I’ve got my eyes on Lavoce and BMS
4. 18-inch vs. 21-inch: is bigger really better?
5. WinISD is fun until it throws a “division by zero” tantrum. Can I really trust it to model things down to 20Hz?
6. Driver shortlist so far: Dayton UM1,Lavoce SAN214.50 (Refered fellow forum mate!) & BMS 18N862
7. Looks BMS is good.
8. Sealed or Ported which one to go.
9. And for designing Sealed do i need to depend Win ISD or i can go with the capacity details provided in OEM Site BMS says 150 Liter for Sealed Enclosure.

Goal: cinematic bass that punches you in the chest—just like the legendary JBL 4645C.

Oh, and fun fact—ChatGPT’s been throwing out suggestions like it’s got a subwoofer PhD. Honestly, it's like talking to an audio-obsessed buddy who never sleeps. 😄

I've been trying to figure out why there's car amplifiers that use 12-16vdc and produce 1,000+ watts rms easily but there's no easy solution to power class d amps boards in a car? I want to be able to power some class d amplifiers that require 32-36vdc in my car. If a car amplifier can do it why aren't there any power supplies available to boost 12v to 36v for this purpose? The only things I've seen even close to working are the cheap boost converters on eBay, Amazon, and AliExpress but the quality, lack of directions, and even worse the believability of the specs are less than ideal. I would have tried them anyway if I thought there was a chance they would work without introducing noise or other headaches.

Hello DiyAudio community,

TL;DR: This is a USB transport for audio output (up to 8ch) and input (2ch for now) and user device control (e.g., DSP management) from PC (MacOS/Linux/Windows) thru I2C/GPIO. Including bootloader for remote firmware update, flexible device configuration, HID interface, multiple audio output options, integration with SigmaStudio (simultanious audio playback and ADAU DSP configuration). Compact module available in USB-C and USB-B versions, isolated or non-isolated.

Latest docs, config tool, link to buy:
http://york.eclipsevl.org/

So, the story started when I needed a USB transport for audio output and another channel for device control, such as for managing DSP.

Last time, I solved this problem using a USB hub and a USB to Serial converter. This way, I assembled the required interface from 4 chips:
https://www.diyaudio.com/community/threads/audio-transport-module-york.406025/
(first post)

I tried to combine everything into one XMOS chip. This was also a working solution, but there was a big problem with the availability of the chips, especially in the desired package.

Thus, developing the transport on the pic32mz seemed the most logical to me. The first prototype module was based on Dortonyan’s project from Vegalab: in terms of circuitry, it almost completely repeated the original project and was designed for two-channel audio output, plus a serial interface. During testing of the firmware based on the original source code, several problems were identified when working in different OSs. The serial port was also not fully implemented.
I posted my version of the firmware with fixes in the original thread and on GitHub: https://github.com/eclipsevl/york_pic32mz_uac2_osfw/

After that, I continued working on the functions I needed. The following features were implemented:

1. Bootloader. Necessary for firmware updates via USB, without a programmer.
2. Flexible device configuration using a PC utility (which is also used for firmware updates).
3. HID interface for transmitting service information and device configuration. Initially, I planned to use the serial port, but HID is much more suitable.
4. Serial interface, which can be optionally enabled.
5. Various additional audio output options: s/pdif, 8-channel TDM, dual i2s (for using some DACs channel-by-channel in mono mode). Recently I also added support for old dacs with clock/data/latch interface. I.e. direct connection to dac ICs such as AD1862/PCM1702 etc is also possible without glue logic.
6. Various clocking options: module oscillators, external clock (slave mode), internal PLL clocking.
7. I2C port and library for integration with SigmaStudio: for simultaneous audio input/output and DSP ADAU programming.
This last feature is currently a unique combination and fits well into DSP projects: https://www.diyaudio.com/community/threads/freedsp-octavia.393804/page-18#post-7625685

The configuration and firmware update utility looks like this:

Device name and PID/VID settings


Audio interface settings:


Input of up to 2 PCM channels, up to 192kHz (384kHz with external clocking at 1024fs)
Output of 2 PCM channels, up to 192kHz
(384kHz with external clocking)
Output of 8 channels in TDM8 mode (up to 96kHz, clocked by PLL)
Output of 8 channels in 2xI2S mode
Output of 2 S/PDIF channels, up to 192kHz clocked by PLL



And of course, in the simplest version, this module can work similarly to any USB interface, in slave or master mode.
DSD has not been tested yet.


Photo of the module, second version.

The additional 4-pin connector can be configured as i2c/uart/gpio, and is also used for forced entry into the bootloader (in case of unsuccessful software update).



Available in two versions: USB-C and USB-B. The transport itself is quite compact: it fits into an area of about 2x2 cm - which is important for integration into other devices.
So I'm working on a smaller PCB with same functionality.

Later I'm going to release a software library to basically enable anyone to create their own app for PC to control custom devices: toggle GPIO of USB module/send I2C transactions.


The module is available with USB-B or USB-C, the price is 49 Euro.
For purchase PM me. Also available on Tindie:
Multichannel USB UAC2+HID+CDC interface York

I'm open for requests for additional features but whether it is going to be added to the firmware depends on hardware limitations and time required to implement it.

Latest version of config tool:
https://york.eclipsevl.org/york_config_tool.zip

P.S. Huge thanks to:
1. Alexey (Dortonyan) for publishing the project. His work laid the foundation, and without it, this project would never happen.
2. Beta testers of the first version: @CyberPit, @Ludilu

UPD Oct 28th 2024:
Documentation draft attached

UPD Feb 17th 2025:
Latest docs, config tool, link to buy:
http://york.eclipsevl.org/

I plan to use a Hypex SMPS400A400 to power a stereo AB100 integrated amplifier. My question is: has anyone added additional capactitance on the amplifier boards with this SMPS? I contacted Hypex and they just said "Our SMPS’s have sufficient capacitance for the power they can deliver. Adding extra capacitance may be possible but it is not required and not supported. Doing so will be at your own risk." Not very helpful. If you aren't familiar, the AB100 boards have places for four capacitors on the board. I have 4700uF 80V capacitors to put in those spots if it can handle it.