Today, I listened JBL Paragon short video and sound was 5X greater than the youtube videos. I have bose speakers connected to pc. Than I looked JBL Paragon drawings and there was curved wall in front of it. I thought I cant buy the parts , wood etc but I can make a plastic curved wall with painted with epoxy paint. I am poor and can afford solid 2mm thick polypropylene sheet with 10 foot to 5 foot and can cut it in two to make the curved walls for cabins in house. I am thinking to paint or spray epoxy on reflecting surface to cut the loss of sound. I found another solution to keep the curve easy and controllable with sailors rope and tighting it for two edges, final shape will be a big C.
I am thinking these walls inspired by architectural curved standalone park concert acoustical walls. What do you think ? Or I can use single curve and turn the speakers to the wall at the center or two speaker with stereo angles. What should one to do to make this wall is a real improvement ? I will use lower volume levels.

Mustafa Umut Sarac
Istanbul

Hi all!

After reading this forum and playing a guitar for a few months, I think I know a bit better what I want in terms of simple home studio of sorts. I'm asking for a sanity check and advice. I am looking for an inexpensive and efficient set of equipment - I am not opposed to spending more if it goes anywhere I understand but generally I want to stay lean.

I am mainly playing guitar (strat) by myself or with my daughter on bass (or she plays by herself). We also have an acoustic with a piezo pickup, a son/brother, who occasionally plays electric piano with us, and we will add a mic. All this in a living room.

• I want to have instruments feeding into some pedals, then a mixer, which goes to the monitors.
  
  
• We want to record/layer too, so it makes sense to have a mixer with a computer interface.
  • But I do not want to be tied to a computer and to have to power it up every time I want to play a couple of chords, so the mixer and monitors should work without the computer too.
  • It would be convenient to also be able to save tracks and layer on the console, without the computer - but perhaps this is too much to ask from an inexpensive mixer: to be a mixer, interface, and а tracker (?) all in one. Or is it?
    .
• UPDATE: following the suggestions in the thread, I found another candidate. So, currently choosing between A&H ZEDi-10FX ($300) and Behringer XENYX X1204USB ($209).
• UPDATE 2: I chose in favor of A&H because both people here in the thread and reviewers on amazon raised reliability issues in the Behringer unit.
  
  ORIGINAL TEXT: I found this mixer, which seems to hit most points: Allen & Heath ZEDi 10FX. It has 2 hi-impedance inputs, two more "mic" inputs, and some additional line/stereo/usb inputs. It is $300 on Amazon - not exactly cheap but acceptable if nothing reasonable is available cheaper. There are some cheaper mixers like PYLE PMXU 83BT but they seem weird: questionable quality and no hi-Z inputs, just regular "mic". But they are twice cheaper.
  
  
• I plan to hook up the monitor output to a SS TPA3116 2.1 board, which I power with 24VDC.
  
  
• I am not sure what to use for the monitors. I have ELAC bookshelf speakers (DB-52), and an ELAC powered subwoofer SUB-1010 but perhaps they are too delicate? This calculation following @Gnobuddy 's explanations suggest that they should be fine but I am not entirely sure.
  • If these are no good, what could cheap alternatives be?
  • There is an option to use the now-famous Logitech Z-623 but this is a separate can of worms. 🙂
    .
• The computer setup is less important at this point - but I would like to use a Linux-based DAW if possible since this is the only OS I have in the house.

So... asking for advice on the above. Does it make sense? Are there any better candidates for the mixer/interface?

Thanks!

P.S. Oh, yeah, this is the full list of what I have and use now:

• A Squire strat with the bundled cheap amp, and this fuzz pedal
• A squire PJ bass, and this multipedal for it (preamp, compressor, fuzz, octave, XLR out, effect loop)
• A cheap acoustic with a piezo under-bridge pickup, and this multipedal for it (delay, reverb, preamp, chorus, XLR out)

UPDATE 3 (2023-09-13):
I've now acquired all I had in mind, huge thanks to the advice in this thread. Here's the full list of what I have (also see pictures):

• Allen & Heath Zedi-10FX mixer (with USB 4-channel interface)
• Simple Frontman 10 amplifier, came bundled with the Squier strat
• Using the guitar amp from the above + old Yamaha 8" subwoofer wired together as a monitor for live jamming at home (short-term, will upgrade)
• Sonicake bass multipedal, acoustic multipedal, fuzz; Flamma FS06 digital modeling preamp
• Squier PJ bass, two Squier single-coil strats (hardtail and tremolo), simple acoustic guitar (Johnson)
• SM57 mic (yet to be tried)

Guitars go into some combination of the pedals -> to the mixer -> guitar amp+Yamaha sub

-------------------------
Related threads:

• https://www.diyaudio.com/community/threads/ultimate-audio-interface-for-home-studio-workshop.396928/
• https://www.diyaudio.com/community/threads/instruments-amp-y4-trio.383281/

JFET Symmetric HP-Amp, LSK170A+LSJ74A

JFET input,
ZVN ZVP VAS,
IRF610+IRF9610 output.

Supply = +/-16VDC, 2x12VAC Transformer.

Simplified schematic here to begin with.

Some SD Parts for you fellow repair techs. Please note that these parts all work and are the correct parts that SD uses. Do not ask where I got them from as I will not be able to say.

Hi guys!

Any ideas on the smallest footprint Horn and Driver combo that can go down to 500 Hz?

Listening room is only 9ft x 9ft (hence I need something compact) so I think I can get away with crossing a compression driver (or fullranger) lower to meet a 12" woofer.

Will be using a minidsp for crossover duties and eq to tame the driver and horn.

I live in Southeast Asia and the only direct couriers we have are in the USA. I dont know if horns.pl or stereo-lab.de can ship affordably to asia. So a US supplier would be ideal.

Thanks

Got an email about this and thought somebody around this forum might be interested.

https://www.decwareproducts.com/dfh...l_lp&cid=1d8a1881-d694-4ba4-9c7c-040564b4f676

Much like this project, I'm an old dog, learning some new tricks. Not long ago, I posted part 1 of my redesign of a Magnavox A531 germanium console amplifier chassis into a stand-alone hi-fi component. It turned out amazingly well, one my very favorite amps because of the beautiful sound that germanium has. Perhaps it was the lingering intoxication of striking gold in the belly of a Magnavox console in that project that I just had to do it . . .

"Oh Geez!" my wife says, "He bought another one!"

Not a venerable 60's classic with a tube or germanium amp, but a sketchy, dubious, questionable-at-best solid state Magnavox R344 chassis in a very fine plastic and particle board offering from the golden year of 1978. It was nasty- left out in the weather, covered in dirt, missing 3 wheels and scraping on the floor, top all bubbled and chipped, and at least a dozen wrinkled trucker and cowboy themed 8-tracks rattled around inside. What was I thinking! Probably just wasted that twenty bucks . . .




I was tearing this console down for parts when I realized that the radio/amplifier chassis was actually designed to be built into a stand-alone cabinet. In fact it was used in multiple models mounted both vertically face up, and horizontally. In one model it came with it's own wooden cabinet and the whole thing slid down into a cubby in the console. Suddenly I was inspired by the fond THC soaked memories of high end amplifier designs I had seen in the 70's- with broad brown-toned wooden sides and top panels surrounding the shiny front face, I started thinking this might make a fun retro project (if IT didn't smoke up the joint when I tested it). I set the chassis up on my bench with some good test speakers, cleaned all of the pots and switches, checked everything over visually and gave it a try. The front panel lit up nicely, (I miss that) and it actually worked and sounded pretty good- not excellent, but good. It reminded me that it was the flimsy speakers and cheap sources that were the weak link in these old consoles, and as with the germanium amp chassis, the amp could be a lot better than it's surroundings. Maybe, just maybe if I made some improvements, this might be a fun little retro receiver to play with. Plus, It's cheap.

Amazingly, I found the Sams photofact for the R344xx chassis on eBay for only $7.50. (Hit me up if you need a copy.) The power amplifier portion was all on one modular plug-in card and heatsink, shown below. From my research into the literature, it appears to be about 12W per channel, using push-pull TO-220 package transistors on a 45V supply. Typical of a low-medium power amplifier, perfect for a single room or light listening (and about the same output as my Magnavox 9300 tube amp, and the germanium A531 amp). The bias adjustment trimmers were large black units next to the ceramic emitter resistors. The original output capacitors were an anemic 1000uF at 25V, I assume primarily limited by space on this small board. I upgraded to those dark blue 4700uF at 35V, and I can now hear a much better bass response at the lowest frequencies, with more punch, especially into my 4 Ohm speakers. You have to be careful going too big here too- This design does not have output relays, so I do get a bigger whump in the speakers when I power it on and off now. I considered putting some smaller value poly caps parallel the output electrolytics for higher frequencies, but I doubt my ears would hear any difference. I did reinforce the long thin traces for one speaker output, supply, and ground on the amp board, and made a placebo improvement in my mind at least.



After the few key capacitor upgrades, I pulled every electrolytic capacitor one by one and checked it. I was at first inclined to change them all, but honestly it was not worth the expense to buy all new, and my junk stash had many older capacitors that weren't necessarily guaranteed to live any longer than those already in there. These original caps were all Nichicon, so I only replaced the ones that were actually bad. My LCR meter does not measure ESR, but it does measure capacity and leakage current. I found no cap had leaked or bulged, and all were within spec and had very low electrical leakage (Go Nichicon!) In the end I found only one bad cap, the only green one right next to the hot power resistor as you would expect. It had been replaced before some time in the past, with little wonder.




This design runs everything from one single-ended 45V supply, with hot dropping resistors to generate a slightly lower bias supply, the 22V for the radio, and several others. Cheap and simple if you don't mind the heat, effective until it's not. Dial lamps come from a 6V winding. The primary filter capacitor was originally a tall aluminum can of 2000uF. It was still good, but I replaced that with the modern 4700uF of higher voltage that you see there in dark blue. I could have used a much larger value, but with diminishing returns- Even with the 2000uF there was no audible hum at maximum volume with this unregulated supply, and I really did not want to overtax the small poorly ventilated transformer with higher rectifier current peaks for little return. 4700uF should give improved dynamic response to the bass with little downside.



Lots of other tweaks were necessary- The boards had many thin and broken solder joints from thermal expansion- particularly near hot resistors, places where shields or metal frame was soldered to the ground, and around the board to board connectors. I wire bridged across the two connectors to the radio board rather than remove them, and opted to just clean the connector to the vertical amp board so I can continue to remove it for servicing. I removed the input and output plug-in PCB connectors and soldered leads directly to the board. I replaced a dozen or so ceramic capacitors that were in the audio path through the volume and tone controls with poly type capacitors. The larger brown ones matched the wider lead spacing and are visible on top, and smaller flat green chiclets are mounted flat pad to pad on the bottom as their lead spacing was much smaller. This too made a good improvement in the sound quality- better low bass from some increased values in signal coupling, and less distortion from the improved linearity of poly over ceramic in all of the filtering roles. I removed the laughable "RIAA-ish" filter used for the ceramic cartridge phono input and converted it to a third "aux" input. I had quite a chuckle when I discovered that the headphone jack was wired L-R backwards. (I doubt anybody noticed.)

The sound is now much more transparent, and at least on par in quality with many of the lower-end integrated amplifiers I have listened to in my life. The adjustment range in the tone controls is way too large for my taste (from super boomy to thin and flimsy), but when centered are reasonably flat. The loudness control is far too excessive- definitely cakes on the thick and heavy chocolate- it will probably never be used, but that was the Magnavox console signature. The AM and FM radio come in strong and clear with a nice red stereo LED. Interesting having separate selector options for FM and FM-Stereo. I have three inputs, Aux, Tape, and Phono. With all of the dial lamps replaced it has a nice blue glow with a lit tuning meter.

I built the cabinet from scraps I had around my garage. 3/8" plywood for the bottom with ventilation holes in key hot areas, and oak sides cut from an old cabinet door. The top panel and perforated waffle-board back came from a 1961 console- back when they used hardwood veneer on the sliding top. The big and only expense after the photofact was the Chinese RCA input and speaker binding posts, and screws from Walmart. The stain on the sides is dark walnut Watco Danish Oil - love that stuff. It definitely gives it that 70's wooden side look I was trying for. It badly needs labels on the back for all of the inputs and outputs. If anybody could suggest a reasonably easy way to do that I'm all ears.

Anyway, give it a look. It sounds great, it has a warm funk-a-lishous chocolate vibe (Ron Burgundy approved) It's a lot of fun for a receiver I have less than $50 invested in. What it lacks in pedigree or quality, it makes up for in soul. It is fun! Really takes me back. I can see this taking an honored place in my den, driving some equally vintage 3-way speakers. Maybe I'll don a smoking jacket, and sip some fine brandy . . .

Many here have jumped onto the M2 band wagen, but few have paid attention to the opto-biased output stage in detail.
We have, for example, discussed the use of different MOSFET pairs.
https://www.diyaudio.com/community/threads/complementary-power-mosfets.378024/post-6889027

Nelson has previously dropped some hints that it works well also in Class A/B.
But how ?

This is an attempt to analyse precisely that, using LT Spice simulations.
It is a waste, IMHO, not to use this clever OPS design in more applications.
For example, how about in a WHAMMY Power amp ?


Patrick

.

Recently acquired on of these beasts, been lusting after one since I first read of their existence back in the early 90's. It arrived with all 4 power supply bipolar transistors blown open, rather than sourcing replacements I converted it to use TO-220 packaged logic level MOSFETs, for reliability's sake. It now works well but I can't seem to locate a manual online for it. Milbert's website says to email them and they will provide one, but I haven't heard anything in nearly a week's time and their phone number isn't working either. I know it's a long shot, but has anyone had any interaction with them or have they slowly faded away?

Moderation Edit: For updated information and new links regarding this project, please refer to post #2386.

New Links and Information

My latest audiophile approach named Q17.
So far the best sounding amplifier I ever made. Harmonic reach with exceptional bass response and 3d soundstage.
Archive contain everything. Full amplifier description and operation. Ltspice asc file for simulation. KiCad source for Q17 and his power supply. Gerbers zipped, so you may pick and drop on your favourite PCB manufacture. BOM with current active parts for Q17 and power supply.
Detailed description on how I made the PCB inductor and more, is also incuded in zip.
Inside zip:
LTSpice - contain asc file
ps - power supply KiCad source files
Q17 - amplifier KiCad source files
Spiki - inductor generator - python script to generate inductors in KiCad
Q17project_presentation.pdf - detailed presentation - bill of materials included
Q17 is on GitHub under GPL3 - Q17 is an open-source hardware project
ngspice simulation
Measurements
How to order your transformer
How to order your own PCB from JLCPCB, or click & order your Q17 with dual output pair PCB @
You have extra money and wanna buy directly from me, subscribe to this GB list.
Purchase parts for one channel with one click from mouser.
Purchase parts for active power supply from mouser. Does not include synchronous rectifiers and optional decoupling MKP.

Enjoy !


Edit 23.July.2021 If you build this amplifier, please give him a serious burn-in. All these mosfet's will start to sing after 200-300 hours.
You may download latest KiCad for your OS from >>> KiCad Downloads

Edit 25.July.2021 GerberQ17.zip is for amplifier pcb and gerberPS.zip for power supply.
Upload each zip file separately to jlcpcb. Jlcpcb will process one file at a time as separate orders.

Edit 26.July.2021 There is an error on output connector on GerberQ17.zip. Don't use this file until I correct this.

Edit 27.July.2021 Output connector issue is cleared. See this post for latest archive > Q17 - an audiophile approach to perfect sound

Edit 12.Aug.2021 Mounting holes have been added to power supply pcb. Gerbers available here >> Q17 - an audiophile approach to perfect sound

SO the band Im in have been using this little Fender Passport PA for practice with good results until recently it stopped working.
When powered on the protect light just lights up and no sound comes out. It does this with nothing plugged in as well (speakers mics). What would be a good starting point to try and repair it?
Thanks!

Starting to come here more and more to expand my horizons further. Who else is here?

Hi guys,
I recently got a pair of Fostex H400 horns.
They were in bad condition.

So I decided to make them pretty again and use them with a pair of Beyma CD11Nd or CP385Nd which I have at hand.
This is what came out



For the moment I paired them with my PPD Subs. Just to hear what the horns are capable of.

The PPD subs are supposed to play up to 300-400hz. So this is by far no perfect match crossed at 1k.

But I really like what I hear from the horns

Hi

I have always been a fan of these mono block amplifiers via velleman .. both the el34 and kt88 version.
Velleman have long since stopped production of these years ago .. I purchased their last 27 boards they had in stock. I have now since decided to make my own new production boards (Velleman where ok) but the issue was that these boards where made before Gerber files.

So after some hard work I produced the artwork the old fashion way then painstakingly converted to a full Gerber file so I can get them made again.

My new boards are fully drilled with 2oz copper tracks. This is the 1st production board being populated, with the wire links 1st .. I will update more on the progress of the first board. This will be A 90w EL34 version. The new PCB boards are thicker as well so no flex ..

Does anyone have schematics or information on VTL De Luxe 120 mono blocks or VTL The Ultimate Preamp?

Thanks,
Brandon

Who's ordered @Gabster 2000 's TD1 board? I ordered two so I'm curious who else is building this one?

https://gabster.ca/Gabster-TD1/

Also known as Cubo HT is designed for Home Theater and low bass songs (B&B Bass Boosted). It extends into the realm of infrasound, frequencies below 20 Hz that are felt rather than heard, nothing too much just enough to justify its name. Experience (truly) scary low sound effects in movies or voice modulating bass lines in songs. Lots of amplifier headroom is advised but only a few hundred Watts is needed to get you started. Full plans and marketing talk here: freespeakerplans

The current plan for Cubo Infra is the 3th prototype, the measurements are from the first, although little has changed. An Imperial version is in the making, plans available on request.

The design allows it to be placed with the driver aimed towards its audience (or any other direction), with rubber feet of 2 cm (4/5”) height it can be used in a down-firing position. This maximizes the effective front chamber volume, creating an acoustical low pass filter that makes it sound like it drops lower and it can also mask mechanical driver noises.

The measurements below show the response in a corner*, in down-firing position, with a 24 dB/ octave Linkwitz-Riley low pass at 60 Hz or 80 Hz without a high pass. Based on hearing alone the response drops off quickly below 17 Hz. Of course measurements inside a room are very sensitive to sub placement, listening position and the dimensions of the room.

* One thick stone wall, one thin stone wall and a wooden floor

Front view:


Side view:


Down-firing:


Construction plan:


Cut Sheet:


Measurements:


Magenta: Microphone in close proximity, 24 dB/ octave low pass filter at 80 Hz
Purple: Microphone in close proximity, 24 dB/ octave low pass filter at 60 Hz
Turquoise: Microphone at 1 meter, 24 dB/ octave low pass filter at 60 Hz

Best regards Johan (a.k.a. Cubo)

Hi everyone!

I am a hobby guy, who is for ever trying to learn how to build design and build electronic circuits for any kind of different applications. Typically, I will have some idea of something I'd like, and then see if I'm able to actually make it work.

My projects typically revolve around microcontroller projects, I'm a fan of ESP devices (and old ATMEL chips).

Recently, I have also started looking into analog electronics, where I'm trying to build sound/music circuits -- like fuzz for guitar, or echo for vocals (microphone).

I designed this amplifier. The original page is lost.
There are a lot of people asking for the parameters of my design. I re - update the web page.
To facilitate their acquisition

ScanaQuad SQ200 (IKALOGIC) 4-channels 200MHz
ScanaQuad (SQ) is a series of high performance 4 channels logic analyzers and digital pattern generators. They are designed to be your best companion when working on serial protocols like UART, SPI, I2C, 1-Wire, USB, I2S, CAN, LIN, RS232, RS485, and more. With ScanaQuad Logic Analyzers, you can capture signals, you can play them back, and you can even build genuine test signals and generate them!
Being able to capture and generate logic patterns simultaneously with the same tool, is extremely useful when debugging Logic signals, and it only comes with ScanaQuad.

LEGAL Information​

Complies with the following applicable European Directives: Electromagnetic Compatibility (EMC) Directive 2004/108/EC, Low-Voltage Directive 2006/95/EC, IEC61326-2.
RoHS Compliant 2011/65/EC. This device does not contain any of the substances in excess of the maximum concentration values (“MCVs”) defined in the EU RoHS Directive.
Complies with part 15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation.
CAN ICES-3 (B) / NMB-3 (B)
ScanaQuad (SQ) is a series of high performance 4 channels logic analyzers and digital pattern generators. They are designed to be your best companion when working on serial protocols like UART, SPI, I2C, 1-Wire, USB, I2S, CAN, LIN, RS232, RS485, and more. With ScanaQuad Logic Analyzers, you can capture signals, you can play them back, and you can even build genuine test signals and generate them!

Being able to capture and generate logic patterns simultaneously with the same tool, is extremely useful when debugging Logic signals, and it only comes with ScanaQuad.

All ScanaQuad devices have 4 channels. They can Capture digital signals, Generate digital patterns, or do both simultaneously.

UNIQUE FEATURES, UNIQUE PRICE TAG

MIXED MODE OPERATION

The ability to capture and generate signals simultaneously open a whole new set of perspectives when it comes to debugging your electronic project.

DECODE AND ANALYZE

Analyze protocols with software tools that gives you a global views of your data and helps you to identify bugs quickly.

ADVANCED TRIGGER

Enjoy a versatile trigger system that can target a specific UART byte, I2C address or any other sophisticated data pattern.

TRIGGER OUT capability to their logic analyzers!

As the name implies, it adds Trigger IN/OUT capability to the existing range of ScanaQuad logic analyzers. The way it works is quite simple: A special synchronization signal is exposed to the 5th pin of the USB connector, and a custom (5 wires) USB cable is used to get that sync signal to TrigBox. Adequate circuitry then provides deterministic propagation delays as well as 50 Ohm drive capability on the Trigger OUT line.

TrigBox allows ScanaQuad to sync with any existing instrument. For instance, you can use the ScanaQuad to trigger an oscilloscope on a specific protocol feature. Alternatively, you can use trigger IN feature of TrigBox to let ScanaQuad start capturing signals according to another instrument in your lab.

new danley sounds cohearix, a combination of paraline and meh

Greetings from Sweden
I am a retired engineer who is trying to realize my dream of building a pair of nice bass horns. I did this in my youth in the 80s but this time I thought of using Hornresp and AKABAK. However, it seems that these programs don't like me. We don't get along very well. Hope to get help with this here among all the experts.
See you.

Guys, im in love with that SKRAM design. Very anxious to have a box you can change vent plugs, goes like stink from 20 to 120hz. But the size!! I was wondering: is it possible to build a 20% smaller version with 18 driver? Im not technical enough to figure that out in Hornresp. 296 liter from 370. or any other config possible?

Also: I read its easier to build than the OTHORN yet im not a woodworker at all. the hours are ranging between 25 thereabouts for experienced guys to way over 60 for guys with no tools and such. Is CNC a 100% necesity? I mean if I could do it myself then yes, but if i have to hire someone that will cost thousands but also after its finished, if I want to sell it, passive DIY subwoofers like therse are hard tos ell and probably for a whole lot les than the bult evrer costed. What do you think?

EDIT I found a DIYaudio user saying this 2 years ago:""
For bass go 21" ...... lots of high grade options, with strong motors.

B&C 21ds115 works very well in undersized reflex enclosures due to very strong motor (BL 40!!!!).

Check data-bass.com , it has a wealth of independently measured data, also check the forums. There are several diy designs available there, such as the" ckram "(single 21" 30hz tuned reflex that can be modified to 6th order bandpass with a front attachment plate)
And "Skram" single 21" bandpass

15" has ~150sqin Sd, 18" has ~220sqin, 21" has ~300sqin

To get deep, you need a decently sized enclosure, so why not go all out on the driver to achieve higher power density!

30 Hz is a must nowadays, if you want to distinguish yourself & keep up.

The "fancy" european drivers are available on the us market, b&c, lavoce, 18sound, bms, beyma, rcf......""

Apparently there are SKRAM deisgn for smaller drivers? Does anyone know where all those files from DATA BASS.COM went? Im very fascinated and interested.

I got fed up with DIY power amps whose AC mains On-Off switch is on the rear panel. This choice does reduce the amount of metal working required; just buy an all-in-one plastic molded unit containing IEC inlet jack, fuse holder, and AC mains rated switch. Cut one rectangular hole in the rear panel (where small mistakes in drilling or filing are not visible!), drop in the unit, done! But reaching through the equipment rack to operate a switch on the back panel is clumsy to watch, and awkward to do. I wanted something a little more refined.

Front panel switches with a circular pushbutton and colored illumination looked promising; the type called "Anti-Vandal" appealed to me. Figure 1 shows a few examples of them. UNFORTUNATELY, most of these switches are rated for low current DC applications like soda vending machines; very few are rated for AC mains switching at the high currents needed by power amplifiers. Worse yet, the inrush current that flows through the On-Off switch at the instant when the gear is turned on, can be enormous. Far too high for a switch rated 2A @ 24VDC. Figure 2 shows the measured AC mains current which flows, when a rather modest chipamp-based power amp (photo) is turned on. The inrush current peak is 24 amperes: (2.4 vertical div X 10 Amps per div). That's quite a stress for everything in the AC mains current path, including the On-Off switch's contacts. And remember, this is not some super-amp behemoth from Krell; it's just an Antek 200VA toroid, two LM3886 amp channel boards, and a diyAudio Universal Power Supply PCB. Yet its inrush current is frighteningly large.

It seemed to me the only way to use the Anti-Vandal On-Off switches I liked, was to use them at low current & low voltage. They wouldn't directly switch the mains; instead they would merely control a separate big, robust, high-current and high-AC-voltage switch. Perhaps such as a mains rated relay or triac.

Regrettably, more than half of these Anti-Vandal switches are momentary pushbuttons; they don't have latch-in-position action like rocker switches or toggle switches. To use them on the front panel of a power amp, an electronic latch or flipflop would be needed, giving "push on, push off" behavior.


SMALL LIGHTWEIGHT FRONT PANEL SWITCHES

Nelson Pass's 1992 amplifier DIY project called A75 (class-A, 75W/ch) used a triac to switch the AC mains off and on. The triac was controlled by a small front panel switch. The circuit reduced switch current to single digit milliamps, but it did require a front panel switch that can withstand the full AC mains voltage (230V in many countries). Very few of the Anti-Vandal switches meet this requirement, unfortunately.

Canadian high-end power amp manufacturer Bryston used an always-on, micropower, low voltage DC supply for their front panel power switch. It fires a triac, and the triac switches the AC mains on and off.

Australian engineer and prolific DIY project creator Rod Elliott offers Project 166 which is a Push-On, Push-Off Mains Switch. It includes an always-on, micropower, 12V DC supply. This powers a logic flipflop circuit which is the memory element for the push-push function. The flipflop operates a 12V relay, whose hefty contacts switch the AC mains on and off.


INRUSH CURRENT LIMITING WITH BYPASS

The measured inrush current waveform in Figure 2 indicates to me that some type of inrush current limiting is certainly needed, no matter how beefy and robust an on-off relay or triac is used. I still want to avoid over-stress of the transformer, the mains fuse, the internal wiring, etc.

I decided to also include a bypass circuit, which completely removes the inrush current limiter a short time after power on, when mains current has settled down to its normal (non-inrush) condition. Both Douglas Self's textbook and also Bob Cordell's textbook recommend inrush bypass circuits, on their page numbers 633 and 459, respectively. Bypassing the ICL reduces its current to zero, so it cools down and becomes ready for the next power-on event (a so-called "hot restart"). Bypassing the ICL also reduces the voltage dropped across it (to zero), which can be significant when the rectifier current-peaks are large. See Cordell p.459 for more.

The basic idea of inrush current limiting with bypass is sketched in Figure 3. Switch "SW1" turns the amplifier On and Off. Series resistor "NTC" performs the function of inrush current limiting. It guarantees that mains current cannot possibly exceed (Vmains / R_NTC) even if the rest of the primary circuit is an ideal short circuit. Switch "K1" bypasses the inrush current limiter after the inrush event has completed and mains current is down to its normal levels.

Inrush limiters have been successfully deployed using (a) ordinary wirewound resistors [Self], and also (b) specialized Inrush Current Limiter components [Cordell]. I decided to use option (b), the special ICL components. They offer a degree of extra safety in case the bypass mechanism fails open-circuit: an ICL can withstand the full primary current indefinitely. Whereas (a) wirewound resistors would self-destruct in less than one minute.

In construction, ICLs are enormous Negative Temperature Coefficient thermistors; a famous example is the CL-60, whose very part number includes CL for Current Limiter. I prefer not to refer to these as thermistors, to avoid confusion with the tiny little temperature sensors that also happen to have a negative temperature coefficient. So I will make an effort to always say "Inrush Current Limiter". Oh by the way, that happens to be the name that both Digikey and Mouser give to the section of their website where these components are sold. Forget Thermistor, remember Inrush Current Limiter.

When should the inrush limiting cease? When should the bypass turn on? Opinions vary. Texas Instruments bypasses the ICL when the amplifier's DC power rails have ramped up to approx. 60% of their final voltage (Fig.5 on p.7). Self suggests bypassing 1 second after power-on (p. 632). Rod Elliott's inrush current limiter for power amps, Project 39, enables the bypass after 0.1 seconds. A brief and incomplete survey of diyAudio discussion threads, turns up recommendations varying between 10 full cycles of the AC mains (0.2 sec) and 100 cycles of the mains (2.0 sec). All of this suggests to me that there are probably more than one right answer, and maybe the shrewd decision is to provide a few user-selectable options. As has previously been done in the Soft As A Feather Pillow soft start design by diyAudio members jhofland and xrk971, I chose the convenient 2X-per-step settings 0.5 sec, 1.0 sec, 2.0 sec. Thus bypass happens either 0.5, 1.0, or 2.0 seconds after power on, selected by a jumper on the PCB.


CIRCUIT SCHEMATIC DIAGRAM

Figure 4 is the schematic of this PCB. The power handling "business end" of the circuit in on the bottom half of the page; the top half is merely timing and LED management.

The actual front panel switch connects at top left; it switches only 5V DC at 0.5 milliamperes. This ripples through some digital logic gates and produces a digital signal "TRIAC_B", which turns on triac T1 through opto isolator U4. When T1 turns on, it applies mains power to the transformer primary windings. Notice that triac T1 is series connected with an Inrush Current Limiter device, "ICL1".

Component ICL1 a 22mm diameter device made by Ametherm, whose surge energy rating is 125 Joules. For 115V applications I recommend part number SL22-20007, whose cold resistance is 20 ohms and whose max continuous current is 7 amps. Thus the inrush current when the ICL is at room temperature, cannot exceed (115 / 20) = 5.8 amps, and that assumes the transformer primary, mains fuse, and AC wiring all add up to 0.00 ohms of additional resistance. [For 230V applications I recommend the SL22-50004, whose cold resistance is 50 ohms and whose max continuous current is 4 amps.]

After a user-selected delay time (chosen by setting a jumper across one of 3 sets of pins on header P7), the inrush current limiting is bypassed. Digital logic signal BYPASS turns on MOSFET Q4, which energizes the relay RL1 and shorts out the (triac + ICL) devices. Now there is no extra resistance inserted; the transformer primary is connected directly to the AC mains.

The relay, the indicator LEDs, and all of the digital logic are powered from a tiny AC-to-DC converter module, U2. Today these are available which meet the sub-0.5W international requirements for standby power, and they are delightfully small and cheap. The big electronics distributors carry modules made by RECOM, Mean Well, CUI, and TDK-Lambda. However the TDK modules are quite expensive (3X!), and the CUI modules are in stock at DigiKey but not Mouser. Since Mean Well appears to have by far the greatest number units in stock and on the shelf at Mouser, that's the one I selected. Eventually I settled upon a 5V model, after realizing (a) 12V relay coils don't consume any less power than 5V relay coils; and (b) LED circuits actually do consume more power when operated from a 12V supply than from 5V. It's only obvious in retrospect. And that's why the board uses CD4000 logic chips -- they work at 12V (and 5V, and 3V, and 18V)

The AC/DC module output is made available at connector P5, in case other circuits within the chassis may need a low voltage housekeeping supply. For example, a "breathing" style LED. I strongly urge you not to connect either of the housekeeping supply terminals (5V_HOT, 5V_COLD) to the ground pin of other circuits. Treat it as its own, self-contained, floating 5V supply. If interfaces are required, use optoisolators between circuits connected to the housekeeping supply, and audio circuits connected to audio ground. That way you cannot pollute audio ground with noise or garbage from the AC-to-DC module. The module is rated for 400mA but I recommend drawing less than 300mA from the P5 connector, to avoid overload.

To soothe my paranoia about electric shock, I've included a 25 ampere bridge rectifier "BR1" between 5V_COLD and chassis protective earth. Now in the very unlikely event that the (class 2!!) AC/DC module develops a short between AC mains LINE and DC output Minus, the mains voltage is shunted straight to protective earth through a very high current pathway: BR1. Even if somebody had connected 5V_COLD to circuit ground, against my recommendations, it still gets clamped to a non-dangerous voltage through BR1.

A couple more details about the circuits connected to the front panel switch. Firstly, a very long debouncing time is conservatively applied (using resistor R2 and capacitor C1), compared to the measured timings of typical switch-bounce events, as reported in The Ganssle Group's debouncing article. After that, a Schmitt trigger (U3A) adds hysteresis. These two mechanisms filter out and remove switch bounces very effectively, and present clean square waves to the downstream logic. Secondly, flipflop U1B is a memory element which remembers the current logical state (am I on, or am I off?) when the front panel switch is a momentary contact device. The PCB user installs a shorting bar across pin header P6 when using a momentary switch on the front panel; this enables U1B. If there is no shorting bar across P6, U1B is disconnected and the non-momentary front panel switch controls On-Off behavior and timing.


STATE TRANSITION DIAGRAM \ TIMING DIAGRAM

Figure 5 illustrates the sequence of events that occur when the equipment is turned On, and when the equipment is turned Off. As you can see, all delays are set by RC timeconstants on the PCB. Ambitious, self-confident DIY builders can modify these timeconstants, if desired, simply by changing component values.


INRUSH CURRENT: BEFORE AND AFTER INSTALLING SOFT START

Figure 2 is the "before" measurement; it shows a stereo chip amp's power on event with no soft start applied. Figures 6 and 7 show the same amplifier, but with this PCB installed between the transformer primary and the AC mains.

In both Figure 6 and Figure 7, the inrush current has been reduced quite dramatically. "Before", with no soft start (Figure 2), inrush current peak was 24 amps. "After", with this PCB installed, inrush current peak was only 4.4 amps (2.2 vertical divisions X 2 Amps / div). (The horizontal scale is 100 msec / div as reported at top left in the Figures.)

In Figure 6, a shorting bar has been connected on header P7 from pin 1 to pin 2. This selects the "0.5 second delay" option for ICL bypass. You can see that when the ICL is bypassed, AC mains current jumps up. Because there is suddenly less total resistance in the transformer primary circuit.

In Figure 7, the shorting bar has been moved to pins 3 and 4 of header P7, which selects the "1.0 second delay" option for ICL bypass. Now the bypass comes later, and the up-jump of AC mains current is smaller.

These figures suggest, at least to me, that a setting of 1.0 second delay before bypass is desirable. And that's what I'll be using in my equipment. But other DIYers have the freedom to choose other settings, as they see fit. Thank goodness the jumper select options permit this easily.


CAUTION: I PREFER MY LEDs VERY DIMLY LIT ; YOU MAY NOT

Resistors R3 and R4 set the current flowing in the power-is-on and/or the power-is-off LEDs. I chose quite large resistances for these because I like dimly lit LEDs. You may want to experiment with the LEDs you plan to use, and to discover the resistor values which give the LED brightness that you prefer, with 5V supply voltage. Do this before stuffing and soldering them into the PCB, of course.


(optional tech info): MORE ABOUT INRUSH LIMITER JOULE RATING

The Ametherm ICL I have recommended in the Parts List, is rated for 125 Joules. Is this too little? Too much? Just right? Let's look at a couple typical scenarios.

Example 1: a chipamp with plus/minus 36V supply rails, using a diyAudio Universal Power Supply, whose PCB has a total of eight electrolytic capacitors. Each capacitor is 15,000 microfarads with a 50V voltage rating. Four of these capacitors filter the +36V supply, and four of them filter the -36V supply.

The energy stored in each of the four V+ capacitors, is 0.5 * C * V * V. Plugging in numbers, (0.5 * 1.5E-2 * (+36) * (+36)) = 9.72 Joules. Since there are four V+ capacitors, that's a total of 39 Joules for V+

Similarly the energy stored in each of the four V- capacitors is 9.72 Joules. {why? because (+36) times (+36) equals (-36) times (-36)} The total for V- is also 39 Joules.

Adding them together, the total energy stored in the PSU is 78 Joules. Which is comfortably below the 125J rating of the Ametherm ICL.

Example 2: the AB100 amplifier by Nelson Pass. This is a 100W/ch class AB amplifier with plus/minus 55V supply rails. Each rail has two electrolytic capacitors rated 10,000 microfarads and 63 WVDC. For each capacitor, E = 0.5 * C * V * V = 15.2 Joules. Since there are four capacitors, the total is 61 Joules. Comfortably below the 125J rating of the Ametherm ICL.

Although it's not written in the official published specifications on the seller's website, I have read here on diyAudio that the Intelligent Soft Start board from Neurochrome Audio, allows 250 Joules of inrush current energy (!!). Twice as much as the 125 Joules of this PCB. DIYers who believe they need more than 125 J (but less than 250 J) of inrush current energy, should consider the Neurochrome board to be a good possibility.


SUMMARY OF PCB CHARACTERISTICS \ FEATURES


AVAILABILITY

I have attached the Gerber CAD manufacturing files for this PCB; its unique identifier is H9KPXG . The suffix "-C" indicates revision C of the layout. Anyone can send the .zip archive of these Gerber files to a PCB fab and have boards made; the design is public domain and uncopyrighted. Use them however you wish. I hope if you end up with extra PCBs, you will consider giving them away for free, or selling them at low cost, to other diyAudio members.

I strongly recommend that you have these built using BOTH 2-ounce-copper {double thick} traces, and also ENIG {gold} finish on the pads. An example is shown in Figure 8.

As of today, 23 May 2020, I have about twenty extra boards (2oz, ENIG) which I am willing to sell at my cost: USD 3.50 plus postage. One board per customer. In the present conditions I will be shipping to US addresses only. The required US Customs paperwork and face-to-face handoff for international shipments is, I regret to say, unacceptable to me. If you live outside the US and want a PCB, send these Gerber files to a PCB fab and have them ship boards straight to you. Give away or sell any extras.

A parts list is attached below. If a diyAudio member wants to turn it into a Shared Shopping Cart at Mouser.com, please go ahead with my best wishes. Just remember that 115V builders need one ICL part# while 230V builders need a different ICL part#.

_

I open up this thread today because on the MPP thread we start to see modifications and subjective impressions of the beta builders. A lot more people will start to build the Parradise R3 soon and this may lead to confusion of what is the "official" version and what to shoot for. I have personally no problem when somebody wants to make a change and does like the change better. It`s your equipment and your music.
What i do like to happen is that a minimum of problems show up when people build the circuit as is. They have paid for and they should get the promised result with not too much frustration.
That should not distract from experimentation. When we find flaws in the original circuit or we can improve the performance there is a chance that someday an R4 version ( make that R5, 4 is an unlucky number in parts of Asia ) sees the light of day. It is my desire though that this kind of adventures should happen on this new thread. Than we we can clean up the MPP thread and make it more digestible for new contributors. That does not say that anybody polluted the thread, quite the contrary. It was and is moderated excellent by Salas and crew. I look on my own fingers as we say in Germany and i got many PM´s and such of people complaining that they lost the overview. It is a complex and fuzzy busyness for sure. that should be improved. Have fun.....

Last month at the Burning Amp Festival, I presented a little PCB called BJT Simple Matcher. That BAF talk, plus some photos, the matcher schematic, and the PCB Gerber file, are available on the diyAudio Forum (here). At the very end of the talk, I showed a little example circuit which greatly benefits from BJT matching, named "Yes It Can Drive an F4" (revision A).

A couple of small tweaks later the newest release (rev B), which I hope is the final release, is presented here in this thread. You can see a top view of its unpopulated PCB, and also the circuit schematic diagram, in the attachments below.

WHAT IS F4 AND WHY DO I MENTION IT HERE?

F4 is one of Nelson Pass's many FirstWatt amplifiers (link). Unusually, the F4's circuitry is arranged as a Unity Gain Buffer: although F4 provides enormous current gain, its voltage gain is only 1X. To get the maximum available output, 20V (peak), you must supply an input of 20V (peak). A large number of DACs, streaming devices, smart phones, and linestages are unable to swing their outputs to 20V (peak), so these are not a good match for the FirstWatt F4.

In fact, it is so rare to encounter a line level device which CAN swing its outputs to 20V (peak), that a snarky catch phrase has arisen here on diyAudio:
I'm pleased to report that the little example circuit presented here, gives an affirmative answer. Yes it can drive an F4. Yes it can swing its output to ±20V (peak).

The board is designed to be installed within an F4 chassis and to use the existing F4 DC rail voltages, with NO modifications of the F4 power supply or the F4 amp channel. Just bolt in the new boards, one per channel, and wire them up. Done.

DOUBLE DECKER / PIGGY BACK MOUNTING OPTION

This circuit board is 50mm X 90mm; its mounting holes are 40mm X 80mm. Which is a perfect match to the left hand (or right hand) set of four mounting holes on the F4 PCB: 40mm X 80mm. Using spacers, you can mount it directly atop an existing F4 PCB, in Piggy Back (Double Decker) fashion. Please see the attached photographs.

An F4 amplifier channel board (blue PCB) has a Yes It Can Drive An F4 (black PCB) mounted directly above. The spacers in this picture are 20mm tall (M3 female, hex body, M3 male) but Amazon also sells 25mm tall spacers if you prefer longer. The male ends of the spacers replace the M3 screws which secure the F4 amp channel to its heatsink.

Of course, builders are free to mount their Yes It Can Drive An F4 boards wherever they wish, in whatever orientation they like; there's certainly no requirement to use the piggy back mounting scheme.

YOU CAN ZIP-TIE Q5 AND Q9 FOR BEST THERMAL CONTACT

I've arranged the PCB footprints of the matched pair Q5-Q9, so you can zip tie them together (face to face) before stuffing and soldering. Personally, I solder all six legs and then cut off the zip tie, before I scrub the board with 99% isopropyl alcohol to remove flux and grunge. See the attached photo with a thin red arrowhead that points to the matched pair of transistors.

After I'm all finished with IPA and other solvents, then and only then do I apply a dab of heatsink thermal grease between Q5 and Q9. Finally I zip tie them back together again. This time: forever.


CIRCUIT DESIGN OF YES IT CAN DRIVE AN F4

The main goals of this board are (i) to use and to exploit tightly matched BJTs, thanks to the BJT Simple Matcher from BAF 2024; (ii) to provide high gain and wide output swing -- 20V (peak) -- without any shenanigans or funny business on the power supplies. Just use the same ±23V rails as the amp channel, and be able to swing > ±20V on the output.

Goal (i) is met by choosing Q5 and Q9 in the circuit, to be a pair of tightly matched BC546C devices. The schematic asks for matching within plus or minus 0.3 mV; but in the real world you'd just test 30 or 50 NPNs and then select the two pairs with best matching (one pair for the Left channel, one for the Right). If you can't quite get 0.3 millivolt matching, it's not the end of the earth. And if you can get better than 0.3 mV matching, how lovely for you.

Notice that there is only one capacitor in the signal path: input coupler C1, a high quality polypropylene film capacitor by Wurth. C1 blocks any DC offset that may be present on the input ("J6") from polluting the circuit and its output.

The circuit is designed to drive an F4, which includes a pair of AC coupling capacitors between the F4 input buffer and the (large) F4 push-pull output stage. Therefore I have omitted a coupling capacitor at the output of Yes It Can Drive An F4; one is already installed, on the F4 amp-channel PCB. If anyone plans to use Yes It Can Drive An F4 in a completely different application, without an F4 amp-channel, they ought to think very hard about whether or not to AC couple the YICDAF4 output.


CIRCUIT DESIGN: PART 2

Yes It Can Drive An F4 is a two stage amplifier; the first stage is (Q5+Q9) with (Q6+Q8) current mirror load. The second "voltage amplifying" stage is (Q10+Q11) with Q12 constant current load.

Because it's a two stage amplifier taking its output directly from the collector of the 2nd stage, the circuit is not able to drive a low impedance load. But that doesn't matter since the F4 amp channel board is a high impedance load. It's 100K ohms, which is easy to drive, even for this scrawny two stage amplifier.

Bias current for the two amplifying stages is provided by reference current generator Q3, which drives a first current mirror (Q4, Q7) for the first stage, and which also drives a second current mirror (Q4, Q12) for the second stage. The bias current flowing in Q3 is set by zener diode ZD1 and emitter resistor R6:

IcollectorQ3 = Iemitter = (Vzener - VBE) / R6

Plugging in numbers, Icollector of Q3 works out to be approx 2.7 milliamps. Oh by the way, after enduring the tenacious, obstinate, unrelenting "recommendation" of member @6L6, I eventually installed a pilot light LED "D1" in this 2.7mA current path. When power is applied and current is flowing, this LED lights up and reassures you that the circuit board is properly connected to the supplies.

The first stage's bias current is just a resistor ratio multiplication of the reference current

IcollectorQ7 = IcollectorQ4 * (R7 / R11)

In the same exact fashion, the second stage's bias current is another resistor ratio times Ireference

IcollectorQ12 = IcollectorQ4 * (R7 / R20)

It's illuminating to notice that the second stage transistors which drive the output (Q11, Q12), can EACH swing almost all the way to their respective supplies, VTOP and VBOT. Other designs for the second stage constant current source (Q12 here), cannot swing nearly as close to the CCS supply. The current mirror topology is the king of this particular hill.

I have specified the ultra fast and ultra low capacitance diode 1N4151 for position D2. Its part number is exactly 3 greater than the ubiquitous junkbox diode 1N4148, and indeed these two diodes are blood brothers. Diodes whose final eTest measurements are excellent, fall into the 1N4151 bin. And the rest of the diode batch, the average and the mediocre, fall into the 1N4148 bin. I think it's a Good Idea for serious DIYers to have a bag of ten or twenty 1N4151s in their parts cabinet, for those occasions when a super fast or super low capacitance diode, matters. But if you disagree, if you don't want to spend 20 cents on a new diode part number, by all means please feel free to use 1N4148s instead. The circuit's performance will suffer only a very small amount.

FOR EXPERT DESIGNERS: GAIN-BANDWIDTH, SLEW RATE, AND INDUCTOR L1

Yes It Can Drive An F4 uses an extremely well studied topology with (at first glance) completely standard Pole-Splitting "Miller" compensation. James Solomon's 1974 tutorial article (link) models it quite well; we can populate Solomon's equations by inspection, yielding the circuit's gain bandwidth product and slew rate:

GBWP = { 1/[(1/gmQ5)+R10] } / C5

SR = IcQ7 / C5

thus GBWP = 9.0E7 radians/sec = 14 MHz

and SR = 100 volts/microsecond.

Now let's consider the effect of the inductor L1 in the "tail" region of the Long Tailed Pair, Q5-Q9. This turns out to be quite an old idea; according to Scott Wurcer here on the DIYA Forums, Richard Burwen of Analog Devices used it in 1966(!) on the hybrid discrete opamp "ADI121", shown in the attachments below. Later, Deane Jensen was able to take out a patent (US 4,287,479) on the same circuit. I've snipped out "Figure 3" of Jensen's patent and attached it below.

At very low frequencies, the inductor acts like a short circuit, so the emitter degeneration resistors R10,R13 are removed from the circuit at low frequencies. This has two benefits: (i) the noise contributed by R10 and R13 is eliminated at low frequencies; and (ii) the effective transconductance of the long tailed pair stage is greatly increased. In this circuit, gm rises from 2.4 millisiemens to 52 millisiemens -- a growth of 20X (26 dB). There is 26 dB more gain available at low frequencies, thanks to the inductor. That's 26 dB more negative feedback for distortion reduction. And, oh by the way, the Gain-Bandwidth Product increases by this same factor of 20X at low frequencies. It rises from 14 MHz to 280 MHz. Cowabunga.

I encourage expert readers to derive expressions for "f1" and "f2" (the pole frequency and the zero frequency introduced by inductor L1). I think you'll be pleasantly surprised at how high these break frequencies actually are. I think you'll be delighted at how much extra gain / distortion reduction is made available at 20 kilohertz. All for the cost of a single inductor.

DON'T DO ANY OF THE FOLLOWING, THEY ARE BAD IDEAS

First: I have set the gain of Yes It Can Drive An F4 to 20 volts per volt (+26 dB). An input of 1 volt (peak) produces an output of 20 volts (peak), which is the maximum input signal allowed by an F4. Don't attempt to change the gain unless you really know what you're doing. If you need someone to help you change the gain, that means: you don't really know what you're doing. Stop now.

Second: Don't reduce the gain below four volts per volt (4.0X = +12 dB). If you do, there is a very good chance the circuit will oscillate, blowing your tweeters and perhaps also destroying the power transistors on your F4. If you need gain less than 4X, switch to a different design without this limitation.

Third: Don't put this circuit into an amplifier that is wildly different from the standard, stock F4 as documented by Nelson Pass. In particular, don't run at supply rail voltages greater than 29 volts. Neither the transistors nor the capacitors will survive at higher voltage; you will need to design a new circuit with different components and possibly different topology.

Fourth: Don't drive a load impedance less than about 25 Kohms. If you NEED to drive a lower impedance load, you will need a different circuit. This one is designed for the F4 which presents a high impedance load.

Fifth: Don't experiment with other values of inductance. Either use 680 microhenries (choose an inductor whose self resonance frequency is the highest Mouser will sell you), or else omit the inductor entirely and don't stuff or solder anything into the L1 footprint on the board.

i have a rotel power amp the mains transformer blew
i have a transformer of the correct voltage but only one output 40-0-40 the original had 2x 40-0-40 one for each amp board
the amp is bridgeable
what i would like to know is can i feed both amp boards which have on board rectifiers before the smoothing caps with just one output
or should i remove the rectifiers from each board and just use one to feed both boards schematic included
many thanks Glenn

Hello,

So I had this Adcom 5802 from which I got the enclosure, the transformer, the big caps in the ps filter(which were recently replaced and measure 4miliohm and 22000u) and the small power supply that controls the big toroid.

So I decided to use the enclosure to build a nice 20w/8r and 27w/4r.
The amp uses the Stasis Fe as a front end, accepts rca and xlr inputs and has a voltage gain of 10x.
The output stage is similar to the one in Sit 5 and uses same working principle.

Here how the pcbs look. There is the main pcb with the output stage and a slave pcb with the front end.


The amp looks like this
From outside(good shape cosmetically, can send more photos using direct messaging)


And here the inside


A few more photos with the amp





You will notice that I used the small 12v power supply to control the big toroid from the front switch.
I installed some temperature sensors connected directly to each sit capsule to monitor the temp meanwhile testing. If needed I can leave these in place else they can be removed though it won’t change a thing soundwise.
The sensors will help if you use the amp with improper cooling because once the capsule reaches 75C they will turn off the main trafo.

Here are a few thd graphs
1w/4r


1w/8r


27w/4r


20w/8r


The power supply is dual mono, each channel is powered from its own winding from the trafo.
Initially the Adcom trafo gave too much output voltage which I had to lower a bit so I took the trafo at a trafo winding factory near my town so they rewinded completely the secondaries. When I got the transformer back it made a bit of noise so I have to return it back for a 2nd attempt. Now the noise got lower but there still is some. I will askthem to make another attempt.
I didn`t get a new trafo because I wanted to be able to use the amp at 120/240v like it iniatially was designed and still can be used.
In the meantime that the trafo was at the factory I used smps power supplies with great success and practically no noise and very low cost(320w meanwell)

Because I want to go ahead with my projects I need to pull some funds from somewhere and this amp will save me this time. So don`t have other funds to put a new trafo inside.

I was thinking at the following..
I was planning to ask 1300e for the amp it the trafo would have worked as expected.
A trafo for this amp is around 250e at Toroidiy so what about 1000e for the amp as it is?
I will send it with the actual trafo.
The amp weights a bit more than 25kg. I can ship it in the UK and some countries in the EU(Austria, Italy, Germany, Belgium and probably others) at 1.2 pounds/euros. For the UK no custom taxes will be implied. So in these cases the shipping will be quite cheap.

If you have a pair of field coil drivers that that you want to trade I am all eyes and ears 🙂

As for the sound.. what I can tell is that it`s explosive! I built a few sit amps but none sounded so powerful like this one.


Edit:
Somebody asked how h2 looks so here we go
10w/8r


10w/4r

This amplifier has rather low distortion and is very fast.
All symmetric with JFET input and MOSFET output.
The supply is +/-12V because LSJ74 can only take 25V.
So max output is ~4.5 Watt which should be enough for most speakers.

Hi all,
is anyone using car-sound deadening materials in home construction to reduce vibrations?
I am talking about stuff like "Dynamat", "Noico" and many others, self adhesive material.

I have an interior basement wall that vibrates a lot due to HT subwoofer.
The wall is build of 1/8" panelling (the 70's style old house) on 6" studs.
It is a sound pressure created vibration, not direct transfer from the sub.

I have access to the wall from both sides and can get to the panels from the inside. I'd prefer not to change the panelling to drywall, hence my question about sound/vibration deadening materials.

Thanks in advance!

Call for testers!

Daphile is an audiophile class music server & player OS – targeted to dedicated headless PC.

Features

• Headless music server OS
• Bitperfect and gapless playback of common audio formats up to 192kHz/24bit
• Automatic configuration of music player for each audio device
• Multiplayer support (eg. multiple USB DACs)
• CD ripping with automatic metadata tagging and cover art
• Supports external file servers as music source (CIFS, NFS, sshfs & ftp)
• Easy configuration and installation through the web interface
• WiFi hotspot support (if compatible hardware exists)
• Software update via web interface
• Whole system image ISO-file is below 200MB

Daphile is based on the open source Squeezebox Server, SqueezePlay and Linux.

Get more info and downloads at www.daphile.com.

Any feedback is warmly welcomed!

Presented here is a small Class A single ended output headphone amp, unusual because it uses germanium devices.
Why germanium ? Well I have few of these wonderful old devices, and a heaphone amp seemed an ideal little project. There's another reason too... I am working on a new amp (eventually) and headphones are a great way of evaluating circuitry... so I wanted something good.
To anyone not used to seeing and working with germanium, well it probably looks a bit "all upside down" using a positive earth.

Let me say first off that this isn't by any means a "finished" design... it's still a work in progress.
It's 15 years or more since I did any serious headphone listening and my old Sony MDRV-7's were used in the design of this. I must look into more modern replacements.

The low supply voltage probably stands out, 6.8 volts... is it enough for decent output ? Well it appears so for me, and it drives the Sony's to way higher than what would be considered "safe" levels, although the efficiency of these old cans is high I think, compared to some. The supply can always be altered though, and in actual fact isn't the only limiting factor here. That comes from the constant current source Q4 and 5. The 3.3 ohm gives around 50ma in the outputs and at 6.8 volts they run at a reasonable temperature. This is germanium after all. Even swapping this resistor out to 43 ohms still allows decent levels before soft asymetrical clipping occurs.
One thing this design is if nothing else is tweakable... so non of these issues are a major stumbling block. As I say, it's not a final design yet.
The ACY21 is used as a buffer to provide a known input impedance, and to enable the volume control to be unaffected by the signal source... probably an unnecessary extra stage you might think, however listening tests reveal no problems including it... it may even help "soften" the presentation... in the nicest possible way of course. I tried direct onto the volume control, and then with the ACY21 and "think" the extra stage possibly adds a little indefinable something. Q2 and Q6 form a classic two stage amplifier. The supply (no decoupling shown) comes from a 7906 regulator with a 1n4148 lifting the ground leg to provide around 6.8 volts.

So how does it all sound... well I think it's pretty good, but then I would say that wouldn't I. It's certainly more "engaging" to listen to than a normal opamp offering... in fact it's pretty stunning. There is a razor sharp clarity, yet with absolutely no hint of harshness or stridency. I suspect the "not so low" output impedance is actually a match made in heaven for the phones, and playing around with the output impedance is something I want to look further into. The inherently slow (are they slow ? in absolute terms maybe) nature of germanium means that the amp has a natural and well defined HF limit. The 150 pf cap isn't strictly required... it satisfied the more objective aspects of the design ie squarewave testing, which made me feel better which in turn makes the amp sound better 🙂

So that's the design as it stands V1.0

No CD's were harmed in the development of this project. Material used included amongst many others,

Mozart-- Oboe Quartet in F major. Sony 88697285852

Nigel Kennedy-- Polish Spirit. EMI 094637993422

Elton John-- The Superior Sound of. DJM 8100622

Hello all! After reading many threads here for years, I wanted to give it a try. So, this is my first post and attempt at building speakers!

I love JBL speakers and horns, especially the 4349 and 4367. I have a relatively small living room, so I’ve decided to build a 60-80L vented box with a 12-inch horn and a 12-inch woofer. To match the directivity, I plan to cross the woofer between 700-900 Hz with an LR4. It will be an active speaker, so the crossover is digital. The F3 is important, but as long as it’s below 45 Hz, I probably won’t use DSP to boost it.

So far, I’ve found the Faital 12RS430 and Acoustic Elegance TD12S. I’m willing to pay a premium for two drivers, but I really want to do something special. What are some of the best woofers I can buy that suit my project? Especially with low THD.

Any help is greatly appreciated!

I built Doug Self's MM/Mc turntable preamp. This circuit is the Elektor premaplifier 2012 circuit, but there is an increasing noise parallel to the volume pot. As I open the volume potentiometer, the noise increases. I took the power supply completely out of the box and the problem has not improved. I replaced the signal cables with shielded coaxial cable and the problem persists. I am trying it as MM only. I have attached pictures and a video of the noise. Please help.

Hello everyone,
below there are pictures of my latest version of XMOS based USB to I2S interface. I've been working on it for some time now... drivers are from Thesycon and firmware from XMOS with slight changes to adapt it to my PCB needs. It's 2 channel, 24 bit / 192 khz with some improvements over XMOS' reference design:
* 4 layer, gold plated PCB with 1/2/2/1 oz (or 35/70/70/35 um) copper thickness. There are a lot of vias to minimize the impedance.
* there are two USB inputs available: USB-B connector (mounted on the board) and 5-pin external header for something like this.
* isolated (using GMR isolators) or nonisolated (6GHz microBNC) I2S output signals.
* seven LINEAR voltage regulators are present on the card: every oscillator (and the ICs in the digital audio signal's paths) have its own dedicated LDO (the three chips soldered on the botom side of the board).
* Heavy decoupling all over the places, including OsCONs SVP electrolytics.
* USB OR external PSU. There's one green Power LED indicator on the board.
* SPDIF output signal at 1:1 trafo.
* Two LEDs can be wired to L7 and L8 pin header heaving the following functions: "Host Active" and "Audio Stream".

The oscillators are from Fox but I do have 7 pairs of ultra low phase noise NDKs available for assembling. I must admit, NDKs are very hard to aquire due to the long time of delivering (up to three months!)

May 2013 Updates:
The WaveIO's pictures are outdated as I now use oscillators only from japanese manufacturer NDK. I'll upload new pictures soon to reflect the changes.
--------
To update WaveIO's firmware to handle 32 bit @ 384 KHz audio streams, please follow my instructions listed below (valid for Windows OSes).
From the beginning I would like to say that there are two ways to upload the .bin file but only one to revert the changes to defaults.

The easiest upload way would be via "TUSB Audio Control Panel" app:
1. First thing to check is if your WaveIO card does have DFU (Device Firmware Upgrade) implemented in firmware.
All boards following September 2012 should have this feature enabled.
a) Plug in your WaveIO card and make sure that is detected by your computer (drivers are installed).
b) Open "TUSBAudio Control Panel" located in "Luckit" installation folder and click on the "Device 0" text on the left side of main window.
c) If the RevisionID value listed there is greater or equal with 0x0331 (Hex) then you can move on to the next step. If not, then you'll have to
wait to finish the daughter board and I'll make both the hardware and software changes in the same time.

2. Download the file named "WaveIO_32Bit_384Fs.bin" on your computer.
3. To effectively upload the new firmware on your WaveIO you simply have to click on the "Firmware Upgrade" text on the left side, browse
for the file you have downloaded and hit the "Start Firmware Update" button.
4. The process should take up to 10 seconds after that you can easily check it by clicking again the "Device 0" text and see that the value for
"RevisionID" is changed to 0x0333 (Hex). You don't have to unplug your WaveIO board.
5. Done


LEDs: Since WaveIO hardware does not support two additional signals to drive LEDs for 352.8 and 384 KHz sample rates I thought that a
simple approach will be to display these freqs. by driving two LEDs simultaneously so:
* 352.8 fs will enable 88.2 & 176.4 LEDs while
* for 384 fs, 96 & 192 LEDs will be on.
If anyone has a better and simple idea please let me know!

Power: @ USB enumeration, WaveIO will report to be external powered device so it should work okay in both configurations: USB and
external powered. This measure was taken especially to work with iPad, iPhone or other portable/battery-powered devices.

That's it for now! I'll be back soon with the second uploading/revert_to_defaults option.

Kind regards,
L

(Thread title inspired by @Boden from @bwaslo's thread "Xsim-3D development... I could use some math help")

A little background first: I've done many loudspeaker measurements with the tool set that many DIYers have, a calibrated mic and some free software, but I'm looking for a better system. I've been looking more closely at Earl Geddes measurement method, and Gabriel Weinreich's paper "Method for measuring acoustic radiation fields" and what I can find on the function of Klippel's Near Field Scanner. Bwaslo's recent thread "Xsim-3D development... I could use some math help" took a turn that made me want to start getting some help on understanding all this. And like Bill's thread, I could use some math help...

In addition, a number of very interesting questions were rasied to Earl Geddes about more advanced measurement methods that I think could use their own thread.

So so far, I have two big questions I'd like some help with, please:
1. what is needed to turn a measurement set into a view of radiation modes (for lack better way to ask the question)? Geddes uses a FORTRAN routine, but would something like open source mathematical software work, or is it best to have purpose built software?

2. Weinreich's paper and the Near Field Scanner both are capable of "sound field separation", in other words, they can pull the room reflections out of the measurement and give you just the direct sound from the speaker (no need for an anechoic chamber!). Klippel only appears to do this up to 1kHz (plus or minus an octave) and uses measurement windowing above that, Weinreich didn't appear to us it very high in frequency either. It appears to work by looking at the timing of sound waves passing through a pair of measurement shells? But what I'd really like to know is if Weinreich could do that with technology from the late 1970's, can intrepid DIYers do the same with available software and hardware today?

And please, if anyone else has questions about the function of more advanced speaker measurement systems, please post them here.

Hi all,

might be iteresting, relatively new brand from Republic of Korea

https://ascilab.cafe24.com/shop2/company/about.html

https://audiosciencereview.com/forum/index.php?threads/ascilab-f6b-bookshelf-speaker-review.63387/

Amirm says best buy for the price of 835$ / pair

- best regards, Stefano

Hi,

Obviously a 8" midrange can be louder (and I'm guessing less harsh) than a 4" midrange, but how might the "speed/accuracy" (or transient response), especially in the high-mids, compare (appropriately-sized sealed chamber for each driver)?

More specifically, something like an 18 Sound 8M400 8 inch midrange vs. a Dayton Audio PC105 4 inch "full-range but you can use it as a midrange" (both 8 ohm versions).

The 4" is currently being run 500Hz-3.5KHz between a sealed-enclosed 12" woofer and the tweeter, but I'd want to run the 8" 200Hz-3.5KHz.

Looking forward to hearing back 🙂

Can someone explain how the output is grounded and what is grounded? How would this be wired up? I'm not getting how the output shorted doesn't fry the amp output.
Is it sending the speaker positive to ground? I looked at 3 other circuits and articles to make sense of it. One of the circuits in the store looks like it disconnects the + and - outputs entirely, but seems like people don't like the signal going through the relay. That correct?

Th protection circuit found in the diy store. https://diyaudiostore.com/products/nelsons-psu-filter

Hello, Okay I’m just getting around to the making ACAmini and it occurred to me would it be easily possible to eliminate the output capacitor and use a plus / minus bipolar supply? Has anyone gone down this road? Thanks Bob

I designed and built my current system during 2006, it is a WWMT with Scan-speak 21W8555-01 8" woofers, Accuton c2-6-78 midrange and Accuton c30-6-24 tweeter, height some 110cm. Due to other priorities i had to stop spending time on speaker design and thus remained some unfinished work on my speakers as of 2007.


Current set up in living-room, 5.3m wide and quite asymmetric, to the right there is another small room ;-)

A few of years ago i started to look into this matter of speaker design again, studied the publications on the subject (Toole, Griesinger, Johnston, diyaudio.com, etc) on design as well as sound reproduction and played with 6mm felt sheets to reduce the unwanted diffraction effects, not much improvement.
Last year just as a "just do it" step i took some absorption panel (cotton, shredded jeans, very difficult to cut), and quickly made something and put it on the speakers.
Some difference!, most striking was the timbre of voices, instruments, stayed much more constant across the room, also more involvement.


I got intrigued, and this spring re-assembled measurement rigs, made a turntable, and together with Arta and VituixCAD started to get experienced again in measuring and analysis. I decided to start basically all over again, be it with the assembled enclosure with drivers as a given. The passive crossover, and any physical adaptations for the edge diffraction, be it within the limitations of the front of the speaker, are to be engineered from scratch.

The objectives:
• Flat anechoic on-axis response, note: will require tilting the tweeter forward as baffle is tilted backwards.
• Smooth off-axis response & early reflections response and sound power & DI
• Distortion focus, be it with given drivers. Note: Filter choices can influence distortion.
• Spurious noises (already non-existent sofar, but the metal grilles are suspect)
• From my experience and listening preference:

This all gave me a starting point in my loudspeaker project, named:

GAYA2-Final: Finishing the unfinished after 15 years

As the name implies it is to finish what I started in 2005 and had to stop end of 2006, engineered to the current state of the art.
Key to being successful is the following: “Through measurements to knowledge” (door meten tot weten) , the quote of Heike Kamerlingh Onnes.

This principle is fundamental, therefore from the desktop research:

• Arta, Limps and Steps for measurements and some of the analysis.
• Calibrated microphone
• VituixCAD for the crossover design and simulation, taking a holistic view on and off-axis and step performance.
• And very important: Listening test, not only to correlate sims with the reality, also to achieve my perception based objectives.

And when done to incorporate what currently is possible for room corrections in the currently well-developed digital dimension of sound reproduction. I already have purchased Uli’s Acourate Pro and Mitch’s Hang Loose Convolver for this aspect.

The start:
I learned the tools by exercising measurements, study forum and article/book publications and perform analyses to fully understand them, before beginning with the real thing.
To cut a long story short: See also the thread Tmuikku started: https://www.diyaudio.com/community/...sover-and-tilt-experiment.388389/post-7097857

The outcome , quite humbling, is first of all the absorbing matter did make a difference, but also I have to get my measurements correct and repeatedly consistent, including getting rid of some DC error / very low frequency rubbish and the floor and ceiling reflection to get a decent gate in msec. Last but not least, my current baffle-shape for midrange and tweeter needs serious improvement (no surprises there 😉)

So back to the drawing board.

And not to forget enough space around the speaker during measurements, which means temporarily re-arranging the living room, thus only possible when I am alone and having enough time. 😉

GAYA2P, the purification completed

Sounds are propagating air pressure variations in a time continuum, recorded and processed by our complex audiology system and subsequently ‘heard and understood’ in our brain. Based on this fundament it implies for reproduction that a correct amplitude and time reproduction is needed.

This is a new thread, as successor to my rather long thread in DIYAUDIO: GAYA2-Final, finishing the unfinished after 15 years

The Goal: The Purification & Active filtering completed, a very fine connection to the soul of musicians as result.
The main triggers are: the progress in active filtering and of course the arrival of the PURIFI tweeter. Especially the upgrade to the PURIFI tweeter will imply some serious effort in measuring, comparison, publications, and perhaps some enclosure adaptation.

As shown in the previous thread, and basically most of that interesting Alice in Wonderland journey is done, already very happy with results, as is my wife!
So this final round is to complete the purification, the active filtering and remaining JDI’s in the entire system.

Next post will be a 'short' walk-through of the period from 2005 to June 2025.

Hi All,

Here is a stock of new low noise BJTs for sale (mostly) in pairs. They where bought from Mouser and Farnell some years ago:

2N4401 (130) + 2N4403 (130)

2N5087 (10) + 2N5088 (10)

BC327-40 (18) + BC337-40 (20)

KSA992 (20) + KSC1845 (20)

KSA1015GR (10) + KSA1015Y (10) + KSC1815Y (10)

MPSA18 (11)

Offers accepted.

Regards,
VS

Hi All,

Here is a stock of new BJTs for sale in pairs. They where bought from Mouser and Farnell some years ago:

50 V, 150 mA, 250 mW (Onsemi):

KSC945 G (10) + KSA733 G (10)
KSC945 Y (10) + KSA733 Y (10)

80 V, 500 mA, 625 mW (Onsemi):

MPSA06 (G, 10) + MPSA56 (G, 10)

120 V, 800 mA, 900 mW (Onsemi):

KSC2316Y (10) + KSA916Y (10)

160 V, 600 mA, 625 mW (Onsemi):

2N5551 (G, 10, TA 10)+ 2N5401 (G, 10)

300 V, 500 mA, 635 mW (Onsemi):

MPSA42 (G, 20) + MPSA92 (G, 20)

350-400 V, 500 mA, 625 mW (Onsemi):

2N6517 (TA, 20) + 2N6520 (TA, 20)

400 V 300 mA, 625 mW (Onsemi):

KSP44 (TA, 20; BU, 10) + KSP94 (TA, 18; BU, 10)

Offers accepted.

Regards,
VS

I gathered these few simple two transistor compound preamplifiers. Would like to know if anyone built some like it, how it worked, which are best performing.
Are there web pages dedicated in designing these Sziklai simple circuit with explaining what part does what?
I noticed some sport feedback cap from output, some don't.
Just want to learn some more. Plus i just optimized nice 3 tap loudness volume pot, which reduces signal to some degree, so instead of simple buffer, i could built simple nice pre.

Hello,

Had this one in my office for a while. I did a full recapping job, calibrated most of the things, per service manual, then ran into a bias problem and left it there.
6 months after picking it up where I stopped.

What I did:

1. all electrolytic capacitors changed​
2. both Thyristor replaced (were damaged)​
3. Supply voltage adjustment & check done and working
4. Adjustment of load impedance detection circuit done and working
5. DC balance calibrated and working
6. Protection circuit calibrated and working
7. Overload detection circuit calibrated and working
8. replaced output relay
9. cleaned controls and potentiometers

The problem, adjustment of Ico (bias).
The right channel is nicely biased @ 2mV, as the manual suggests. Everything works smoothly with it.
The left channel, oh boy oh boy... Looking at the schematic, my Q317 collector measures 0.5V, Q319 collector measures -3.3V, with respect to 0V. Everything after that is not even close to the values in service manual...
My thought, I have one channel working, and I don't have these transistors in stock, let's change transistors and diodes from one channel to the other and see.
I exchanged (left channel with right channel): Q317, Q319, D305, D307, D311, D309, Q321, Q331, Q333, Q327, Q329. Of course, I didn't replace everything just like that.
I check the bias adjustment after each exchange. The thing is the same... I also checked resistors, all good. I resoldered most of the left channel. Still the same. I measured all ceramic capacitors on the left channel, all good.

The amplifier works, it passed the signal, but after few watts of power, as expected, the left channel waveform (the bottom sine wave) starts to distort.

Any ideas? Any help will be appreciated. Maybe I'm missing something, or everything 😀

Ciao,
John

EDIT: VR301 is also replaced with new trimmer, Z301 power resistor has been replaced also. Schematic is attached, so you don't have to search through manual 😉 The voltages before bias circuit (Q301, Q303, Q309, Q313, Q315, Q311) are correct.

I saw one at a picture from BAF2016. So i googled this:

Pass Laboratories XA25

Read the description and look at the artwork picture. A new amp SE with hockeypucks. 700W 40 A 😀
Nice!

Cambridge CD2 manuals, schematics & mods

Does anyone have any information on the Cambridge CD2 manuals, schematics and modifications.

I intend to fit the Elso Kwak clock.
Also a common problem with the cd2 is with the display switching left to right.

I built this circuit back in the year 2002 and I used it as a driving stage for my class A power buffer PM-A1 (gain 1x, 30W/8ohm). It was working very well with quite good parameters, as can be seen below.



(D1 is a Zener diode 6V8)



I am showing above the THD+N plot (1kHz, BW40kHz) to include noise part as well. This is often omitted, unfortunately.

Loopgain stability is perfect, with 87° phase margin.

I salvaged a pair of mini piezos and was thinking of adding them as super-tweeters.
Do they need a cap? They were originally installed behind 2.2uF?
Do they need a resistor? What does the resistor do?

I came across these seemingly obscure drivers but could find very little about them. They look to be well engineered but outer beauty can be deceptive.

Anyone try these? How do they compare with other higher end cone drivers?

We discussed previously whether or not EnABL would improve a MAOP driver, and I volunteered my MAOP-7 to try it. Sorry, couldn't do it earlier. Preparations for winter occupied most of my time - I live in a countryside.

First, I tried to do tap test to determine optimal positions of the patterns. It didn't work - at any distance on the cone radius tap sound was the same. Tapping worked on a 4" Sony speaker with a paper cone, I could find positions where tap produced the loudest sound. I guess the MAOP cone is pretty well-damped as-is. Without tap test guidance, I decided to apply patterns 1) close to surround; 2) on former close to cone boundary, and 3) on top of the dome. See pictures for details.

The patterns were applied free-hand using extra fine Utilto acrylic paint markers, black and white. Surround pattern was applied to back side of the cone to conceal my ugly handiwork. To build up thickness (pens lay paint pretty thin), three layers were applied: black-white-black. On the cap, a single layer of white.

The difference showed on the first test record, Beatrice Rana Chopin Preludes. This is a very well recorded piano album. The second prelude is played between piano and pianissimo. Control untreated driver played soft, but with the EnABLed driver I could hear string buzz that wasn't apparent before. Those familiar with the sound of top level instruments like Steinway or Fazioli will understand what I am talking about.

Further testing is unnecessary. It worked.

Time for another set of bedroom speakers.
Of the SB tweeters available here in Oz which model comes closest to the old Vifa D27-45-06?
I just scored some NOS Vifa P-14s and I've got no suitable tweeter as easy to use in the stash.
Cost effectiveness is paramount so nothing from the Satori line

I'm building a subwoofer for the trunk of my Subaru Crosstrek.

I have 42.5in x 26in and would prefer the height be 2-4in. I'm having trouble with tuning of the box, as if the port is too wide the speakers go into over excursion, if they are thinner they easily go over 18m/s. I'm not sure how to model tapered waveguides or adding a flare in terms of port length. I'm a mech e but new to audio design, so joined here to learn as much as I could about complex analysis and subwoofer optimization.

I've tried sealed box, ported, 4th order (to basically act as a spring to reduce excursion, but I'm limited on tuning frequency because the back volume can't get any bigger without overexcursion), 6th order series and parallel, have played with 12in low profile sub, but don't know how to simulate downfiring and top must be encloseed as it will be the new trunk surface. The 12in is also pretty tall.

I'd be using an LX1300.1 with about 550W at a 2ohm load. Design shown is 6 Dayton Audio 2x7s. I'm trying to get low. I will have 2x 10in self powered el cheapos behind the driver and passenger seats, which will cover the 60-100hz range fine. That's what I did in the truck and it did the trick. I've never had low low bass and don't want to miss out on this opportunity with a brand new car.

I've attached a few pictures, but I've been through very many iterations, these are just some recent ones to show the process of building the subwoofer.

I'm also not opposed to passive radiators, but the only 2in or less ones are cheapo knockoffs with no TS parameters for excursion or moving mass or anything so I don't know how to begin that calculation.

We can't always afford polypropylene capacitors, much less
Teflon (tm), so we often make do with electrolytics when
we need those higher values.

There's all sorts of caps out there with claims to being better,
and they certainly are more expensive.

My favorite is the Elna silk capacitors, available from Digikey,
and really cheap. The measure spectacularly, sound great,
and the manufacturer's translated description of why they
sound better is a Babelfish classic.

So before you run out and spend a lot on the highly hyped
spendy caps, try the Elna silk caps.

There's hemp in them, too.

😎

The post in the newcomer reporting area has been closed

This device is purely a personal DIY, and all the components come from online shopping. There is only one device, and I am not some kind of equipment supplier.

Basic idea:
TPA3255(600w) x 6 + 50V1000W + 15V+/- Switching Power + LT3045/LT3094 + 7*JRC5532DD + miniPC + Terratec 7.1USB sound card + Roon Server

Hi There
I have this amp which just stopped working. No Power.
I opened it up and so far I’ve spotted 2 resistors that look like they’ve heated up. On the right.
No schematics ,so only have visuals at the moment.
Any help would be good.
I’ve got it’s bigger brother (IPS3700) aswell with the same fault but leaving that one for now.
Cheers 🍻

Single-Supply Preamps For Guitar

Here are six preamps for guitar using op-amps; each operated from a single 9V battery as the power supply, thus making them suitable for stomp-box applications. Sources for the diagrams are noted with the diagrams. The circuits are faithful to original diagrams but for minor changes, such as the adding of an input level control or the use of a pot in a negative feedback loop to provide the user with a gain control. However, while based on proven designs, these designs must still be considered experimental. Also, the diagrams originally posted had flaws, which were corrected according to advice from forum members here, who I thank for their help. The diagrams shown now are updated versions (but some may be updated again if needed).



The specified IC is the OPA2156 dual low-noise op-amp IC, but an original TI unit can be expensive and may take a long time for delivery. So, I have listed a number of pin-compatible substitutes at the end of this post, all of which will provide good results.

The next circuit features a Baxandall type of 2-band tone-control, followed by one with a Fender-style 2-band control, which works differently from the Baxandall-type, though both leave the mids fixed. For those who prefer three bands, all other circuits below involve 3-band Fender-style tone stacks.



Note the seemingly redundant Tone control at the output below. That is included to provide an additional means of shaping high-end response.



What follows next are the preamps with the 3-band tone-stacks having default Fender Bassman part-values.



Consider the difference in the way the input and output jacks are shown below. This would be a good way to do the foregoing circuits too, but I show this configuration only for the last two circuits. This one also has a fuzz circuit using diodes, providing clipping without having to overdrive op-amps. But if that is desired, an older dual JFET-input IC can be considered, having minimum supply voltage <= 4.5 V, such as the LF353.



The circuit below likewise has a diode-type fuzz circuit, but which is highly adjustable.




Here is the pinout diagram for a dual op-amp, along with pin-compatible example ICs.



ICs listed here should perform satisfactorily for the electric guitar and offer the least noise and distortion for circuits operated using a single 9V battery. They were chosen from among the best op-amps available today but whose minimum supply voltage is <= 4.5 VDC, since half of 9V is 4.5V, and a single-supply circuit sets the zero signal-reference operating-point midway between ground and the supply.

To get more distortion than the diodes provide, by overdriving a given circuit, use the LF353 instead of any of the ICs listed above. It is an older op-amp that will work with a single 9V battery as the supply and is a JFET-input op-amp, which means its overdriven tonality is preferred over that of most non-JFET-input op-amps. However, it will also work fine without being overdriven, and will not drain the battery nearly as fast as those listed above.

Note On Fuzz Circuits: The diode-based fuzz circuit used in the last two preamps are based on article “Build The Mod Box” by Thomas Henry and Jack Orman, in the May 1997 Electronics Now magazine (p.45). Obtain the article free online with this link.
https://www.muzique.com/news/images2/Mod_Box_May_1997.pdf
It originally specified the TL074 quad op-amp since it had four sub-circuits, and was operated from a split-supply. And despite being somewhat dated (25 years old), the circuitry has remained valid to this day, though I have adapted the diode arrays to conform to a single-supply circuit.

Additional Notes: Most circuits here are based on designs in the book “Building Electric Guitars” by Martin Koch. [$33.56 on eBay.] But the input stage of the second preamp is instead based on a design by Craig Anderton appearing in the Nov. 1981 issue of Guitar Player magazine. [Original issues start at $10.00 on eBay.] Those exact circuits cannot be found online, so a link to them is not available. Yet, I have depicted them accurately but for the minor additions I mentioned at the beginning of this post. Furthermore, while I receive much help diagnosing the diagrams I posted at first, I supplemented that help with online research, and an electronics textbook used at an accredited university. Consequently, all diagrams above have been updated and should be viable, but I would still treat them as experimental until after breadboard testing efforts have been completed.

Final Note: For comparison, the diagrams originally posted have been retained in the Attachments folder at the bottom of this post, while the updated versions are also there. I am embarrassed by the original ones, but proud of the new ones, though members who helped me with them deserve lots of credit. One thing should be mentioned, however, regarding the 0.2uF capacitors going to ground on the second stages of most of the diagrams. That, along with its associated resistor, sets the low-frequency cutoff for the associated op-amp at 80 Hz for 6-string guitar. But for bass-guitar, and 7-string or 8-string guitar, make it 0.43uF, for a 40 Hz cutoff. And for no low cutoff at all, it can be omitted. And by the way, the following replies correspond to the original diagrams, not the new ones. So, if you want to know what they are talking about, refer to the old diagrams as left in the Attachments.

EOF

Thanks for having me.
Recently heard some Tube amps and they blew my class D amps out of the water.

Nice to meet you!

I finally registered here as I find here lots of good information from which I learned a lot and want to learn more and help too if I know about someone's doubt.

Need to replace the left front speaker terminal. The plastic broke and now I only have the rear channel.

Hi all. I run this little Magnavox amp on my desk. Just recently, I started hearing different frequency sounds coming out of my speakers. The frequency is higher pitched at lower volumes and goes lower as I turn up the volume. Any ideas what going on?

I started this thread for IRFP240 Quasi: https://www.diyaudio.com/community/threads/quasi-irfp240-amplifier.428495/
Now I use another MOSFET transistor for better qualities.
I use Exicon Lateral ECX10N20 for this thread.

The benefits are three:
1. Somewhat less distortion, THD 0.005%
2. Higher max output, 25 Watt in 8 Ohm
3. No need for MOSFET protection diodes, are already built in

Here is the schematic. Enjoy 🙂

If I clog up the high side of my sixth order parallel band pass i I still get those quarter wave resonances ?

Are they based on the 120 cm resonator (it’s a big vented reflex) ?

(Video by @maxolini )

Some amplifiers, like the P3A by Rod Elliott for example, use a PNP transistor for the VAS, others use NPN, like the blameless amp by mr Self.
Does it matter? looking forward to your well founded opinions.

Hi everyone!

It's been almost two years since I last built a speaker, and I'm getting the urge again.

[TLDR:
A more compact floorstander using an SB Acoustics SB15SFCR39 for bass in a vented enclosure, extending further down to the early 20hz with boundary/room loading. The B&C 5CXN44 in a sealed enclosure for mid and high.]

My current setup is really good (here), but a few aspects have me considering a change:

Size: The speakers are quite large.
Low-end Limitation: They only reach down to 35hz with boundary loading.
Horizontal Coverage: It's not as wide as I'd like.
Practicality: The horn on top is a bit of a concern, especially with baby GG² about to start walking.



As you can see below the measured 'anechoic' response yields quite a bit more low-end extension when in the listening position. I thought this would be fine but there has been the odd movie or song that lacks a little in the rumble department. Boundary loading really is my friend here (more on that later).




Shortly after building these I added Dirac and that improved things a lot over my manual room correction (manual not shown).



When I made my workshop speakers (here) I really enjoyed the sound of the B&C 5CXN44 coaxial drivers and found that they gave quite a nice wide image even when I am at the end of the sofa. I think this is because of the diffraction and tuning the response to work best slightly off axis due to the choppy response a 0 degrees.



SO...

An idea I have been chucking around in my head for a couple of weeks is this:

A more compact floorstander using an SB Acoustics SB15SFCR39 for bass in a vented enclosure, extending further down to the early 20hz with boundary/room loading. For mid and high, the B&C 5CXN44 in a sealed enclosure. Both drivers would share the floorstanding unit, reducing both physical and perceived space.

SB15SFCR39 ≈ 26L / 35hz vented
5CXN44 ≈ 4L sealed

I am totally up for other drivers but I don't want to go beyond 200mm wide baffle and I'm not too keen on a side mounted woofer owing to little fingers.

As discussed here I have implemented a good approximation to a 4th order LR to a Seas tweeter by estimating the existing Q and frequency, and using an LT to raise the cut frequency and make the response align to the 2nd order Butterworth - and then overlaying a further second order.

This gives a pleasing result with my miniDSP crossover.

I'd like to use this technique again - but with a conventional (if external) crossover. Is there any software that would do this, or should I just try to replicate the response by simply fiddling in XSim with trial and error? I don't need to introduce any gain, its all attenuation and raising the cut frequency.

(And yes that leaves me with the Z offset to mess up phase, which I deal with in pure delay in DSP - but it will be a starting point, and for some tweeters with very low Fs, I might still look for around 1.2kHz 4th order - like my Wavecors for example - which should reduce the phase error somewhat).

There are a lot of recommendations to keep the distance between speaker drive units as close to each other as possible, especially the drivers for higher frequency, because a point source configuration is preferred.

Let’s look at these “Avalon” speakers. They are two ways system comprising a woofer and a tweeter per cabinet. But, as can be seen, those two drivers are positioned so far from each other, and work together (crossover point) at high frequency as a typical two-way system. So, does the distance between drivers really matter?

Or, in fact, the statement above is true, but Avalon has applied the special techniques on them. Is there anyone used to reversed engineering these speakers? Could you please advise us what are the magic/secret designs—the special approaches/techniques, e.g., using higher-order crossovers with large overlap, or using lower-order with a lot of EQ circuits including baffle step correction, etc.?

Hi All. I have the 5500 Mk2 tuner that works perfectly and sounds good, but every "locked" station shows a frequency of +0.05MHz. QUESTION, is this just a case of the frequency counter of the display needing a fine adjustment or does more complex adjustment need to be done? I realise it would benefit from a total accurate realignment, but cannot find anyone in the UK that can do it. If it is just the frequency counter how do I adjust this please? .....Thanks for reading.

Hi All. I just cannot get my head around this BIAS measuring of the L12 Amp to enable to set the bias trimmer! QUESTION: What VOLTAGE should I set across the combined emitter resistors please in the shown example below to achieve a bias current of 40mA? It's a BASIC OHMS Law calculation, but do I use the SUM of the resistors (0.44 Ohms) or just the single value of 0.22 Ohms?. ... In my amp the emitter resistors are in fact just 0.1 Ohms.... Perhaps I should measure each collector voltage separately to ground across it's single resistor instead? Many thanks for your help.

Good morning and happy Monday, y'all! I hope everyone had an excellent Thanksgiving this year. 🙂

Before I get into it, I want to thank Victor Kung @ VKAudio for his assistance and patience in answering my questions, particularly about Capacitor upgrades. His knowledge is invaluable and he's a good guy to have in your corner when you need help. I also want to thank TubeDepot which is where I purchased my kits from. Honestly, if I'd have spotted Victor's store earlier, I probably would have purchased directly through him. However, I have no complaints whatsoever with my transaction as TubeDepot has always provided me with great products at fair prices.

With that done, I'll move on.

I've been an audiophile for more than 35 years. Music is the one thing I can depend on to get me through each day. It make a bad day turn good and make a good day even better.

Now, I'm not an audiophile in the "I own a set of Infinity IRS-V speakers powered by 3 Macintosh Amplifiers" sort of sense. I would love to be there, but that's a financial commitment I'm simply just not able to justify. However, I absolutely love how a quality sound system can bring out deep emotions, stir up memories of people and places, and transport you to another time. It has that much influence.

Until recently, I've always been a solid state kind of guy. Part of the reason is that the generation I grew up in was on the leading edge of Transistor technology with vacuum tube well on their way out the door. In fact, about the only thing you could buy with vacuum tubes in the '80s were guitar amps, and even those were rapidly declining in favor of the less expensive, sturdier solid state equipment. I do remember as a younger child, my parents and grandparent having old vacuum tube equipment, and I would spend many hours listing to records on this equipement. This probably set my mental bar for audio quality.

I guess that the other reason I was into solid state was that I was very big into car audio in the '80s and '90s, and there just isn't any place for vacuum tubes in car audio! I see there are companies that actually specialize in this kind of equipment, but that all seems a bit gimmicky. LOL

A couple years ago, I really started getting back into the idea of analog sound. I have always love vinyl and actually had an impressive collection until our house burned down in 2015 and we lost everything. Almost 40 years of collecting lost forever. Anyway, while I enjoy the analog nature of vinyl, it's always been through solid state amplification. Not that it was bad... In fact it's quite the opposite. It's just that I always did feel like something was lacking. Some level of depth or warmth that I remembered from my childhood that I just couldn't seem to reproduce. So decided to start looking around at what's available.

Vacuum tube audio equipment is clearly big business these days, with companies building preamps, amps, active crossovers, and other stuff that vary in range from hundreds to hundreds of thousands of dollars. The array is as dizzying as it is almost endless. Reviews leave you with more questions than answers, and almost anything with decent reviews will empty your wallet in short order. But the question at the end of the day is, "Is a $10,000 amplifier any better than a $1,000 one?" I'm sure that at the end of the day the answer would be, "Yes, the $10,000 amp is better." But the next question would logically be, "Is it $9,000 better?" In many cases the answer would be solidly, "No."

With this in mind, I started looking at many different amplifiers in the "under $5,000" range. There are a number of what I presume to be quality products on the market in this price range, but still couldn't make heads or tails of what was actually worth dropping my hard-earned cash on. Oddly enough, while researching different amplifiers, I came across a DarkVoice 336SE headphone amplifier in a pawn shop and scooped it up. Even before I plugged it in and turned it on, I was already looking at reviews, schematics, and upgrades to this unit. My first 20 hours into it, I had already order a variety of tubes so that I could start rolling as time permitted. I do realize that this little amp met with mixed reviews, but was mostly positive. I also discovered that there were some very easy improvements that could improve the quality and reduce the idle noise of the amp. I, personally, love this little amp. It brought back that sound I so vividly remember from my youth. The depth and presence that I felt I was missing was restored. Even with high quality digital recordings, there was a sonic improvement. That's was it. I'm now hooked on tube amplifiers!

I figured the best place to start would be at the bottom of the chain and slowly work my way up. After digging around for several months, I came across the TU-8200R Amp kit along with the TU-8500 Pre-Amp kit. I found it intriguing as it was a full-on, build-it-yourself sort of kit that came with everything. The reviews on it were outstanding, and the documentation on component upgrades was well defined and easy to follow. The only pitfall was that some of the upgrade components were no longer available, such as the AMTRANS components, but c'est la vie. The biggest improvements seemed to come from using Mundorf coupling caps and Lundahl transformers anyway, so that was what I opted with for the final build. Tube rolling will come later. Ordered through TubeDepot and then waited for the brown Santa to show up in his trusty delivery van.

I did my process a little backwards, starting with the pre-amp first as I was waiting for my Lundahl transformers to come from VKAudio. Out of the box, this thing is a solid work of engineering. The chassis was remarkably clean in design and fit together well. I may, at some point, disassemble and have the chassis powder coated a different color, but it's great as it is. The PCB is laid out very well, all the components were grouped in nice, organized baggies, and the instructions were very clear to understand. Now, I'm an electronics engineer by trade (digital, not analog), so following a schematic and PCB layout is a simple thing for me. Not to mention that I have a helluva soldering station. PCB assembly and soldering didn't take long, and within a couple hours I had the whole thing put together. Slapped in the J/J tubes, connected my Technics 1200 turntable to it, and hooked it up to an old Pioneer SX-980 with Kenwood KL-777A speakers and then let 'er rip.

I let the pre-amp run for about 2 hours just to let things settle in and then dropped Tom Petty on the platter. Even with the tubes barely even run, the difference was immediate. The sound was warmer with vocals that were more in the foreground. I played this album and then moved on to my next favorite, Pink Floyd - Wish You Were Here. Shine On You Crazy Diamond (Parts 1-5) is my quintessential sound system test. For me, this track reveals the soundstage and depth of a sound system, or even the capability of individual components, and let me tell you, the TU-8500 did not disappoint. I went with this setup for several days, and with each passing hour the pre-amp continued to provide subtle improvements in sonic and tonal quality. I now realized I needed to get the amplifier built!

Thanksgiving intervened and I had to do all that family stuff. While I always enjoy being with family, I was quietly preoccupied with wanting to put the amp together. Our office was closed for Black Friday so I headed in and began the process of assembling the amplifier in peace and quiet. Again, all the details were impeccable and assembly was completed in a matter of hours. I'm a bit of a perfectionist, and since the tubes are exposed in this unit, I spent a little extra time making sure that the tube sockets were level and even. Unfortunately, during my final assembly of all the PCB units, I forgot to solder the bridges on the bottom of the input/output board which left me a little perplexed when I fired up the amp and had no sound. A few muttered curse words and 45 minutes later, all was good.

I've always heard from people that 10 watts of power from a tube amp is like 50 watts of power from a solid state amp. I always took these comments with a large degree of skepticism since I've always argued that "Watts is watts." I guess that it's true that wattage is a pretty stable measure of audio output. But let me tell you right now... This little amplifier that only runs 8 watts per channel sounds better than my 100 watt per channel Onkyo receiver (in stereo mode) running through a set of B&W 603 S2 speakers! Of course, the Onkyo running in 7.1 will simply outmatch almost any 2 channel amplifier for range and soundstage, but in 2 channel mode I would take the little tube amp every time. There is no absolutely no hint of hiss or hum at idle with no input, even with headphones on. It is dead quiet until you feed the signal back into it and then it just fills the room.

I just couldn't believe the quality of the sound the TU-8200R produces at a pretty solid listening level. It's certainly loud enough that I can feel immersed in the music, and that even with the TU-8500 pre-amp on a flat 1x gain on the inputs. I haven't even turned it up to 3x yet, and from what I've read so far, the amplifier can certainly handle the higher input levels without becoming too distorted. I may give that a try tonight and see how well it handles. I expect it may pick up some signal noise, but I can't say until I give it a spin.

As an aside, the amplifier works great with a set of high quality, high impedence headphones. My Sennheiser HD-650, which works fantastic with my DarkVoice 336SE, sounds fantastic with the TU-8200R. Of course, you don't buy an amp like this just to plug in a set of headphones. It's just nice to know I'm not losing anything in sonic quality. My DarkVoice picks up stray interference with the volume turned up and no live signal coming in, but I think that has more to do with being pretty close to my monitors on my computer desk. I've moved it to another location and that noise disappeared. However, I do not get any noise with this combination of pre-amp and amp sitting in the same location, so the DarkVoice might need a little more input filtering. I did install the PS-3249 DAC module and that works just as I would expect. I don't expect to use it much, but it's there if I need it and it doesn't take away from anything when it's not plugged in.

Overall, this has been an absolutely fantastic kit build. It's a pretty easy go, provided you have a resonable level of skill with soldering. The rest is just following instructions. I did take the amp apart and installed the original transformers just for a quick listen, and the sound did have a little bit of dryness to it. I can't quite explain it, but it felt like some of the low frequencies were a little muddled and washed out which took away from some of the depth of the soundstage. I can't say for highs since I have tinnitus (too many years of concerts and loud car audio) and can't hear anything above about 12,000 Hz, but the range I can hear still seemed to be pretty clear and sharp.

Tube equipment is clearly the way I need to go in the future. I got my feet wet and I'm now ready to jump in. Just for grins, I've already ordered and received the TU-8550 pre-amp and will soon order the TU-8900 amplifier. I'm ejoying the building of these kits and have been absolutely floored by the sound quality of both the pre-amp and amplifier. I mean, for the price you just can beat it. And with the ability to swap power tubes in the TU-8200 between the 6L6, KT88, EL34, 6550, and possibly even the KT90 (I don't know if KT120 would work), all with auto biasing, it makes tube rolling a fun (albeit potentially expensive) adventure. But if you're going to tube roll, then this is a great way to start out.

Again, many thanks to Victor Kung, TubeDepot, and also to Elekit and the infamous Mr. Fujita. Excellent work by everyone in bringing this exceptional product to market and making it affordable to the average hobbyist. It provides entry level access to quality tube audio equipment as well as a platform in which to build a solid foundation of understanding how tube audio equipment works. Well done!!!

TL;DR

I bought the Elekit TU-8200R amp and TU-8500 pre-amp kits along with Lundahl transformers and Mundorf caps for the amp. I assembled them and they are *******' AWESOME! Do yourself a favor and get these for yourselves. You can't beat the quality, especially for the price.

Hi gents, I am a new to the group.

Hello

Currently I am trying to design and build prototypes similar to danley BC218, divided in half for easier transport and multiple possible deployment configurations. I will be using 18SW115 8ohm version as i already own them and it seems Danley also uses them a lot in his designs. My goal is not to make a identical copy of his design but to make a sub with similar preformance and same design benefits. Aiming for same dimensions and about same frequency response (-3db@ 28hz).

I've been following @weltersys folding prediction and made a 3d model, but 18SW115 sadly doesnt fit and i am not exactly sure how to model in hornersp with decent accuracy.

I am relatively skilled in Solidworks, Catia and cabinet building (designed and built my own version of TH118 and 4 way synergy horn), but lack experience for simulating such advanced horn sub.
Any help would be appreciated, as i said looking forward do design 3d model first from usable simulations, then build 2 or 4 prototypes - I have a huge free area near my house and I will keep this thread updated with build progress and measurements.

Looking for support in simulating this folding or exploring possible different horn foldings

Attaching picture of a very rough model

I own 2 atmos avrs and find both disappointing. IMO, height sounds such as aircraft, birds, thunder, rain, wind, spacecraft, rocket and other height sounds should only come through height speakers. Atmos is overhyped and basically snake-oil. There's no unique content in the heights. It's just a replication of the bed layer channels. Manufacturers are no longer making bluray players, because physical media is dead. Everyone is streaming. But streaming has a bandwidth problem. You can have quality 5.1, or compressed & lossy atmos. Since the heights are just the same data repeated, it makes sense to save the bandwidth and do the replication in the avrs instead.

I also want to boost the bass only for gunshots, thunder and explosions, and not music. And have automatic attenuation.

These are my goals...

Hello all! Came upon a Crown DC300A. Bringing it back to life.

I was thinking about replacing the motor in my TD166 MkII with a brushless DC motor. The main aim is electronic speed adjustment to replace the mechanical belt-shifter.

Has anyone tried this? I did find the Origin Live and would like to built my own "cheaper" version of that.

Was thinking of using PWM speed control, possibly with hall sensor. Or an Arduino-based controller.

Can use soft-start or keep the original clutch-pulley

Is this a good idea? I would appreciate any advice on the best motor/pulley to use and reliable speed controllers.

Alternatively; let me know if it is a stupid idea and that I should just leave it alone 🙂

Good afternoon. After ordering a qcc5125 + es9038q2m + NE5532 board on aliexpress, a problem arose. In general, the board gives me good enough sound for me, but there is a nuance. After I stop the sound from Spotify, there is silence. There is no sound in games, no sound in Discord either. I saw that this is a problem with the firmware of the Bluetooth module itself. Can I reflash it? I'm not a handyman, but I don't have very deep knowledge of electronics. Is there any way to fix it? I'd be very grateful if someone could throw a link to the instructions or write how to fix it.


Module: link