So I connected the transformer board to the preamp and plugged it in and ... nothing., The LED on the board didn't go on. There is a pair of "K" terminals on the transformer board so I suspect it has something to do with these. Shouldn't the transformers however work without these connected? Not sure ... please advise. Thanks.
I am getting too old to be sticking my hand into 500-volt electronics...lol I want to sell off all the stuff I have acquired over the years.
I have a working Navy tube tester that I have used for the last 10 without issue.
The TV-10 is not a very well known tube tester, they were made for the Navy to replace the TV-3's.
There just weren't very many of them made. It is basically an upscale version of the ever popular TV-7.
Makes the TV-7's look like a toy.
Like most other US military tube testers, the TV-10 is a mutual transconductance tester using the Hickok circuit.
Uses the same Data as the TV-3's
The TV-10 reads the mutual transconductance directly on the meter in micromhos while the TV-7 only reads a relative number on it's meter.
If you are looking for one of these, you likely don't need me to tell you what it is or does.
I will consider all reasonable offers. I will also post some photos of new JJ Tubes for sale so that you can make offers. They came from EuroTubes, who supplied me with tubes to build tube guitar amps.
Hello,
I am looking forward to build a headphone amplifier and looking for a project/schematic I could use as a reference. I am looking for the following:
capable of driving a pair of Beyerdynamic DT880 250 Ohm which I am using at the moment (if possible also the 600Ohm version), probably will upgrade later
Class A (the single ended class a topology or in genereal the designs by Nelson Pass really appeal to me 🙂 )
external PSU prefered
parts which are available at the known large distributors, if possible also in a few years
maximum size around 200x200x50mm
For me its okay to do the PCB layout by myself/use SMD components to shrink the needed size a bit (and accept maybe a bit less performance)
IMPORTANT NOTE: A Zener diode from the MOSFET Gate to Source is needed to protect the MOSFET from inductor "kickback" I got away without this when using MOSFETs with high maximum VGS and lower supply voltages, but had not anticipate the range of FETs, inductors, and supply voltage that might be used. Even when the stars align, it's too close to not consider the Zener mandatory.
There's already a place for this diode on the PCB sold here in the store. When my stars align, I'll update the article pdf.
Inspired by the Rutcho Dm601 S3 mod, I decided to tackle the crossover on my pair of DM603 S3's using his design. It is my first time diving into this and I will admit I have no idea what I am doing. I imported the response graphs of the Kevlar woofer and tweeter into VituixCAD, and simulated the stock and Rutcho crossovers. I came up with a 1.5ohm R1, a 3.3uF C1, short of the resistor on the tweeter inductor. For the mid-woofer, I created a zobel network just like Rutcho, at 4.7uf cap, but a different resistor value of 4.7ohm.
Using a 2.5 Allen key, I removed the silver woofer. There are three screws - one at the middle and one at each bottom corner. The top corners are held by expansion pins. I used a pair of pliers to squeeze them and put my hand behind the board to apply pressure inward. For the zobel, I cut open and balded the blue and brown driver wires and mounted the capacitor and the resistor from the leads vertically. The mounted components are held by glue, and I had to use a knife to pry them off. I did this to both speakers but my beginner soldering skills had my zobel bridge compromised and open. This gave me a unique opportunity to A/B the before and after while running simultaneous modified tweeters.
The highly crossed Kevlar woofer was essentially distorting at the high frequencies. It sounded like outright white noise and heavily muddied the mid-range.
VituixCAD shows on the stock crossover a terrible phase shift after 7.5Khz and both the directivity graphs and phase patterns looked ugly. My post-mod speakers sound huge, delicate, spacious. it is definitely an improvement all around. However, I lost a lot of mid range and I suspect it is due to the 4.7ohm resistor on the zobel. I could play with 1.8ohms but the second problem is that the phase is nearly completely opposed from 2-4.7 kHz. The vocals sound delicate but they are also distant and lacking bark. I suspect I am not getting the full sound stage yet out of these drivers.
After some research, it occurred to me that the zobel increases the order of the filter. For parity, do I have to do the same to the tweeter? I ran this through VItuixCAD and came up with a 4.3uf C1 (I add a 1uf cap in parallel to the existing one), a 4th order filter created by a 5.6uf cap and a 200uh 15AWG inductor, and reducing the Zobel resistor to 3.3ohms.
TL;DR - do I have adequate info to model a box with the limited info I have?
The story...
I recently, maybe foolishly, got a couple polk DR1242SVC drivers from P E in Ohio. I was hoping to use these shallow (110 mm) drivers in a box indoors and out... and have put some effort into extracting info from Polk. I got... some... see the pic attached. When I asked for explanation or additional data, I got the explanation below. But there is no info about the voice coil or the magnetic characteristics. Am I missing something? It seems without Bl and Xmax in particular I can't really model this driver. I could measure/estimate Xmax... and maybe even the voice coil size, etc. Looking at the definitions... maybe I could derive BL? Or just return these and get something better characterized Any help or suggestions would be appreciated !! (I also attach the WinISD file I made... I include a guess that Xmax is ~10mm)
I should probably add that the manufacturer does give recommended box sizes for sealed and "vented" saying sealed should be btw 1.25 and 3.5 ft3; vented 1.75-2.25... but no vent info for tuning
"MLSSA Sd / Added Mass
• Sd (cm²): Effective surface area of the cone. Larger Sd generally means more air movement = more bass output.
• Added Mass (g): The mass added during testing (typically for the added-mass method of T/S parameter measurement).
- Added mass refers to a known weight temporarily placed on the speaker cone during Thiele/Small parameter measurement, specifically to calculate mechanical parameters like Vas, Cms, and Mms using the Added Mass Method (also called the mass-loading method). Added mass is NOT a permanent part of the driver. It's a temporary tool used during testing to help extract mechanical properties. The accuracy of Vas and Mms depends greatly on the precision of this added mass and the measurement setup.
Fs (Hz) – Resonant Frequency
• Frequency at which the driver naturally resonates.
• Lower Fs means deeper bass response.
8" = 45.56 Hz (less deep bass)
12" = 31.57 Hz (better for sub-bass)
Re (Ω) – DC Voice Coil Resistance
• Resistance of the voice coil, relevant for amplifier matching.
- All are around 3.3–3.5Ω (likely designed for 4-ohm systems)
Res (Ω) – Mechanical Resistance at Resonance
• Resistance at Fs, reflecting both electrical and mechanical damping.
Qes – Electrical Q
• Electrical damping from the voice coil and amplifier.
• Lower = tighter control, but might reduce efficiency.
Qts – Total Q
• The total damping factor of a speaker, combining electrical (Qes) and mechanical (Qms) losses.
• Qts = 0.51 falls in the mid-range:
• Not too tight (overdamped), not too loose (underdamped)
• This value suggests flexibility in enclosure types:
Suitable for sealed enclosures (tight, controlled bass)
Also works in ported enclosures, though some optimization may be needed
L1 & L2 (mH) – Voice Coil Inductance
Inductance values at different frequencies.
Affects high-frequency roll-off.
R1 (Ω) – Voice Coil Loss Resistance
- Resistance due to eddy currents and other losses.
Vas (L) – Equivalent Compliance Volume
• Imaginary volume of air that has the same compliance as the speaker’s suspension.
• Bigger Vas = softer suspension (needs larger box)
12" = 43.13 L (requires large box)
8" = 8.65 L (compact box)
Mms (g) – Moving Mass
• Total mass of moving parts: cone, coil, air load, etc.
• Affects sensitivity and response speed
- Higher Mms → lower efficiency but deeper bass"
I'm having a clear out and have a lovely, historical Wiercliffe bulk reel-to-reel tape eraser to find a new home for. This is free to a good home, but must be collected from York in the UK. It's far too heavy to ship. It would be great to preserve this interesting piece of audio history, rather than it going to scrap.
The eraser is a large, nicely made wooden box containing a heavy AC electromagnet. It's complete with original instructions and is in really excellent condition for its age. However, I'm passing it on as a decorative item only. I do not encourage or condone anybody plugging this thing into the mains. The plug socket has been removed to reinforce this point!
Happy to answer any questions about it. Hopefully we can find it a new home.
Hello. My name is Phil, I own a pair of Antique Sound Labs AQ-1006 DT (845) Monoblock amps. These units are about 25 years old and have served me quite well during that time. I like them so much that about 2 years ago I had them refurbished and swapped out the stock Power Capacitors and just general clean-up and maintenance to prolong the life of the amps for as long as possible. Recently one of the amps failed and no longer puts out any sound to the speakers. It powers up just fine, and from what I can see the digital bias meter and tubes all light up. In the course the warm-up I hear a very loud "POP" through the speaker and then a very loud and noticeable "HUM" while the system is in standby. Not the normal hum you will get from tube gear but a very loud hum that can be heard from more than 10 feet away. I surmise that the pop and hum were the preamble to the amp now not playing. Based on what little I know about the amps my guess is that I have lost a Power Supply, but I could be wrong. Whatever it is I believe (or rather hope) it is a straightforward fix to the problem, but was hoping to find someone that would be able to properly diagnose and fix the problem for me. Is there anyone in the community who is able/willing to work on it for me?. Or direct me to someone who can. Any suggestions or information is greatly appreciated. Thanks in advance. BTW, I currently reside in Central Texas but I am also willing and able to ship the Amps outside of my home state for repair services if necessary.
Pico's Dumb (ie Pico is Dumb) Biasing Trick for F6 and other Amps
As you may or may not appreciate many devices have a positive, negative, or virtually zero temperature coefficient.
Most mosfets have a positive temperature coefficient, ie as they heat up they draw more current when biased with a constant voltage source applied to the gate of the mosfet (provided the bloody voltage is adequate to turn it on).
To reduce the effect of this increasing current draw as the device gets hotter we can do a few things.
1) Reduce Delta T of the devices, eg larger heatsinks, better thermal pads ie improving Case to Heatsink performance, etc
2) Using source degeneration which acts as a form of negative feedback, ie as the device attempts to draw more current, an increasing voltage drop occurs across the source resistor which effectively reduces Vgs which then stops or reduces the amount of thermal current drift.
3) Using NTC thermistors in the bias circuitry, which effectively reduces the Voltage at the gate as the thermistors heat up, eg as seen in F5, and elsewhere. I used a similar method to this, in F4 beast builders ie building push-pull amps (mostly with hockey pucks) with zero degeneration.
4) Active bias control circuitry, optocouplers, hall sensors, discrete designs.
You can obviously also combine some or even all of these methods, to get the result you need.
There are probably quite a few more techniques than this, but I am telling the story and I don't want it to be batshit boring.
Even though I have successfully used method 3, and 4 in the past, I don't like "intelligent solutions", I like dumb simple solutions that don't require maths or too much intelligence.
I have often thought, there has to be a dumber way to do this.
So I am sitting on the toilet (it always happens like this), and I am considering my options with regards to voltage references, TL431, LM329, Zeners, Leds.
I start thinking about Zeners and leds, since they are dumber than the other devices, and remembering that Zeners below 5V have a negative temperature coefficient and zeners above 5V have a positive temperature coefficient, the light bulb turns on "aahhh you bloody dumb bastard" and gave myself an uppercut for not considering this earlier.
Many here have either personally built the F6, or are at least aware that in many cases, the 5.1V zener shown in the original circuit diagram is not quite adequate to produce the required voltage at the gate to bias the F6 to the required value.
Most of us have used 5.6V, 6.2V, 6.8V zeners etc, some have used LM329.
Well, all these devices have a positive temperature coefficient, you could probably say that LM329 is effectively zero but it has a very small positive temp coefficient.
So, what is stopping us from using 2 smaller valued Zeners to achieve the required zener voltage eg 2 x 2.7V zeners, and achieve a slight negative temperature coefficient.
LEDs also have a negative temperature coefficient with regards to Vf, but we will first consider comparing Zener diodes.
Anyway, so I decided to compare the observed measured differences in biasing up a Vishay IRFP150 mosfet (with zero degeneration) using 3 different zener configurations:
1) 2 x 2.7V zeners in series (effectively 5.4V)
2) 5.6V zener
3) 6.2V Zener
4) 3 x Green LED LTL 4231N - tested at a later date
The zener diodes were fed around 5mA in each case (more detailed information below)
A multiturn trimpot was used in each case (just like F6 circuit) to achieve exactly 3.925V at the gate.
I used zero degeneration on the mosfet to better illustrate the effect.
Case (1) 2 x 2.7V Zener Configuration
Initial Vgs: 3.925 V
Ambient Temperature: 21.8 deg C
Id at turn on: 1.32 A
Id at thermal equilibrium: 1.68 A
Vgs at thermal equilibrium: 3.899 V
Delta Id after thermal equilibrium: 1.68-1.32 = 0.36 A
Delta Vgs due to Zener: 3.899-3.925= -0.026V (negative 26mV)
------------------------------------------------------------------------
Case (2) 5.6V Zener
Initial Vgs: 3.925 V
Ambient Temperature: 21.8 deg C
Id at turn on: 1.32 A
Id at thermal equilibrium: 1.95 A
Vgs at thermal equilibrium: 3.936 V
Delta Id after thermal equilibrium: 1.95-1.32 = 0.63 A
Delta Vgs due to Zener: 3.936-3.925= 0.011 V (positive 11mV)
------------------------------------------------------------------------
Case (3) 6.2V Zener
Initial Vgs: 3.925 V
Ambient Temperature: 21.8 deg C
Id at turn on: 1.32 A
Id at thermal equilibrium: 2.25 A
Vgs at thermal equilibrium: 3.954 V
Delta Id after thermal equilibrium: 2.25-1.32 = 0.93 A
Delta Vgs due to Zener: 3.954-3.925= 0.029 V (positive 29mV)
------------------------------------------------------------------------
Brief Discussion of Results
As predicted during my toilet brainstorming, the 2 x 2.7V zener in series produces an excellent result, with a delta Id between turn on and thermal equilibrium of 0.36 Amps. This was achieved using the negative temperature coefficient of a 2.7V zener to our advantage.
The 5.6V zener is twice as bad, and the 6.2V, almost 3 times as bad as our 2.7V zener setup.
The test was performed on a large heatsink (relative to the heat dissipation of the device) flat on the floor, with excellent ventilation.
If the same test was repeated on a relatively smaller heatsink with regards to dissipation like one you might be using, and performed inside the chassis of an amp, we would expect to see an even greater positive affect with regards to using 2 x 2.7V zeners.
However, if I were to repeat this test exactly as performed above using degenerating resistors the results would be closer together.
This was merely done the way it was to clearly illustrate the effect, and I always try to avoid any kind of degeneration, so this kind of thing is of interest to me.
You could also try 3 x 1.8V Zeners for an ever greater effect.
Or you could use 2 x 3V or whatever you might have. Basically anything below a 4V Zener configured for the voltage you need, is going to have a nice affect. EDIT: A 3 x LTL 4231N LED combination was also tested and gave favourable results like 2 x 2.7V zeners, however it has better regulation than the Zener combination.
It's up to you if you want to try this out. I am not into preaching, just sharing.
Test Conditions:
Zeners tested:
5.4V (2.7V x 2 in series) BZX55C2V7-TR
5.6V BZX79-B5V6.113
6.2V BZX79-B6V2.113
LED: LTL 4231N
Mosfet:
Vishay IRFP150
Trimpot:
10k 0.5W Bourns 3299Y series
Resistor feeding current to Zener
3.3k CMF55 (zeners biased at approximately 5mA - close enough)
Powersupply Feeding Circuit
Linear regulated lab powersupply set to 23V
Howdy. Looks like I made my account in 2015, but this is my first post. Looking to learn, first with mods to what I have, and hopefully full speaker building once I am prepared.
I am mostly networking/hardware guy, but deeply interested of DIY power amplifiers in hope to squeese better possible sound from vintage speakers (in my particular case - "Mitsubishi" from mid 80's). Hope this is enough for first introduction.
Hello! I have often referred to diyaudio for answers to tech questions. Thank you for the help I received. My focus now is a series of cabs to house my collection of raw drivers. btw My first serious cabs I built were a pair of VOTT's for my band. I found out that reproducing recorded sound and supporting live sound was a whole new challenge. My current goal is to mostly use vintage drivers to make high end sound, like the A7's so loud and clear you just have to laugh.
I just recently started building loudspeakers and amps myself, am new to soldering, still learning about resistors and all these kinds of things and am little bit stuck with a problem.
For both networks, the load on the amp is around 8 ohm (which the amp is designed for) and the attenuation is almost exactly the same.
Depending on which headphones I use, the damping factors are 1:4 and 1:19 (the 60 ohm headphones) respectively 1:44 and 1:223 (the 300 ohm headphones).
Now, my understanding is that a very low damping factor like 1:4 should give me a bit more boomier bass which is less precise, because the amp is not able to control the driver that well. Other than that I would not expect too much audible differences.
However for both headphones the setting with the lower damping factor leads to much worse sound quality. The soundstage collapses, it becomes very noticeably less wide and everything sounds a little bit less clear. Somehow much cheaper. It is the same on both channels and it has nothing to do with the volume.
I am using MOX resistors everywhere and from what I can tell everything measures fine (the resistances I measure are consistent everywhere, I don't see anything that seems odd). I bought most of the resistors at a local store and I don't know how old they were. They might have been lying there for a while.
Can you help me figuring out what the issue could be? Might some resistors be too old for example? Could I measure anything else besides the resistances in the circuit? The circuit itself does not seem to be the problem as the setting with the higher damping factor sounds pretty good.
I am very confused as both headphones usually sound fabulous, but really cheap with that one setting…
Thanks a lot!
Michael
PS: Sorry if used wrong terms or weird language. As I said, still new to all this 🙂
I bought a pair of knockdown 0.23 boxes from Parts Express to put my 11 MS's in. I have them in big pencils right now, but I'm not too impressed. The ponderous nature of the pencils are a bit too slow for me.
In a 0.23 box would you go with sealed or ported? Would sealed increase the speed even if if it loses some low end. What other benefits would sealed have in comparison to ported? Madisound has the following posted.
Suggested box alignments
Sealed box of 0.15 to 0.2 cubic feet (4.3 liters, 259 cubic inches) 3dB down at 110Hz
Vented box of 0.3 cubic feet (8.5 liters, 516 cubic inches) 1.5" diameter vent by 5.5" long. 3dB down at 65Hz
Extended Bass Shelf Vented Box (lower bass, but also lower power handling)
0.5 cubic feet (14 liters, 864 cubic inches) 1.5" dieameter vent by 4.5" long. 3dB down at 55Hz
I'm fuming with myself.
During a cabinet tweek which involved removing the drivers, I've dropped one of the screws on the dust cover and it's dented it.
I'm sure this has been discussed before, at length I imagine, but these drivers are new territory for me. I don't fully understand what they're made of and how heat etc will affect them.
So far, I've superglued an o ring to a plastic straw and used this to try and suck it out. Nope.
I could use more power by attaching the hoover hose to the straw?
I've tried a piece of Blu Tak and a swift yank. Nope. 🙁
Any other ideas.
I just bought a simple Victor VC2002 generator. It works but the Voltage display seems to be fully out of range. On the front panel it says Vp-p but actually it measures something half of p-p , so a Vmax. But even this value is about 20% wrong. I found an older post with a link to the schematics here (downloadable from Scribd). Unfortunately, the quality is so bad, that I can not identify anything. Does anyone have a better schematics? Or can tell me, which potentiometer is responsible for calibration of the voltage display value?
I am selling an ACP+ and Amp Camp mini combo that is already packed up and ready to ship. I built the combination stacked on standoffs with two ACP+ grounding planes and two Mark Johnson SMPS filters.
The Amp and the SMPS filters are only being held on by very strong plastic surface mount standoffs, so if you cant to take apart the setup, it would be easy to do. I will also include the Green Canare rcas you see in the picture on the back.
I am only selling because I no longer have a work office and need some cash. Asking $400 and I will ship for free anywhere CONUS.
I have been away from audio for quite a while and while getting back into it I see that subject evaluation has almost disappeared, I understand that the DIY forum was never exactly the hub of subjective evaluation but I was hoping we could bring a little of that back so with only friendly judgment and no harsh criticism, would any of you like to share your favorite mods or tweaks that may not be 100% grounded in engineering facts or at least not currently measured by standard engineering?
I will go first, My three biggest tweaks :
1: I have a 100 year house, I wired a direct line to the breaker box , this greatly improved the sound quality, way " blacker " background, inner detail, the stage got more 3D especially in front to back depth.
2. I introduced a silver rock, silver wound TVC into my system as a passive preamp, the was the most significant improvement I have ever had in any system with any piece of gear, It transformed the sound quality into something different, detail level soundstage, low-level detail high frequency overall clarity it was like three veils were lifted I can't say enough about using this piece of equipment it's truly amazing a friend of mine also got one with the same results.
3. Okay I might break people's head here but yes capacitors do make a difference especially in loudspeaker design, I personally found that copper foil and oil capacitors are amazing, I added these to my crossover circuit and again three dimensionality front to back depth inner detail we're all greatly improved what's interesting is I tried many different types of inductors with basically no Sonic difference The only one I thought may have improved things a little, and I mean a little, was a litz wound inductor from Solon, during these tests we were only changing the inductors of the mid-range not the tweeter but again zero difference in inductors at least for me.
GSASysCon Release 3.15.1 is now available for download. It includes an automated setup script that should make it easier to get started. You can find the project at GitHub using this link: https://github.com/charlielaub/GSASysCon
To download the GZIP+TAR compressed file:
Under Linux you can use wget to download the file into the current directory instead of having to use a browser: wget https://github.com/charlielaub/GSASysCon/archive/refs/tags/3.15.1.tar.gz
To unpack the compressed files: first gunzip the gz archive and then run tar -xvf on the resulting tar file. Follow the instructions in the SetupGuide.txt doc in the directory /system_control/docs.
ABOUT GSASysCon:
During the past 10 years I have been using Gstreamer pipelines plus my LADSPA plugins to implement software based IIR DSP processing. I find this approach much easier to configure and use than CamillaDSP so I would like to share this with the rest of the DIY community.
Gstreamer is a powerful multi-media platform that is continually undergoing development and improvement. Because Gstreamer pipelines are difficult to write from scratch, I wrote an application that generates them based on the contents of a user input file that describes the DSP processing. The application consists of a couple of input files and a large bash script (currently about 3000 lines long). The app will also launch and kill the pipelines when directed by you, and basic in-app volume control of local devices is available. Together these can be used to turn your computer into a “preamp” with input switching and volume control plus everything you need to do DSP processing. I use this to run all my own systems.
The app has the name “Gstreamer Streaming Audio System Controller”, or GSASysCon. GSASysCon is completely open source and you are welcome to modify or develop it.
Here is a quick overview of the behavior and features:
• Control of and Interaction with the program is 100% text based via simple input files and a text-based user-interface.
• Substitution and channel duplication capabilities make it easy to describe the DSP processing for complicated, multichannel setups.
• Filtering and routing is easy to configure via an intuitive configuration file structure
• The control interface can turn systems on and off and control playback volume. These features can be controlled remotely over your LAN via SSH.
• Gstreamer provides several source and sink TIME (not rate) based synchronization mechanisms. This makes it possible for synchronized playback of multiple, disparate sinks and adaptive rate playback.
• Can be run under Debian/Ubuntu based OSes (including Rasberry Pi OS) or Windows 11 WSL2 in which the bash shell is available.
• GSASysCon has no external dependencies except Gstreamer, and bash built-in commands.
• Input audio is typically via Pipewire/PulseAudio monitor or ALSA Loopback (use VB-Audio Virtual Cable under Windows)
• The Gstreamer command string for any pipeline can captured and run outside of the app, if desired.
Some differences between GSASysCon and CamillaDSP are:
• GSASysCon can create playback systems made up of multiple remote clients. Audio is sent using RTP over the local network (hardcable or WiFi) to one or more playback endpoints (computer+audio device/DAC). Tight playback synchronization between endpoints can be achieved when their clocks are synchronized using chrony (NTP).
• GSASysCon has no fancy level meters, no GUI, and no flow or filter diagrams except what can be generated manually using Gstreamer
• In GSASysCon, DSP is exclusively IIR filtering via LADSPA as filter-chains. FIR filtering is not currently available.
• GSASysCon was designed for music playback without any particular concerns for latency. Buffer size is fixed at 1024 samples.
• Gstreamer pipelines created with GSASysCon run at a fixed audio rate and bit depth that is chosen by the user. There is a high quality resampler built into Gstreamer that handles SR conversions.
The combination of Gstreamer and LADSPA is a robust and reliable DSP platform for DIY audio processing under Linux, or WSL2 under Windows. I've been using this to implement IIR DSP crossovers for almost 10 years in tandem with my ACDf LADSPA plugin. ACDf implements all the first and second order filter types – it’s all you need for loudspeaker crossovers and PEQ duty.
I have learnt a lot from all the contributors to this thread and have built a number of successful DML’s. Its a very interesting technology that gives great results for not a lot of money which is one of the reasons its been popular on this forum.
This thread is best seen as a stub of the DML thread exploring exciter design. There are many exciters available commercially and they are relatively cheap compared to other loudspeaker drivers so why should we bother looking at DIY options? Same as any other AudioDIY project, to see if it is possible to build cost effective alternatives to commercial designs and, especially in this case, to explore new ideas, push the state of the art and, explore options not available to commercial manufacturers.
Commercial exciter design is targeted at the mass market and so engineering costs are constrained by what the market will bare. All commercial products have to absorb a lot of costs that don’t apply to DIY designs. Take away profit margins, distribution costs, marketing expenses, and some of the engineering costs targeting presentation rather than performance and the amount of money dedicated to sound engineering (sorry!) is less than 50% of the purchase price and for some components can be a lot less.
In the next three posts I will outline each of the areas I am looking at and then, as time goes by, post the results of experiments as they happen. All contributions and questions are warmly welcome as always, but if I end up talking to myself that’s cool to, I do that a lot.
I'm a beginner enthusiast from France, just getting started in the world of DIY subwoofer building for now (maybe more in the future). I appreciate all the great builds and shared knowledge on this forum!
A bit about me:
Totally new to DIY audio— my background is primarily in music production and mixing/mastering at an amateur level.
I’m focused solely on building my first subwoofer—for music and my home studio use
I’m looking for recommendations on drivers, enclosure types (sealed vs ported), amps, and a bit of hand‑holding through the basics etc.
So far, here’s what I’m thinking:
Start simple with a proven design (like a small sealed box).
Choose a budget-friendly driver (around 8″–10″).
Pair it with a plate amplifier for simplicity.
Learn the basics: cabinet building, sealing, wiring.
Hi, sorry in advance for such a dumb request, I'm almost embarassed...
I took apart a BX-9 but forgot to take a picture of how the connectors of the On/off switch go. Is anyone able to help me?
When taking it apart I stickytaped the two reds together, which makes me think they go in the top left and right, and the white ones which have a bigger connector would go in the bottom as they are further apart.
The two reds measure 45VAC between them, between white and red it measures 22.5VAC.
The reds are also connected to ground.
I'm guessing this is how they should be connected since otherwise i would be shorting the two red wires.
Can anyone confirm just for peace of mind so i dont make it smoke up the room? Thanks a lot!!
Iroquois
I am very pleased with this design.
But be aware this amplifier only exist in SPICE simulation.
Supply shuld be 2x18VAC transformer giving like +/-24VDC.
THD at 1 Watt in 8 Ohm is like 0.00035%.
Max output in simulation is 25 Watt.
Upper frequency is like 1.3 MHz, so a lowpass filter should be put at the input.
Hello, I have a problem with this amplifier. It's stuck in the protect. Everything's been repaired, and there are no more problems. If I remove the Q50 transistor (A1266), it runs smoothly and nothing heats up. So I suspect there's a fault in the protection circuit. I've already replaced the TL494 and LM293N just in case, and then it ran for about 15 minutes. Suddenly, it started cycling on and off three times, and then it got stuck in the circuit breaker again.
I have some ROAR15 cabs for sale if anyone is interested. I recently built some paraflex double 15 type c crams which are now used as the subs for my main system. Consequently, my ROAR15 cabs are surplus to requirements and are taking up quite a lot of space. They are great cabs, I wanted more power for a similar size which the type c offer, the trade off is I could move the get the roars in and out of a car / van myself (about 60kgs). whereas i can no longer do that with these type cs (90kgs each).
The current bolt pattern is for Faital Pro 15hp1060 drivers. These are great drivers and work well in those cabs. I guess you could offset your own bolt pattern for other drivers, however, the cab itself will only accept a limited depth of driver due to the cross brace in the mouth. The Faital pro 15 Xl 1400 does not fit (without some hacking of the internal brace) because it has too greater depth!
They don't have handles, you could cut / fit some yourself, but otherwise a rear connector plate, braked swivel castors, custom made waterproof protective covers, and painted using black bedec barn paint (very similar to warnex but more of a matt finish and cheaper). I am open to offers I'd like them to go to a good home and be used. I have at least 4, and possibly 8 available (someone already has shown interest in some but that is not 100% confirmed).
4 of them are made from birch, 4 of them are cheaper class 2 hardwood faced poplar ply. These are the ones I built first and have been used quite a lot with no issue. The accepted price would be reflected based on which ones you chose.
Location near Cambridge. Open to offers. Message me i interested. i can potentially deliver for fuel costs, within reason (2 at a time).
Hi all,
As i said if I managed to fix the Axis motor board I would post accordingly, so here goes.
FIRST A WARNING, the motor board is connected directly to the mains supply so will need VERY careful handling if you are not to give yourself some unpleasant surprises.
Remember electricity will not give you a second chance and until the fuse blows Battersea Power Station or its equivalent is sitting at the other end of the cable. Also any test equipment that is connected to this circuit will be "live", so earthed equipment (scopes etc) MUST either have the earth wire lifted from the plugtop or not used at all. If you have any doubts about your capabilities do not work on this board.
There is a way around it and that is to use an isolating transormer and then you can earth the secondary but it is somewhat a costly exercise.
Also remember the "one hand in the trouser/skirt pocket" rule when using test equipment to measure voltages etc
So..........
My unit would not turn the motor, so first check was voltages on the bridge rectifier, about 320V DC, correct, and also across C5 about 12-13VDC, correct, so I initially suspected the two main coupling capacitors C8/C9, which were removed, measured (both read 33µF, correct value) and substituted for good measure. No improvement of course.
I then turned my attention to the other other electrolytic capacitors C6,C11,C16,and C17. These are all 22µF/63V capacitors and they all measured very low capacity generally 5-6µF. Replacement appeared to effect a total cure, however I was somewhat puzzled when I looked at the sine wave feeding the motor to see it drop in level and the motor stop rotating.
I then remembered Linn's penchant for reducing motor volts so it just turns the platter, the motor relying on the inertia of the platter to keep the motor rotating and vice versa. Once assembled totally all worked fine, with speed being fractionally fast to overcome stylus drag etc, however the speed adjust controls are available through the base of the case if required.
I reckon that the electrolytic capacitors dried out over the years 'til they reached the point of no return and I think that the best solution is to replace them all with new.
Hope this helps you all out there but if you have any further problems send me a pm with phone nr etc and I will try to assist
One Ohm model. One speaker OK, while other has bass ribbon panel damaged & removed...a multimeter fell onto the foil as I was completing the "Silicone fix" years ago (still have not gotten over it)...
Both tweeters are there and functional, albeit both ribbons appear stretched.
I do have a pair of replacement Bass Foils (actually the entire Kit) from Patrick in Germany that I bought many years ago (not cheap at $1300).
Covers are painted Matte Black (used to be that ugly Beige)
Been sitting in my back room past few years....wife would like them gone.
Pics taken years ago attached...
Pick up in West Sub of Chicago, no shipping. I also cannot assist on the repair.
I have a Tact S2150X Digital Amplifier (Silver Faceplate) with volume remote for sale, It has the Equibit chipset, and has Digital only inputs (no analog inputs)...basically an integrated PowerDAC. Its 150 watts @ 8 ohms and 300 watts @ 4 ohms. Amp also has the ability to do high or low pass crossover outputs.
The amp firmware has the latest version (unfortunately, the latest firmware REMOVED the Room Correction ability that the peramps have - why would they do this?)
Works great - I have two, but will be holding onto one. Also have a Tact RCS 2 Preamp (Black Faceplate) that does the room correction also available...
Asking $725 + ship...
Specs:
Power (RMS. per channel) 8 ohm 2 x 150W Power (RMS. per channel) 4 ohm 2 x 300W
Output current (peak, per channel) >50 A
Signal-to-noise ratio ( A-weighted) >110dB
Dynamic range (20 Hz - 20 kHz) >130dB
THD+N (all power levels 20Hz-20kHz) <0.01%
Digital resolution 16-24 bit Linearity (-120dB) +- 0,2dB
Dimensions (WxHxD) 450x140x420 mm 17.7x5.5x16.5 in.
Weight (shipping) 18 kg / 37 lbs
For Sale: NuTube B1 Based Preamp . Two inputs, one output
Unit functions perfectly and chassis is a bit dirty. There is no noise, and only have the two inputs, but switch has ability to add two more inputs. I never needed more than two so it has only two.
Amp has Fender 6L6 tubes, RCA 6SN7 (original used a 6SL7 in driver seat and 6SN7 as PI, but I had two 6SN7), and an RCA 5R4 Rectifier.
Unit originally was a 6L6 / 6SN7 / 5R4 based Carillon amp powering a Church Bell. The power supply reused the original power transformer and tubes, while the output tranny was replaced with a Fender Hot Rod Deluxe Transformer. There were 5 more octal tubes in the original amp (two of which were more 6L6 output tubes) that were not needed for the B15N, so power transformer has lots of spare power.
There were a few minor changes, but most of the circuit is the B15N circuit on the website and attached schematic. Volume, Bass and Treble pots, as well as another pot on front corner for setting feedback, and the one on top to add additional resistance to Cathode resistor. Single input only, Cathode Bias (common 300 ohm) instead of Fixed Bias were biggest changes. Don't remember as this was over 10 years ago. Amp is bolted to a Bamboo cutting board.
This is a working amp and sounds great with my guitars.
Working on rebuilding an NVA AP50 amplifier from scratch, I have a question about the output transistors. NVA is known to have used BDV64/65 and TIP142/147 darlington output transistors in these amps, driven by TIP31/32 transistors. Mine actually has a mix of these BDV / TIP output transistors. Whatever parts one has laying around I guess...
Would a pair of Sanken 2SB1560 / 2SD2390 darlingtons be drop-in replacements? This here thread from 2013 suggests they may be. I looked for spice models but couldn't find any. It looks like the maximum collector current of the BDV is 15A vs the TIP's 10A and the Sankens are 10A also. The Sanken's gain looks to be considerably higher. Internal construction too. The Sanken omits an internal resistor (and a protection diode?).
After building quite a few subwoofers and 2-way speakers over the years, I was looking for something different, and line arrays fit the bill nicely. Line arrays have some intriguing characteristics such as little loss of volume as you get further away, low distortion, and very slim enclosures.
Designing one isn’t simple, though, since drivers interfere with each other, causing anomalies in both frequency response and sound dispersion. To minimize destructive interference and use it to our advantage, it’s vital to minimize driver-to-driver spacing.
When making line array speakers, some companies use a large number of 1" tweeters coupled with several small mid-bass drivers. Unfortunately, the distance between two 1" tweeters can’t be made much smaller than 40 mm, and they don’t really go much below 2 kHz without a waveguide anyway. On the other hand, while 2" full-range drivers result in a slightly wider driver-to-driver spacing of around 60 mm, this is a small price to pay given their ability to reproduce frequencies down to 150 Hz, which is particularly useful for a home theater surround speaker use case I have in mind.
I’ve settled on using eight 2" Lavoce FSN020.71F drivers, featuring neodymium magnets and 3/4" voice coils. I could have used more, but in a typical sized room, they should be more than enough.
Curves certainly look nice, but they aren’t easy to make. One method for creating a curved enclosure is to form it from layers cut from a sheet of wood. I’m not very fond of this technique, as a lot of material goes to waste, but in this case, the enclosure will be very slim, resulting in little waste. Overall, I’ve cut nine layers from 18 mm MDF on a CNC router. Some layers have inner lips to mount the speaker baffle and the rear panel, plus one in the middle for central support. I also applied 1/2" roundovers to the speaker cutouts.
Eight drivers mean a lot of unsightly screws, which led me to mount the drivers from the back. However, that requires either a removable rear panel or baffle (or both) to access the drivers in the future. Since space is tight, I decided to make both removable by using threaded inserts at the back of the baffle and long screws, inserted from the rear panel, to attach the front and rear together.
There are four banks of drivers, each with two 8-ohm drivers connected in series. The banks are then connected in parallel, resulting in a nominal load of 4 ohms. However, two filters are necessary to flatten the frequency response for all drivers: a low-shelf cut and a notch. These bring the nominal impedance up to 6 ohms.
From top to bottom, drivers 1 and 8, 2 and 7, 3 and 6, and finally 4 and 5 form each bank. High frequencies are increasingly shaded by 2nd-order filters for each bank as they move further from the center, which helps reduce destructive interference and control vertical directivity.
The speaker was originally designed for wall mounting, but I made a quick stand from scrap wood, incorporating the wall mount, so that it could stand on its own.
So, how’s the sound? The speaker needs a subwoofer to fill in the missing low end, but once I’d dialed in the subwoofer integration, I was greeted with clean, smooth sound, particularly in the mid-range, and the speaker can go loud for sure. At 95 dB/1m, there was no noticeable distortion thanks to the eight drivers working in tandem. As for the controlled vertical directivity, you can tell the upper frequencies decrease in volume when your ears are not on the speaker’s center axis. The effect is subtle but quite noticeable, especially when you go increasingly off-axis.
The frequency response is mostly within ±3 dB between 150 Hz and 17 kHz (measured outside, gated at 10 ms). The critical 1 to 5 kHz region, where the ear is most sensitive, is quite flat with only a ±1.5 dB variation. There are some response anomalies starting at 10 kHz since the 2" driver isn’t really suited to reproducing these frequencies well, and there are also some destructive effects from nearby drivers. Fortunately, there’s very little content at these frequencies.
Horizontal directivity is wide at ±60 degrees nominal up until 5 kHz, but it starts to narrow somewhat after that.
Vertical directivity is near ruler flat at ±15 degrees nominal down to 1 kHz and gradually gets wider below that. This well-controlled and narrow directivity allows precise steering of sound, which is especially well suited for applications such as surround sound speakers in a home theater, where some listeners are closer to a speaker than others.
The impedance and phase plot shows no major issues other than a few minor resonances. The bump centered at around 2.3 kHz is due to the notch filter that flattens the response.
Finally, a complete build video is available below. Please let me know what you think.
Toshiba TA8233H fully built chip amp 30wpc BTL stereo
Designed & built by polish enthusiast Fryderyk Wrobel (FRD1996) based on Toshiba datasheet circuit diagram compete with generous heatsinking & all cables soldered into the board for power, input & output and including a pair of russian military matched output capacitors (see detailed pix)
The TA8233H was originally supplied to the automobile market and it will run on any DC PSU in the range of 9-18 vdc including of course 12v car battery
The TA8233H chip amp is highly regarded especially in Polish hi-fi circles and this is a great simple well-designed board, for anyone who wants to give it a go. only used once to test it working. part of a larger project that never came to pass
These chips are now somewhat difficult to find new (as this one is) with most pulled from used equipment
£25 priced to sell.
Would prefer to post only to UK. Royal Mail 2nd class small parcel cost of £3, but pleased to use more secure service for royal mail cost
So the other day I picked up a used set of Martin Logan Aerius i ESL's in great shape and I have been really enjoying them. I was thinking it might be a good idea to think about replacing the 20+ year old electrolytic caps in the crossovers with something better such as Clarity caps. I'm fairly new to this but know this is popular with conventional style loudspeakers. I'm comfortable working through parts values but I'm not really 100% sure where my time and money would be best spent so I'm here to take in any info offered.
Martin Logan has proved to be extremely helpful and provided me with crossover schematics and a detailed electrical print on the whole speaker. I have not opened these up yet to take a look for myself but plan on doing so this week at some point. I've attached all the electrical info for everyone's viewing pleasure.
I’m honored to have partnered with @Nelson Pass to incorporate the PASSdiy logo into a Modushop chassis. We are offering this chassis for a limited time.
As you all know, Nelson is incredibly generous toward the DIY community. In honor of his love of animals, we will be donating 10 Euro from the sale of from each chassis to support animal rescue through the Canile Comunale di Bologna (Bologna’s local dog shelter).
I’ve visited the location personally, and their efforts to care for animals in need and find them forever homes is heartwarming. You can learn more through their Facebook and/or Instragram pages and through the Bologna City Hall.
The chassis itself is a 4U/400 Dissipante Deluxe. It includes the standard 4U/400 chassis components with heatsinks pre-drilled for UMS compatibility.
In addition, you’ll receive the perforated base plate and a pre-cut back panel designed for use with the diyAudio store back panel parts kit.
The back panel has two sets of holes compatible with Neutrik D-style jacks. It’s convenient for amplifiers using both XLR and RCA inputs, like the Aleph J. The front panel, featuring a CNC’ed PASSdiy logo, has two pilot holes for power indicating LEDs.
The chassis will work well with all First Watt style amplifiers and has plenty of heatsink for those that like to run their amps a little cooler or crank the bias a little higher. It fits the sweet spot right between the current 4U and 5U deluxe chassis.
We think they’re beautiful, and we hope you do also!
The price for each chassis is 319 Euro + VAT as applicable + shipping. The final cost of your order will be calculated at checkout.
Click below to pre-order your chassis. All critical dimensions and performance characteristics for the chassis can be found there as well.
There is no limit to quantity, but again, this is a limited time offering. At the moment we don’t have plans to offer this front panel again. Pre-orders will be accepted through May 31, 2024. Chassis will be shipped in batches on a first-ordered, first-shipped basis directly from our factory in Italy.
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!
The mentioned thread was very confusing for many members like me. Of course it is full of good knowledge and wise people. But many members are fond of selfishness and some others have been mocking arround.
Please be so kind to contribute with an open mind, avoiding mysteries and sharing knowledge, with a real hobbyst point of view. That's is what a forum stands for: help each other and share good information.
Lastly, please upload schematic diagrams and try to clearly explain what you think and/or suggest.
I bought 2 original boards that work with just 4 MOSFET transistors as power output (not 3). I suggest to start with this first schematic and try to stick to this.
I suggest to order this first part and then we could jump to many other ideas and circuit revisions. But I kindly ask not to fall in the other's forum disorder .
I am uploading the original schematics and board drawings as NagysAudio provided.
Hi, i'm new in those things, this circuit would work well for converting balanced or unbalanced audio from condenser mics to a fully differential balanced signal? the balanced output is meant to be plugged to a mixer or interface TRS input. the 20uF caps are solid polymer ones on anode to anode configuration to work as non-polarized, my concerns are about the SSM2019 driving the transformer, idon't know if i did everything right, i will respond to any question
I've been avidly reading threads and watching vidoes on this site for the last few days and I'm only part way through the Wolverine build thread, so apologies if this question has been asked before (although, I couldn't find one).
I'm registered for the Wolverine 5th Group Buy for two sets of 2x E3-4 boards (including the optional boards), with the primary aim of replacing an aging, but recently re-capped (by me) Meridian 577 (200w per channel driving 8ohm Monitor Audio GR20 speakers). The second set of boards are an "insurance policy", in case I screw-up something on the first set.
So, to my question. Is there a "sweet spot" for the wolverine amp ? My current thoughts are going for the 200w design or perhaps easier (safer?) for a novice, the 180w with 64v rails. Also, given that I'll have a second set of boards, I have the option of selecting different components where applicable, to see how that plays out. Finally, the second set could be built as two mono blocks, rather than the dual-mono design I have in mind for the first set.
I was cleaning out my tool room and found this old measurement mic. It is the "Mitey Mic", I ordered it as a kit from Old Colony Sound Lab in ~ 1991. It used a Panasonic mic capsule in the end of a long brass tube. It came with a small project box and a PCB. I had to solder all the components to the PCB. Power was a 9V battery.
I am sure that others have fond memories of this cool little mic... It helped me design my first speakers back then.
I had no use for it, and it no longer functions, so I pitched it... but I took a few photos before it went into the trash.
Arround 2010 I completed an amp based on LM3886 and was looking for a good speaker. Many friends here in this forum helped me making the amp and finally a great suggestion came from Picowallspeaker directed me to Martin King open baffle. At first the size frightened me. But nevertheless I jumped into it for its simplicity and built one. The details are in the thread Suitable Speakers.
Since then my work and home forced me to look away from electronics. Enjoyed the open baffle for several years and then for a couple of year the music also took a back seat. Termite sneaked in and started eating the baffle without my knowledge. About a month back I tried to put back the system and noticed time has taken its toll on the machines also. Lots of work now.
In the mean time around 2017 Martin King revised his design depending on the full range model used.
At the moment I am now working on this project. Got back my Marantz CD5003 back on its feet. The turn table is alright, the amp is as new as before. The radio FM needs attention as it has gone silent.
The revised crossover suitable for Fostex 103En is done. Only the baffle needs a complete new making. Still looking for material for it to make it better than before. Any experienced suggestion is welcome.
I have a question now. In an open baffle speaker the drivers can be attached to the baffle in two ways. One is place the driver over the hole and another is passing the magnet through the hole and then fix it. In either case I will be curving out the rim of the hole for free flow of the sound waves. My question is which is better/correct sonically. Aesthetically the first option looks better because the screws aren't visible. Thanks for response in advance.
Hi all,
Not sure how these group buys work I haven't ever done one. But the price gulf between buying one versus 12 of these units drove me to post here to see if there's any interest from others in buying a full carton so we can get the lower price. At the lower price I would probably buy two so really looking for 10 other units.
As far as the process goes I have no idea how to make it safe and secure for everyone or what the accepted process is. Would love any advice in that domain
I'm going to pick up some Acoustat M3s (I think they are M3s, not 100% sure of the model) from someone's garage tomorrow. Don't know much about the condition except I was told they work. Looking forward to checking them out. I've always wanted to play with some electrocstatics. Do folks think these are a good speaker, and what do I need to know about them?
How should they be transported? I was planning to remove the assembly from the back and stack them flat on a piece of plywood.
I am planning to power them from a McIntosh MC2105, does this seem like a good choice? It has a 4 ohm output stage.
Or, what I’ll be working on instead of PA speakers and amps over the next couple years.
And so it begins - groundbreaking was this week. I can now drive the site without ending up on the ditch and needing a tow truck. When they cut the hay out there this week they actually left my stakes. I thought I was going to have to mow around the building pads this morning.
Contract for the metal buildings is in the final stage. The temporary mobile home is supposed to go in on the 27th.
I am retired EE, having worked on a wide range of stuff, including undewater acoustics, food processing, self-guided vehicles, signal processing... . I also play the piano, classical, and have a collection of piano recording, both LPs and 78s, which has me interested in mid-range quality audio gear.
Has anyone gotten a Raspberry Pi 5 to work successfully as a DSP crossover music streamier?
I have a working DSP crossover streamer configuration on my Raspberry Pi 2B using Charlie Laub's ACDf filters. The OS is Bullseye 5.10.103-v7+. The 2-channel (stereo) audio stream from mplayer (using -ao alsa) is successfully split into multiple channels and processed by the filters in /etc/asound.conf and this outputs multi-channel PCM 96khz to my AV receiver via HDMI and the resulting sound quality is excellent in my 2.5 way speaker setup. The tweeters go tweet, the squawkers go squawk and the woofers go woof!.
With my new Raspberry Pi 5 running Bookworm 6.6.20+rpt-rpi-27172 I have not been as successful. With sudo raspi-config I have selected 1 PulseAudio (only other choice is 2 Pipewire) and audio output 0 vc4-hdmi-0. On the GUI desktop, I have the options for Stereo, 5.1 Surround, and 7.1 Surround. I have installed the ACDf filters in the (I believe) correct folders /usr/lib/ladspa and for good measure also /usr/local/lib/ladspa, and I have alsa configuration in /etc/asound.conf.
With 7.1 Surround selected on the Desktop:
In terminal $ speaker-test -t wav -Dpulse -c 8
speaker-test 1.2.8
Playback device is pulse
Stream parameters are 48000Hz, S16_LE, 8 channels
WAV file(s)
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 24 to 262144
Period size range from 8 to 87382
Using max buffer size 262144
Periods = 4
was set period_size = 65536
was set buffer_size = 262144
0 - Front Left
4 - Center
giving sequential 8 channel audio output Front Left, Front Center, etc. to all 6 drivers in my system. The audio is mismatched with the drivers, but I had the same issue on the Raspberry Pi2 and corrected everything using the t-table in /etc/asound.conf.
In terminal
$ sudo speaker-test -t wav -Dpulse - c 8
speaker-test 1.2.8
Playback device is pulse
Stream parameters are 48000Hz, S16_LE, 1 channels
WAV file(s)
ALSA lib pulse.c:242🙁pulse_connect) PulseAudio: Unable to connect: Connection refused
Playback open error: -111,Connection refused
In terminal
$ sudo speaker-test -t wav -Ddefault -c 8
speaker-test 1.2.8
Playback device is default
Stream parameters are 48000Hz, S16_LE, 8 channels
WAV file(s)
ALSA lib pcm_params.c:2226🙁snd1_pcm_hw_refine_slave) Slave PCM not usable
ALSA lib pcm_params.c:2226🙁snd1_pcm_hw_refine_slave) Slave PCM not usable
Broken configuration for playback: no configurations available: Invalid argument
Setting of hwparams failed: Invalid argument
Using the sudo causes the action to fail.
mplayer (using -ao alsa) now gives audio output to Front Left and Front Right channels only at 44.1khz which is the same 44.1khz as the internet radio stream indicating the resampling of /etc/asound.conf is being ignored.
mplayer (using -ao pulse) gives audio output to all channels at 44.1khz which indicates the resampling of /etc/asound.conf is being ignored. This is confirmed by turning off all the outputs in the t-table.
aplay -l reports the following. ** List of PLAYBACK Hardware Devices **
card 0: vc4hdmi0 [vc4-hdmi-0], device 0: MAI PCM i2s-hifi-0 [MAI PCM i2s-hifi-0]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: vc4hdmi1 [vc4-hdmi-1], device 0: MAI PCM i2s-hifi-0 [MAI PCM i2s-hifi-0]
Subdevices: 1/1
Subdevice #0: subdevice #0
I know that PulseAudio is looking at the configurationi in /etc/asound.conf because I can comment out a formatting character such as #} and this will throw an error with mplayer. It just appears that the filters are being ignored by PulseAudio. Maybe the Pi5 OS being 64bit and the filters were written for 32bit has something to do with it?
Kit acquired from @hifiamps (no affiliation otherwise). Original intention was to pair with Rasberry Pi, then I saw @6L6 implemented without. I used:
Mark’s kit so far bog stock
9V trafo
Simple LM317 PS graciously provided by @manniraj (he threw that in when I bought a working MiroAD1862 PCB and PSU1)
Amanero 384 combo input board, an Ali clone
IE socket and fuse holder sourced long ago from ApexJr.
A bit of scrap MDF board
Two issues now resolved- silkscreen on the input card connections didn’t match the ProtoDAC assembly instructions which is apparently a common problem. Searching terms on this forum informed how to move one wire. First fire up yielded a loud crackly sound both channels. @6L6 advised quickly and accurately to add a ground wire between input board and the Proto. Fixed!
So far only played through a scratch built El84 tube amp based on an early Mullard circuit and tiny full range bookshelf speakers, Dell laptop win11 straight in. Sound is very pleasing, quite prominent bass.
Quite something the ProtoDAC, four capacitors five resistors and a daughter card with eight little Philips Dac chips under $100. Next step to listen on the bigger system IronPre + biamped SIT amps.
Not sure if it would benefit from an AmyAlice filter- one is on the way from @ItsAllInMyHead
Since the MOSFETs used are tricky to solder effectively at home, the boards will be provided with the MOSFETs already soldered. The rest of the parts will need to be supplied by the subscriber. They are easily available and in stock at Mouser, DK etc. The modules use SMD parts with the smallest at 0603, but most are larger than this - I'm only saying this as some people do not like using SMD parts. Full BoM and build/test instructions will be provided on paper as is usual.
Technical details:
The built units can be used for amps with total voltage from 24V to 100V, either single or dual rails. It must be run from the existing amplifier rails and you do not need an additional auxiliary supply. Current draw is less than 15mA per rail. The boards are designed to be mounted on the amplifier heatsink - one TO-126 transistor is configured to be bolted to the heatsink - it will dissipate up to 1W.
If using BoM values the units provide a start up delay to avoid turn-on thumps of about 3s and will trigger after the DC level exceeds the threshold for around 2s. The trigger voltage is around 1.2-2V depending on components used.
Board dimensions are 30x58mm
The boards are designed so that a mirrored pair can be made using the same PCB, making input and output wiring simpler and neater for each channel in a stereo amp.
By design, the protection is latching - i.e. it does not automatically reset for a long period. You will want to figure out why there is DC on the output before resetting the latch! It is also possible to manually reset the latch for testing etc.
Costs:
€60 for a set of 4x boards, with 2x MOSFETs per board pre-soldered. Tracked and insured postage anywhere in the world is €15. The rest of the parts will cost about €30 per stereo pair without any volume discount, depending on where you shop for them.
If you would like to take part, please add your DIYaudio name below here along with the number of sets you would like to sign up for. 1 set = 4 channel protection PCBs (2x stereo amps).
Example:
woodturner-fran - 1 set
The group buy will require a minimum of 20 participants/sets. When the group buy closes, it is expected that ordering and delivery will take about 3 weeks, and once the units are here and verified, I will send invoices to each participant. As usual, no payment until the boards are here and ready to distribute. Previous experience shows it will take an average of 2 weeks to arrive anywhere else in the world.
As electronic engineer and audiophile, I've always dreamed of building an amplifier that would satisfy my own standards. This would not only test my professional knowledge but also be a way to combine my hobby with my work. Though the idea sounds wonderful, the reality was far from easy. Constant interruptions and delays kept pushing back this grand plan.
In truth, the delays weren't just because I was busy, but also because I was unsure of what kind of amplifier I wanted to DIY. Common designs on the market—whether tube, BJT, or MOSFET—didn't seem to be the answer I was looking for. After years of indecision, I finally made a choice in the summer of 2020: to build a no-feedback amplifier that would meet the standards of high-end amplifiers. When I made this decision, it even shocked me. As an electronics engineer, my instinct told me it was an almost impossible task. Despite countless thoughts of giving up, my competitive spirit kept bringing me back. I was too curious to know what a no-feedback amplifier would sound like. This question lingered in my mind for a long time. Out of curiosity, I asked ChatGPT, and here's the response I got (Oh, how I wish ChatGPT had been around back then):
Thus began my DIY journey, filled with challenges and joys, to experience the sound of a fully no-feedback amplifier.
Since Bell Labs engineer Harold S. Black introduced the concept of negative feedback in 1927, it quickly gained widespread application in audio amplifiers, achieving great success. For example, the Williamson feedback high-fidelity amplifier, released in 1947, used deep global feedback to significantly reduce distortion, extend frequency response, lower output impedance, and increase damping factor. These improvements made the Williamson amplifier the benchmark for high-fidelity sound systems at the time and a classic cherished by audiophiles today. Most modern amplifiers are still designed with appropriate feedback to improve measurable data.
However, humanity never lacks pioneers who dare to challenge conventions. Many engineers have been working to reduce the amount of feedback, hoping to achieve better sound through shallow feedback. Products from brands like First Watt, Pass Labs, and NAT Audio are successful examples of this approach. These amplifiers remove global feedback and only use local feedback to maintain performance, aiming to reduce the negative impact of global feedback on sound quality. Though their test data may not be perfect, they focus on sound performance and have been well-received in the market.
The success of these products greatly inspired me. I believe that following in the footsteps of earlier engineers can take me further. As an engineer, I value objective data, but as an audiophile, I also think subjective listening experience is crucial. After all, good test data doesn’t always guarantee good sound, and if the sound is pleasant, why worry about perfect data? The quality of an amplifier cannot be entirely defined by one set of parameters, at least not with today’s technology. Objective data gives engineers direction, but subjective listening remains the ultimate measure of an amplifier's quality.
So why did I ultimately choose to challenge myself with the “hell-level” difficulty of a no-feedback amplifier? There’s a small story here. I had a friend who was an audiophile. He once told me that he had listened to amplifiers without global feedback and found the sound more natural, lively, and impactful. He asked if I could build a fully no-feedback amplifier for him to try. At the time, I just smiled and didn’t give it much thought. But his words planted a seed in my mind, which eventually led me to make this decision and explore the mystery myself.
In fact, I wasn’t sure what a no-feedback amplifier would sound like either, or if it would really be as natural and lively as ChatGPT suggested. But debates in technology often have their complexities. Feedback indeed brings many benefits but can also introduce issues like transient intermodulation distortion and dynamic errors caused by feedback delay. Could these distortions negatively affect subjective listening? Perhaps building a no-feedback amplifier with excellent parameters is the best way to answer these questions.
This is the story and motivation behind my DIY no-feedback amplifier, as well as my views on audio technology and subjective evaluation. In the upcoming posts, I’ll share why I chose Class D as the main architecture, and the challenges and results I encountered during simulation, production, and debugging. This DIY journey took four years, and some details may have faded from memory, but I’ll try to consider readers of all backgrounds, using analogies and simplified technical details to help everyone understand. I hope my sharing will bring you some inspiration.
In theory and in practice, if you put multiple drivers together as in a line array, you are increasing the surface area of the cones. That in itself should be enough to push more air as if one were using a singe larger driver. However, according to one speaker designg program, this configuration does not afford any lower frequencies as if it was a single large driver.
So, could some one tell me why this would be so?
Thanks in advance.
Various reports by Pocket Lint AudioXpress Igloo Dome What Hi-Fi
This thread is a measurement/review of the custom 6” drive unit from the system. A few of us were fortunate to acquire this unicorn.
Background- the midrange unit is based on a midrange unit from the CT8LR custom-install system:
The main difference between this and the CT8 driver is that this has a yellow cone, but the similarities include a twin neo magnet, allegedly for symmetric BL(x) and Le(x).
Like the 800 series midranges, it uses the woven Kevlar cone, and dispenses with the traditional surround, instead having a thin foam “Fixed Suspension”. This surround (and spider) affords the cone only a few mm of movement, BUT perhaps because of this, reduces surround radiation distortion
So low, In fact, that the microphone was not able to accurately measure the harmonics:
Sonarworks Xref20 mic:
Incredibly, it has the sensitivity of pro-sound midranges- ie. >95dB/2.83V/1m, yet displays a smooth and extended frequency response to past 10KHz, and mild breakups only occurring past 3KHz.
B&W measurement of individual drivers in CT8 cabinet (blue trace)
When this first arrived to me in Australia, I plopped this driver into an existing test box and took some measurements, including on axis, 15, 30 and 45 degrees off axis.
The sine sweep had a gate of of 5ms (measurements good down to 200Hz) and since it is an 4 ohm nominal driver, I tested with only 2V drive at 1m.
The baffle is 8 1/4” wide; with no roundovers for edge diffraction control- this contributes at least 2dB to the peaking at 4KHz. Also the driver is surface mounted, which causes raggedness of at least +/-1dB anywhere above 2Khz. So this quick/dirty measurement on an unoptimised baffle.
Just add 3dB to figure out the 2.83V sensitivity- that’s almost 100dB /2.83V
Now, to find woofers and tweeters that can keep up!
Edit: One of the Sound System’s tweeter is measured is post #37
Edit 2: April 2025- less fiction
I have kitted up Matched IFP240's and IRFP240 / IRFP9240's. Plus I have a few other odds and ends in stock.
June 2025 Stock Update
MATCHED
Fairchild IRFP240 - In Stock (~400+)
International Rectifier IRFP9240 - In Stock (~600+)
Harris IRFP240 - Qty 24
Vishay IRFP240 (Qty 1500) to be matched when Fairchild 240 stock is low.
MATCHED
Vishay IRF9610 - In Stock (~40)
Harris / Fairchild SFP9610 - In Stock (~300)
LIMITED STOCK / DISCONTINUED
On-Semi FQP3P20 - SOLD OUT
On-Semi FQP3N30 - In Stock (19)
MISC (not matched)
Toshiba 2SC5200N (Qty 23) - make offer
Vishay MUR3020 (Qty 8) - make offer
SMC - MBR20200CT Rectifier (Qty 32) - make offer
Toshiba 2SA1837 (F,M) Qty. 10 - make offer
Toshiba 2SC4793 (F,M) Qty. 10 - make offer
IRFP240 & 9240 MOSFETs Vgs is measured at 170mA in a steady temperature room and with a timer circuit for consistent and precise matching. 9610's and FQP's Vgs is measured at 16mA.
Each set is matched within +/- 5mV (0.1%). Often closer. See pix.
Building an Aleph J Clone?
8 pcs Precision Matched Vishay IRFP240 MOSFETs (1x8N or 2x4N)
$40 + Shipping
Building an Aleph 60 or Aleph 2 Clone?
24 pcs Precision Matched Vishay IRFP240 MOSFETs (4x6N)
6pcs Precision MatchedHarris / Fairchild SFP9610 (2x3P)
$144 + Shipping
Building an F5 V2 Turbo?
8 pcs Precision Matched Vishay IRFP240 IRFP9240 MOSFET Pairs (2x2N + 2x2P)
2 x Matched Pair of IRFP240
2 x Matched Pair of IRFP9240
$45 + Shipping
Building an F5 V3 Turbo?
16 pcs Precision Matched Vishay IRFP240 IRFP9240 MOSFET Quads (2x4N + 2x4P)
2 x Matched Quad of IRFP240
2 x Matched Quad of IRFP9240
$85 + Shipping
Inquire for other quantities & Combinations. Pairs, Triples, Quads, Sextets, Octets, etc. Aleph Classic with IRF240 & 9610s. F1? F2? Fixing a monster Ampeg bass amp?
$6/kit shipping to USA. Inquire for multiples
Payment by PayPal.
International Shipping via USPS First-Class Package International Service® - Estimated Delivery Time Varies by destination. Price per first kit. Inquire for multiples.
$13 Canada
$16 UK
$17 Western Europe
$19 Australia/NZ
$19 Japan
I acquired this amp three months ago, but with life getting in the way, I literally only powered it up last Monday afternoon. Essentially it runs through a start up procedure, the screen works, a fair bit of relay clicking and then it switches off. My own fault for not checking it sooner. I’m half decent with investigating and repairing, and even managed to get the schematics from XTZ. A lot of Chinese though and could be clearer. One 4700uf 63v cap on the power supply board had leaked, actually leaving a leg on the board. Lots of nasty glue everywhere but I’ve cleaned that away, checked the four main 12000uf caps, and they test within spec. I’ve replaced both 4700uf caps, sorted out a couple if dry joints and thought that obviously bad cap could be the issue. No. Found more dry joints on various voltage regulators and elsewhere. Still no joy. There is no response to any control input apart from power on. The overall build of this thing makes it worth the effort, but if anyone has any ideas they would be much appreciated. I’ve never seen an amp with so many screws, it’s depressing!
Hello,
I am looking for an equivalent for a discontinued diode with the same specifications. D571 in schematic
MTZJ5.1B Glass type
5.1v 500mw 5µA@1.5 V 70ohms
Thank you for your help
Edit Feb 19, 2024: The BTSB was tested and verified to produce 49Vpp with balanced 4.9Vpp input and balanced output. This means it can easily drive 0dB amps like F4 to clipping levels. The LME49724 datasheet says that it should be able to drive 600ohm loads to 52Vpp with low distortion levels.
Edit Oct 22, 2020: new boards v1.3 for TH and v1.2 for SMT have corrected them layout error and verified to be working properly in all aspects.
Edit Sept. 4, 2020: Stop Press! Please note that there is an error on J36 (the phase is flipped). I am ordering new boards and will send out replacements to folks who have already purchased these. This only afftects one of the SE outputs. If you have no need for the SE output, it doesn't matter. There is a workaround to it here. So sorry about this but it was the one thing I did not check on the verification build.
Edit Aug 18, 2020: Mouser shopping carts for the BOM here. Thanks to Vunce for making these for us!
This is the Best Thing since Sliced Bread (BTSB) Buffer that I will be offering as a GB. It was designed by Jhofland to address a need that I had with my TPA3255 Class D amp which required balanced input with gain. Jhofland designed a prototype that had just balanced output with gain (selectable via jumpers) with either balanced or SE inputs. I tested it out and it works great.
Here was the v1.0 verification prototype that I built:
Member Redjr has already used the prototype version on his very nice TPA3255 amp build here.
I thought that a more universal buffer would provide a choice of either SE/Balanced input and be able to driver SE/Balanced outputs simultaneously. This might be useful for driving a subwoofer amp, for example. The topology uses a state of the art OPA1656 for the input buffers and state of the art LME49724 balanced line drivers and another OPA1656 for the SE outputs. Another OPA1656 provides a bootstrap for the input amp to allow lower distortion drive at higher drive levels close to rail voltages. The power supply is provided with a very low noise Murata isolated DC/DC converter to take 12vdc from a wall-wart to +/-15v using the DC/DC followed with a CLC filter. The BTSB is absolutely silent and adds no noise or coloration. It is very transparent and can be used as a simple signal booster in cases where you need just a little bit of extra bump in the output from your source or existing preamp.
With the ability to accept SE or balanced and to drive SE or balanced, and with your choice of 0dB, 6dB, 14dB, or 20dB gain selectable with a convenient 8-position DIP switch - you will find it an indispensible part of your electronics toolbox kit. I think that you will agree that is may be the best thing since sliced bread...
To see the schematic, please click below. It is too wide of an image to display inline.
Here is the layout:
Here is a photo of the PCB with a Neutrik XLR jack (plain Molex KK terminals are also provided if XLR is not needed):
Here is a measurement of a 1kHz low distortion signal from Victor’s oscillator and 20dB gain from SE to Bal mode and fed to XLR input of my Focusrite - the so-called loop back test with BTSB in the loop. It has noise and distortion below the measurable limit of the Focusrite.
Here is a measurement of the distortion of my TPA3255 w PFFB as driven by the BTSB in SE to Balance out with 20dB gain applied - the THD is lower than the TI factory data at the same condition. The noise baseline is very low at -130dB:
I'll post a photo of the populated board as soon as I get back from vacation.
We can get the GB interest list started. Boards will be 1.6mm thick, 1oz copper, ENIG finish, and green color solder mask. Construction is straight forward as it is mostly through hole except for the opamps. Price is $23 ea for the bare PCB. The BOM cost is about $65. Please see BOM below - I will post a Mouser shopping cart to make ordering easy (soon).
I have received questions about whether or not a version will be provided with the SMT parts pre-installed. Yes, I can do that and the cost for a BTSB with 4x OPA1656 and 2x LM47924 pre-installed will be $69 ea. The 6 opamps cost $21, and I am charging $25 for the SMT soldering service.
Also available in all SMT - 70mm x 70mm. No XLR jack. Same price as TH version (Schematic and BOM here).
Please add your name, desired boards, SMT pre-pop'd Y/N, and country below.
BTSB Buffer GB Interest List:
-----------------------------
Name No. of boards SMT pre-populated (Y/N) Country
There is a new panel mount version of the BTSB that has separate auto-switched RCA inputs. Anytime something is plugged into XLR/TRS, a relay automatically disconnects RCA's. This version also uses OPA1637 for balanced output driver (uses less quiescent current). Mounts directly to rear panel and no mounting standoffs needed. Very compact and conserves chassis real estate on bottom panel. Also reduces flying leads that can pick up noise.
Fit check with 3D print:
Here is panel hole template. Note that the vertical CTC distance below is for the Elecaudio RCA jacks. If you purchase the RCA jacks from my shop or the RTR BTSB Panel Mount then the distance is 7.25mm:
Board dimensions:
BTSB TH v1.3 board dimensions are 100mm wide x 88m high.
BTSB SMT v1.2 board dimensions are 70mm wide x 70mm high.
BTSB Panel Mount SMT v1.2p is 132mm wide x 45mm high.
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 🙂