I completed this amp today (L12-2 v 4.2) and left the whole amp untouched as bought. My thinking was that I'll do the "Mods" at a later date as I have room to easily remove the amp boards from the enclosure. My initial reaction was a disappointment to me. I have used the LJM L20 V9.2 boards with film input capacitor being the only mod. To me the L20 amp was pretty well perfect. Very neutral with no overall audio signature, just as if the input was driving the speakers with nothing in between (yes, I know it's impossible, just a sort of illusion) However the L12-2 is definitely brighter making all music styles just seem colder and unnatural. The bass is a little odd too.... seemingly less controlled and fatter. Yes, so far, a disappointment but the ads do carry a "Warning" saying the amp is bright!... There is a YouTube video showing a subtle rise in the frequency response. It seems very strange to have a non-flat frequency response in a modern amp, but I reckon this is enough to explain what I can hear, as it's over a wide upper response area. I would be VERY interested to hear what others think of their L12 amp used with highly neutral and detailed speaker. Thanks
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I have always been facinated by the Synergy/Unity/Multi Entry horns. I have read i think most of the threads here on DIY audio and elsewhere on the web including danleys patent and more... After the DFM groupbuy here on DIY I suddenly have three compression drivers in my hands.. So now I no longer had any excuse and have begun experimenting myself!
I also have quite a few of the small SB full rangers (SB65WBAC25-4) on hand. I have no idea how good these drivers will be as MEH mids, but we will se. I was kind of fantasysing about the Scanspeak 10F/4424G00 based on the 2*FS/QES formual they score 562 which to my understanding is pretty good?

The dream is to build three bigger 3-way horn for use as Left, Right and Center fronts in a future home cinema/listening room.

However, for now I just want to do some smaller scale tests with what I have to see if my understanding of the concept is correct as well as do some experimentation.

While I understand that the corner placement of taps in the horn reduces the taps impact on the HF performance I really would prefer an eliptical horn based on the ATH formulas from a visual standpoint. Therefore the first experiment will test if it is possible to get a decent HF performance from such a OSSE horn with taps in it.

So I started by designing a parametric horn that has replacable taps to quickly iterate differnt shapes and sizes of the taps. This is what it looks like with the midranges attached

The recent disputes regarding the lowest-noise operational amplifier, mainly as a single-stage phono EQ, have prompted me to ask:
how can I get a concrete reference point as quickly as possible, comparing individual types of OP?

Whether absolute values are generated is irrelevant, it is the comparison that is of interest. The type with the highest SNR wins.


SCHEMATIC



The win32 program noise.exe opens a cmd and presents itself as follows:

OPu=
OPi=
G1k=
Rq=
Ltc=
Rdc=
R1=
R2=
R3=
df=
T=
1.730631e-007[V] 1.747937e-005[V] 88.53378[dB]
99.0099 7500 3112.292 10000 100 3333 47000

OPu=

C:\Windows\System32>


enter as a normal floating point variable or confirm with the enter key, the rest will become clear as you play.

• OPu stands for en in V/sqrt(Hz) and OPi stands for In in A/sqrt(Hz) of the operational amplifier used
• Rq stands for the resulting effective resistance of our pickup, alternatively a DC resistance (in ohms) and a coil inductance (in H) can be specified
• df stands for the desired bandwidth of our thermal noise (in Hz)
• T for the temperature (in °C) ...

If you assign a value to OPu after the first run, the input starts from the beginning (you can simply confirm (the stored default values) with the enter key).


#
Have fun with it - and of course I cannot guarantee correctness and correct operation.
However,
the theoretical calculation of the noise, manifested here in the SNR (last output value of the first line), is actually trivial.

Nevertheless,
I could have made a dozen mistakes.


HBt.

So I was sitting when all of a sudden a nasty tone started blasting through my speakers that made me jump. I did a little investigatigating and figured out it's my preamp. I took a picture of an oscilloscope hooked up to the output of the preamp but don't know how to interpret it. This is with nothing input to it just the power turned on.

My name is Pablo, I’m from Argentina, South America, and I’m an audio enthusiast with a deep love for high-quality sound reproduction.

Right now, I’m planning to build the Faital 12-430 speakers designed by Troels Gravesen. I already own a pair of DTQWT-mkIII, also designed by Troels (heavily modified, I must say), and they’ve given me countless hours of listening pleasure.
On the electronics side, my main setup consists of push-pull tube monoblocks with 6L6WGC tubes, and a preamp using RCA 6SN7WGTA tubes. I’m also working on another project: a KT120 push-pull tube amplifier, with diode rectification and fully point-to-point wiring.

Aside from all that, I’m also a bit of an IEM fan—love playing around with different DACs and digital sources to explore new ways of enjoying music.

Looking forward to exchanging ideas, learning from you all, and hopefully contributing to this amazing community of audio lovers!

Hi experts,

I'd like to put my thought or plan or whatever to get experts' advices on it.

I have been thinking of a new 3-way speakers around the same ATC SCM75-150s, complemented with Scanspeak D2104 21mm tweeters and some 8" double woofer instead of a 10" Volt B2500.1.
Then I was invited by another DIYer for auditioning his new speakers using Raal 70-20 tweeters, Scanspeak's freakin' expensive 83mm dome midrange, also Scanspeak's illuminator 7" as a midbass, along with Scanspeak's 13" woofer. He originally planned a 3-way without the midbass, but got an advice from a professional to add the midbass in between. It sounded awesome, and also seem to prove the superiority of the 4-way design.

So, I began to wonder what if I add midbass between ATC and 8" double woofers. My logical half tells me professionals and experts will say midbass is not necessary with 8" double woofer below. But I am still curious. What are your thoughts about this?

Regards,
Jay

I am considering using the 6" SB Acoustics ceramic woofer in the system thats going in the kitchen. The wife is questioning the ability to clean this cone. Valid considering we sautee food in there on a consistent basis.

What does it feel like? Is it smooth and easy to clean? Or is it sort of porous and the white might end up looking horrendous in a handful of years?
https://www.madisoundspeakerstore.c...coustics-sb17cac35-8-6-ceramic-woofer-8-ohms/

I'm new to car audio.
Can someone tell me how to best wire up the attached.
I'm done feeling stupid.

I have been looking for 2SA1837 & it's mate and noticed that Digi key has some Transistors made by a company called EVO are these as Good???

Thanks for everyone's' time.

I have a question regarding my preamplifier external psu

We have 4 transformers 2 in each channel all power up with the same mains switch on the front unit. I have read the its not good for the tubes the b+ and the heaters power up at the same time. So because i dont want to mess up with delay boards and too complex soldering, can i put a second mains switch seperate from the heaters because i have the advantage of separate transformers for hv and lv and power on heaters first and after a couple of seconds the hv or leave it as it is now in the risk of starting a fire again?

I forgot to tell you that the second transformer at each channel is output 15v ac for the preamplifier heaters and then goes to diodes, caps and 7812 SPARKOS regulator for a steady 12v dc and 6.3 v ac connected directly unregulated to the ECL82 tube HEATERS.

Is it safe then if i go with the 2 mains switches to power up the heaters transformer first but the ecl82 will have the heaters on but no hv to regulate which is coming next after i power on the hv transformer?

I have an F5 with Toshiba laterals. I have the source on the power lines now; however, with my Maggie at 4Ω it is rather too harsh sounding. So back to some source degradation.
What do you think of the following development path? [I used to have such a version with the source resistors and it sounded good actually so after some years I am thinking of 'going back'.]


The difference with my 'standard F5 build' is the drain resistors and the fad of the 3Farad capacitor. In fact, that forces a nice harmonic degradation on paper.

Hi,
Does anyone know how to access the internal circuit boards? I can't see any obvious or easy way
Thanks
David

For sale is a complete working Zen v4 amp.

I am using Jim’s Audio PCB boards. All of my speakers need a more powerful amp (and I built 2), so time for this project to have a new home. The case is pieced together 4U parts from this site’s store since I already had some previous 4U parts. The back is vinyl wrapped.

New lower price of $400 and now including a set of blank F4 boards, full set of IRFP transistors for it, the original blank Honey Badger boards and the negative side of the blank power supply boards in case you want to put something else in this case.

If you’re in the Detroit area you can hear it, otherwise I’ll ship it. If you’re committed to purchasing it, we can FaceTime as proof of operation.

$650 $400 plus shipping.

I have for sale stereo amplifer, based on 3e audio mono 480-1-29A modules, precisely installed in custom build beautiful enclosure. I put a lot of effort to squeeze the best sonics from this modules, providing special SMPS with quality components and additional capacitor board (designed by myself) to further improve the sound.

Stereo amplifier consist of:

- 2x 3e 480-1-29A amplifier modules (latest version with OPA1656 opamp and Nichicon capacitors), bought at Audiophonics. More info here: https://www.audiosciencereview.com/...tpa3255-amplifier-kit-480-1-29a-review.50283/

- Microaudio SMPS 600R2 power supply (+46Vdc).

- additional capacitor board (per each channel). It consist of Nichicon FW 1000/470 uF and KZ 100 uF capacitors, bypassed with 1 uF Mundorf MCAP MKP capacitor

- aluminium enclosure specially made for this amplifier. Front and side plates are made from 10 mm aluminium (CNC machined). Other plates are made of 3 mm aluminium. Enclosure has extremely fine black power coated finish.


For wiring speaker lines, Jantzen solid core wire was used. For power supply wiring, Jantzen stranded wire was used. Dimensions of amplifier are: 28x34x9 cm (WxDxH). Amplifier is in perfect operating condition. Input is provided with high quality Neutrik XLR terminals. Weight of this amplifier is a little over 6 kg, so not big deal with shipping.


Price is 500 EUR + PayPal fee + post costs



NOTE: I am not responsible for any customs fees, taxes, import duties.

Hi, i would like to build a basic amplifier using readily available kits.
what i read about LJM kits vary a lot from one thread to another. Some say they are great kit and some other simply burn them.

i already have a psu with 42v rails at 500VA so this is my main starting point.
i would like to have at least a modest 50wrs into 8 ohm, more is welcome.
MX50X2 seems a good candidate.
MX50SE seems on the edge for 42v mains.

The amp will be used in a bi amped system, running the upper range (not bass).

any suggestions is welcome, links to trusted sellers too.
thanks.

Απλώς να κάνετε όπως ζητάτε και να λέτε ένα γεια.

Αυτό είναι το πρώτο μου βήμα σε ένα φόρουμ και στα ηλεκτρονικά, γι' αυτό παρακαλώ να είστε ευγενικοί!
Το θετικό είναι ότι είμαι αρκετά πρακτικός έχω ένα καλό εργαστήριο και απολαμβάνω να μαθαίνω νέα πράγματα και να φτιάχνω πράγματα, ειδικά όταν είναι κάτι που δεν έχω ξανακάνει.
Εδώ και πολύ καιρό ήθελα να αναβαθμίσω το Hi-Fi μου και αυτό φαινόταν λογικό.

Hi, I am planning to build a PP amplifier with a global negative feedback. This is a general question: which tube would give the most musical sound?

let's say you're looking for a driver to cover 1.5 khz to 10 khz with high efficiency.

you can use a Planar like the Radian LM8K with response like this:



or you can use a Compression Driver like RCF ND650 with response like this:



between the two of them the Planar slopes up by about 10 db and the Compression slopes down by almost 10 db

i have been trying to research why but the answers are not fully satisfactory.

the explanation for the CD is supposed to be "mass break point" but after extensive googling and reading this JBL paper:

https://jblpro.com/es/site_elements/tech-note-characteristics-of-high-frequency-compression-drivers

i was not able to find a single explanation anywhere on the internet of what exactly mass break point is and what causes it. the formula given by the JBL paper:



takes only motor force ( BL^2 / Re ) and diaphragm mass ( MMS ) as input and magically puts out 3.5 khz as output. it further blatantly states that no matter what driver you use it will always produce about 3.5 khz output because a larger driver will have twice the motor force and twice the diaphragm mass.

this formula is maybe the strangest i have ever seen. why would higher motor force result in higher bandwidth ? my theory is motor force here is simply a stand in for voice coil diameter because all compression drivers have approx 2.0T flux strength in the gap so motor force in real drivers will be proportional to VC diameter and of course MMS will also be proportional to driver size and JBL states in the same paper that Mass Break Point is always around 3.5 khz regardless of what driver you use so it seems the formula is simply designed to basically take driver size as both numerator and denominator in order to have everything cancel out and leave you with just 3.5 khz as result.

the paper further states that the space between diaphragm and phase plug ( about 0.5 mm in most drivers ) is not large enough to matter and is NOT the mechanism behind mass break point.

so what is the mechanism then ?

and it is equally mysterious why Planar response rises. most people intuitively understand that it has something to do with beaming because the power response of the planar is flat, only the on-axis response rises. but why isn't the same effect observed in an array of cone drivers ? why doesn't an array of cones exhibit a rising response ? why only planars and ribbons ?

i was able to come up with a theory that the reason for discrepancy is the low mass of the diaphragm. that is heavy cones are more efficient at lower frequencies where mutual coupling increases acoustical radiation impedance ... while ribbon / planar diaphragms work better with lower acoustical impedance at higher frequencies. this would account for the kind of response rise we see in ribbon tweeters, whereas in the Radian planar there is additional bump around 10 khz due to cavity resonances as well as membrane resonances so i could accept the planar response as being explained by a combination of these factors.

but i still don't understand mass breakpoint of compression drivers. why is it a thing. and what is it ?

i can only assume it has something to do with the phase plug because only drivers with phase plugs seem to exhibit this phenomenon. a 2.5" titanium dome like in RCF in questions should be able to go to 8 khz or so before breaking up yet it begins to roll off much earlier, even in on-axis response, and probably around 3.5 khz as JBL says when measured on plane wave tube / power response.

it must have something to do with the mass of air in the phase plug acting as a bandpass port. but, as i said, i was not able to find a single explanation anywhere online as to exactly what it is.

on a somewhat unrelated note i think it is noteworthy that at 10 khz both the planar and the compression driver are at about the same 105 db efficiency, while at 20 khz a true ribbon beats them both with over 100db efficiency. if radiation patterns could be matched a perfect speaker would use a true ribbon over 10 khz, a planar from about 3 khz to 10 khz and a compression from about 600 hz to 3 khz but nobody optimizes drivers for those frequency ranges and in practice both planars and CDs are designed for the same frequency range and you must pick one or the other, but for that you have to first understand both, which is what i am trying to do.

Hi everyone,

I’m diving into Class D amplifier design and exploring Class TD (Tracking Class D) in parallel, and I’d really value the community’s insights to help me get started. My ultimate goal is to design a 3kW-4kW Class D amp at 4 ohms to power 21" and 18" outdoor speakers within the next two months. As a learning step, I’m focusing on building a 500W RMS Class D amp at 4 ohms. At the same time, I want to learn about Class TD to understand its benefits for high-efficiency, low-THD designs, aiming for a future 5kW output with minimal distortion for my outdoor audio setup.

My background: I’ve successfully built a 1000W RMS Class AB amp at 2 ohms, so I’m familiar with amplifier design, power supplies, and testing. I have a well-equipped workbench with an oscilloscope, load resistors, signal generators, and other tools. While I’m comfortable with Class AB, Class D and Class TD are new territory, and the switching concepts feel daunting. I’ve been searching online for resources, but I’m struggling to find clear, practical guidance on where to begin for both topologies.

Here’s what I’m looking for:

1. Roadmap for a 500W RMS Class D Amp (4 ohms):
   • A step-by-step approach to design a 500W RMS Class D amp, covering key components like PWM modulators, MOSFETs, gate drivers, and output filters.
   • Reference designs or schematics for a 500W Class D amp that I can build and learn from. Are ICs like IRS2092 or IR2110 suitable for beginners?
   • Tips on output filter design to minimize Total Harmonic Distortion (THD) and ensure clean audio.
   • Recommendations for simulation tools (e.g., LTspice, Multisim) and how to model Class D switching circuits accurately.
2. Learning Class TD in Parallel:
   • A beginner-friendly introduction to Class TD—how it differs from standard Class D, its advantages (e.g., efficiency, THD), and why it’s used in high-power audio.
   • Suggestions for resources (books, articles, videos, or tutorials) that explain Class TD design, focusing on practical steps for someone with Class AB experience.
   • Are there reference designs or ICs for Class TD amps I can study? How do Class TD designs adapt the power supply tracking to reduce losses?
   • Tips on how to start experimenting with Class TD concepts while working on my 500W Class D project—any small-scale circuits or simulations to try?
3. Minimizing THD for Both Topologies:
   • Techniques to achieve low THD in Class D and Class TD designs, especially for scaling to 5kW. What impacts distortion most (e.g., feedback loops, filter design, MOSFET choice)?
   • How Class TD’s tracking power supply helps reduce THD compared to Class D—any design tips to leverage this?
   • Testing methods to measure and optimize THD using my oscilloscope and signal generator, particularly for high-power loads like 21" and 18" speakers.
4. Learning Resources:
   • Recommendations for books, courses, or tutorials on Class D and Class TD design, ideally suited for someone transitioning from Class AB.
   • Resources that break down PWM, switching topologies, and tracking power supplies in a hands-on way.
5. Transitioning and Scaling:
   • Key differences between Class AB, Class D, and Class TD to understand (e.g., EMI, thermal management, efficiency).
   • Common pitfalls in Class D and Class TD design and how to avoid them as a beginner.
   • Design choices for the 500W Class D amp that will make scaling to a 3-5kW Class D or Class TD amp easier (e.g., power supply flexibility, thermal planning).
   • Advice on bridging/paralleling amps for 5kW output while maintaining low THD.

If you’ve designed Class D or Class TD amps, I’d love to hear about your journey! For the 500W Class D, what components or approaches worked best? For Class TD, how did you wrap your head around the tracking power supply concept? Are there specific MOSFETs, gate drivers, or filter designs you recommend for either topology? Also, any tips for managing EMI in outdoor speaker setups would be awesome.

I’m excited to learn both Class D and Class TD hands-on, starting with the 500W Class D build while studying Class TD theory and simulations. My focus is on low THD for pristine audio at high power. I plan to share my progress here and would deeply appreciate any guidance, resources, or encouragement to stay on track.

I've been looking at getting a new set of speakers for my desktop for sometime now, I've been using a pair of KEF Q15s for years but they've always been pretty underwhelming. After looking around at a bunch of options I decided to pull the trigger on the NX Bravo kit from GR Research, I've always wanted to build a set of speakers but starting from scratch is currently out of my wheelhouse.

There doesn't seem to be a ton of information around the Internet on these so I figured I'd create a thread detailing my build of this set of speakers. I purchased the kit late last week and after some international shipping paperwork issues it arrived at my shop in Ontario Canada today in 2 well packed boxes.

The kit includes pretty much everything you will need to assemble the speakers including solder, shrink tube, hook up wire and 3 giant sheets of no rez. I didn't buy the flat pack as I opted to design my own enclosures and because of cost, I already spent enough money.


The drivers look pretty nice, the woofers are small polymer basket drivers that look to have pretty low excursion l but since these are for small rooms and desks they should be fine. The planar magnetic tweeters are surprisingly heavy and seem to have fur coming out of them, must be some kind of damping. Maybe they could be wrapped in some very thin grill cloth to hide the fuz?



Included are a pair of 3D printed waveguides for the tweeters and rear cups to seal the back of the tweeters. While this is a cool idea the 3D print quality on the back side is not great. I have a 3D printer and can definitely do better, GR Research was kind enough to provide me with the waveguide file so I can make my own out of different materials other than PLA.


The ports provided are unfortunately not very nice at all and really look like plumbing parts. They have hard edges everywhere and the diameter actually tapers down towards the port entrance, since they have 3D printing capabilities in house why they don't print their own ports is beyond me. I won't be using these since I can do so much better.


They certainly didn't cheap out on the crossover components though, they provided air core inductors and huge poly caps for everything. Included are the tube connectors Danny goes on and on about on his YouTube channel but these are really just banana plugs 🍌, I don't care how you sugar coat it. One thing that I do find kind of annoying is that they do not provide any mounting board for the crossover components and simply give you a schematic and a photo. For a lot costing as much as this a simple layout board would have been appreciated.


My enclosures will be 3D printed in sections and bonded together, the walls are hollow so they can be filled with epoxy. The only wood will be a 3/4" thick maple baffle.

In an old active three way loudspeaker of a friend one of the TDA7294 has firmly blown a hole in itself. No apparent cause. After cleaning the board from black dust, the surrounding standard components on the standard pcb layout measure fine. The mute is not used. The loudspeaker unit of this mid channel measures ok.

After replacing the TDA7294, an offset of 0,4Vdc appears (input shorted). It can be traced back to the IN- by measuring the voltage at that input in relation to ground.
Without power supply R3 = 22k measures fine.
The other two TDA7294 on the pcb of this loudspeaker measure fine, hardly any offset at the output.
The voltages of the power supply are a stable + / - 25Vdc. The scope shows no sign of oscillation.

What may cause the offset of 0.4V ?

I wonder how many others do this. I've been hemming and hawing about my next speaker project. $5k for bass cabinets and 3-way plate amps. That's a lot of money for me. Then along comes a Stereophile review of a small-ish 3-way powered speaker that's $30K per pair and now $5k is a bargain.

Does this happen to you?

ALL MOSFET Amplifier.
THD at 1 Watt is 0.0008%
Harmonics shows most H2 and a bit H3.
Output should be at 17.9V and the bias is 1.20A.
Max output is like 18 Watt.
Tranformer should be 2x15VAC or 30VAC for like 40VDC supply.
Enjoy!

What do you think of this offer? One of the best drivers. I'm in the mood to try a couple. That would have been fun.

What do the experts and those who have heard them say?

Order the Markaudio MAOP-10.2 White 6" woofer - SoundImports

Boutique Haut-parleurs & Audio DIY

MAOP 10 or 11?

Suggestions for boxes?

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. 😉

Hi,

I am selling this one, Benz Ruby 2 Open Air. https://www.ebay.com/itm/335650663431

Problem is that my cart is completely worn out. Complete rebuilt at Benz (stylus, cantilever and coils new, only body and magnets remain) is between 400 and 600 €. That rebuilt can also take months which is pain, but result should be brand new cart for about 1000 € , normally that stuff new goes for over 2000.

Than again, I never rebuilt cart at Benz Micro, they as matter of fact officially discontinued service, I found way to do it by calling them. In any case , I have no idea if their rebuilt service is reliable ....

D14 diode has shorted due to failed output fets. Need to find some replacements and am not sure what to order. Any help is greatly appreciated, thanks!

So 2” dome mids…….fabric……sound amazing but limited in their lower end response as it relates to power handling……..MAYBE 700hz on a good day with transient peaks. How would I get around it?…….well two of course! Now the problem becomes the C2C spacing……BUT is it really a problem up around 5khz with such wide dispersion driver?………..could a 19mm dome pull off the transition?……the SB Acoustics SB19 certainly has the power handling on paper

https://www.parts-express.com/Dayto...Fabric-Dome-Midrange-8-Ohm-285-022?quantity=1
the smooth response on these is uncanny

https://www.madisoundspeakerstore.c...oustics-sb19st-c000-4-3/4-dome-tweeter-4-ohm/
and just as smooth. I’ve used this tweet before and it’s a hidden gem……just as good as an OM1 IMO

The faceplates of the mids would need to be truncated to get them close as would the face of the tweeter……or mount the tweeter flange over the faceplates of the mids…..closer time alignment sacrificing some diffraction.

I bought two big horns this type, HL-1018:


46.3*24.4*21.4cm Wide*High*Deep*

and some real drivers Kenford comp50 with Titan diaphragm.
Not tried until now the classic driver for the horn but bought just for fun a Piezo screw on drivers GT-1188.

And did measure only a response down to 4khz like with the small tweeters like KSN1005 what left me puzzled


So did the real Motorola KSN1188 ever really reach down to 800 hz as pretended ? (did not find a measurement online of it).

And is the GT-1188 as a spare part (with internal step up coil) only not working on this big HL1018 horn?

I know that the piezos can never be as good as real coil drivers but is it possible that the frequency response is so far away from covering the claimed response down to under 1khz?

A good question to pique interest......
Thoughts?

HD

Voice coils, with their endless crossover problems, must be superior to piezo-driven tweeters, since most, if not all of you use them. Is their non-linearity and "scratchy high frequencies" their only downfalls?

Apart from not needing a xover, are there any advantages to using piezo tweeters? How is development of piezo's going? Are they getting better? Will they ever reach close to a par with voice coils in tweeter applications?

My first gainclone build was built on BrianGT's compact (2.9×1.2inch, 75x30mm) boards. While these days I like larger boards for my builds, I've always liked the challenge of fitting better performance in this small form factor.

So here it is again, this time, with a balanced input and even lower distortion and better sound, using "normal" size (i.e. D=2.5mm L=7mm) axial resistors instead of compact ones.

I am posting some pictures for now and will update this thread in the coming days.

Schematic:


THD 1kHz 1W 8ohm (0.0010%):


THD 1kHz 40W 8ohm (0.0011%):


THD 1kHz 70W 4ohm (0.0029%):


IMD 18+19kHz 1:1 45W 8ohm:


Clip 1kHz 4ohm with +/-28V power supply rails:


CMRR (1% resistors, balanced source):

4A - ULN regulator LT3042 based, with synchronous rectifier - evotronix.eu

Zeno MKII - ultra low noise regulator based on LT3042 and powered by synchronous rectifier - Saligny LC inside 🙂
This is perfect regulator for RPi projects, Shigaclone, digital section in any DAC, etc.

Specs:
- input min.6Vac up to 14Vac, max.16Vac
- output any voltage up to 15Vdc
- max output sustained current 4A

Output voltage can be changed by R7 value.
R7 = Vout x 10Kohm


PCB + Saligny LC = 30euro shipped by tracked mail envelope 8euro.
Full mounted = 65euro shipped by tracked small parcel 22euro EU, 38 euro worldwide.
For EU citizens, 19% VAT is not included and will be added at the end of your invoice.
For EU VAT registered companies, there is no VAT.
For outside EU citizens there is no VAT.



Invoiced payment by PayPal - fee ~4%, or bank transfer.

Regards,
Tibi

Hi. I'm Desmond from Singapore. I call myself an audiophile but i'm pretty sure i'm just another amateur in the hobby. I have engineering backgrounds and as always, College/University only teaches that much of theory and not much of practical.

My aim is to design my own amplifiers and might even step up to things like DSP/EQ/DAC with guidance and support 😀. But first off, I want to design a portable audio amplifier. Please help me!

99% of the time, problems when building the amp come from soldering issues and installing components in the wrong polarity.
The following is a common example. The amp was fixed by Mr. Fujita just today.
Here is my suggestion. Please follow the manual step BY step.
When you finish the main PCB. Please cross reference to the floor plan from the manual


C116 and Q102 are installed incorrectly.



In addition, some soldering points are not covered by solder.
Correct polarity....








ReD: after repair
Pencil : before repair

(Update) all sold. Thanks everyone. I don’t know how to change the tile so am reposting this add.
🌟The Buyer pays for shipping unless otherwise on the Audio products being sold.


Pending/(update) 4-13-2025. So Am willing to trade for a pair of Fostex FE206En drivers and maybe some cash. A. Mark Audio’s MOAP 11 Full Range Drivers brand new on the box never used.
$347.90. my price on these drivers. That’s 50% off with the tax I paid which is $647.90 for both drivers . No low ball offers please. Buyer pays for shipping. E mail for offers and speaker pictures of the fostex FE206En drivers.

Sold/B. Mark Audio’s Puliva Seven HD in copper color. Asking $50.00 A pair. Buyer pays Shipping.

Sold/ C. Mark Audios CHN-50P Paper drivers. Brand new never used in the box. Am asking $20.00. Buyer pays for Shipping.

Fostex Full Range Drivers.

Sold /A. Fostex FE126NV drivers Brand new in box.
Asking $95.00 a pair. Buy for pays Shipping.

Sold/B. Fostex FE126E Drivers Brand new in box. Asking $85.00 a A pair. Buyer pays for shipping.

Dayton Audio.

Sold/ A. RS125 5” inch drivers. Am asking $45.00 a pair. Buyer Pays for shipping.

Completes speaker boxes for sale.

sold /1. Frugal Horn XL for Sale am asking $100.00 for the pair. Offers are welcome if wanting to e mail me? No low ball offers please.

If you have any questions please ask? My E- Mail is;

Jmboo1922@gmail.com.

I no longer have the following Items which have been all sold.

1. Fostex FF165K Enable drivers.

2. Vifa /Peerless Ring tweeters.

3. Paper tweeters (buy out at Parts Express).

My purchased-new Ah! Njoe Tjoeb 4000 has suddenly gone inop (hasn't been used for a few months, in a cool, dry-enough enivorment, but has always worked perfeclty fine up through the last time I ran it late last year.). It powers up, spins, plays, but nothing comes out from the analogue output (a very faint hum when the pre amp is cranked). I have not tried the digital out yet as I have the upsampler board and reportedly there is no digital out with that board present.

I will try to remove the upsampler board (matter of putting in DAC(?) chips where 1 is different from the other 2 (all 3 8 pin) so I will have to figure out which goes where., and see if it works with digital out.

The fuse mentioned in other forums/threads has continuity and the power relay is working. I don't have any spare 6DJ8 tubes handy but both are warming up and I would imagine the likelihood that both suddenly blew is unlikely.

None of the myriad of caps are bulging. The previous 99 CD was supposed to have leaking caps.

Any hints, or a service manual? Ah! appears to have gone out of business years ago.

TIA

Hello.
This is my modified A75 amp - maybe just an A40 from Nelson Pass. It worked the first time it was turned on and has been working great for 2 years.
The chassis is still not made, but it still sounds good.

I wasn't quite sure if this should go in Digital source or PC based so I'll put it here.

I was interested to hear from anyone who has used Tidal or Qobuz on a high quality system (eg not earbuds, car etc) and what they thought of the sound quality, music catalogue etc.

First a confession. 😱 Like many of you I build and tweak speakers and take sound quality very seriously. For many years I used to "sample" music online and anything I liked enough I bought on Redbook CD...

Then when iTunes switched from 128kbit DRM AAC to 256kbit non-DRM AAC I got a bit lazy and started buying albums from iTunes knowing full well that they didn't sound quite as good as an actual CD...but they sounded "close enough" for most music, and could still sound pretty good on a good system, and convenience was the driving factor.

Then a number of years after that I first tried Spotify, lured in by the low monthly cost and more or less unlimited access to browse and instantly play nearly any music and got hooked on the easy discovery of new music. I figured that anything I really liked I would buy the CD's for, but I never did! 😱

Some of you may know that Spotify is alone in using Ogg Vorbis with other services using either MP3 or AAC. Personally, I don't like the sound signature of Ogg Vorbis. Of the three I would rate AAC as the best with Ogg Vorbis considerably worse and MP3 marginally worse than Ogg Vorbis.

My music collection has been a mixture of CD's, CD's ripped to Flac, CD's ripped to AAC, older "acquired" MP3 rips, and Spotify, and after moving country once and house twice since then I no longer even own a CD player, despite keeping my CD's for possible future ripping to FLAC.

My main digital music playback system is now a Raspberry Pi 4 running Volumio with a Behringer UMC204HD USB DAC.

I was recently going through some old CD's and decided to try playing one for the first time in years - the only thing I have that will even take an optical disc now is an Xbox one! This goes to the TV via HDMI and the headphone analogue output of the TV then goes to the amplifier. Hardly an ideal signal path - fine for watching TV Shows and Movies but not what I would want to use for Music.

Surprisingly or perhaps not surprisingly despite this tortuous signal path there was no doubt that the CD played via the Xbox one and TV to the amplifier sounded better than the Spotify version of the album through a much better quality DAC. Much better. I tried many CD's and the result was the same - I had forgotten how good CD's sounded compared to lossy codecs after years of conditioning (brainwashing ?) by listening to lossy audio... 😱

I decided enough was enough and that it was time to research music streaming services that offer lossless audio, which didn't exist back when I first joined Spotify many years ago but now did exist.

So I came across Qobuz and Tidal who seem to be the two main players right up until the very recent announcement of Amazon Music HD. (More on that later)

As it happens, Volumio with the MyVolumio subscription supports full quality playback of both Tidal and Qobuz out of the box, (as well as playing my local FLAC, AAC, MP3 content and Spotify as well) making it very easy to do direct comparisons of the different sources using the same software and same DAC.

So I signed up for a free month trial of Qobuz which runs out in about 10 days and recently signed up for a free month trial of Tidal as well so I can do a direct comparison within the trial periods. So far I have listened more to Qobuz than Tidal since that trial runs out first but here are my thoughts so far on sound quality:

In short Qobuz sounds fantastic. 😀 Although you can choose to play MP3 versions of songs if you want (for example over 3G/4G on your phone when using ear buds to reduce data usage) all but a very very few tracks have at minimum CD quality FLAC at 44.1Khz 16 bit available.

Those songs that are CD quality literally do sound indistinguishable to my actual physical CD's that I had ripped to FLAC previously, and far superior to Spotify's lossy Ogg Vorbis versions of the same albums.

I have no doubt that they are using perfect FLAC rips of CD masters, so sound every bit as good as buying a CD, ripping it to FLAC yourself and storing it on a local media server.

A surprisingly large number of albums, both old stuff that has been re-digitised and current releases are also available in "Hi-Res", with Hi-res meaning 24 bit and/or >44.1Khz, and they show up with a separate badge in the listings, and when played in Volumio the sample rate and bit depth is indicated.

There seem to be a wide variety of sample rate in use depending on the specific album.

I've seen 44.1Khz 24 bit, 48Khz 24 bit, 88.2Khz 24 bit, 96Khz 24 bit and 192Khz 24 bit. My Behringer DAC supports all bitrates and bit depths natively without any resampling.

I'm now in my mid 40's and last time I checked my hearing only goes up to about 16Khz instead of the 18Khz it did in my 20's.... traditionally I've been a bit sceptical about "Hi-Res" formats as the higher sample rates in particular seem to be wasted if 44.1Khz will already work up to 22.05Khz and I can only hear up to 16Khz...

So do they sound any better than CD quality ? Despite my initial scepticism I would say yes, some of their Hi-Res tracks do sound slightly better than the CD versions. It's hard to describe subjectively (I'm more of an objectivist) but there is something about the smoothness and naturalness of the treble and separation of instruments and sound field that is undeniable. It sounds slightly cleaner and more spacious but at the same time effortless without any over emphasis of the high frequencies. Sorry but that's the best description I can give it.

The difference is not huge but it was noticeable with the right playback system and on many but by no means all albums. If I had to come up with a scientific explanation I can think of two possibilities, given that I don't believe in sound frequencies higher than our hearing limit having any impact on what we perceive.

One is that sampling at only 44.1Khz means your anti-aliasing and reconstruction filters have to cut off before 22.05Khz, and if you are trying to include all frequencies up to say 20Khz without attenuation, that's is a very steep filter needed to do that. That filter is inevitably going to have either phaseshift or time domain effects within the audible spectrum which may be audible. And if the filters are not done properly musical content above 22.05Khz may be folded back into the audible spectrum and this is likely to be audible as well as aliasing which may add something artificial to the top end.

Sampling at 96Khz means you can use much shallower more benign filters and also place their cutoff frequency WAY above the audible spectrum so that the filter skirts don't impinge on the audible spectrum at all. Having said that I think going to 192Khz is a complete waste of time over 96Khz, and I did not hear any difference between them, nor was I expecting to.

Something else to consider when comparing 44.1Khz and 96Khz even on the same DAC is that the DAC itself switches to different reconstruction filters for each sampling rate, and that may have an impact on the sound.

On my Behringer which I also use with ARTA for speaker measurement, at 192Khz it is of course ruler flat well beyond human hearing as measured in loopback, however at 44.1Khz there is a small rolloff in the upper treble starting about 15Khz before it drops rapidly at 20Khz. Would this "premature" droop be audible ? Possibly.

Possibility two for why the Hi-Res versions might sound slightly better is simply that the "Hi-Res" masters are probably sometimes different masterings of the original recording, and there's no easy way to know this.

I've read about some albums which when released in "Hi-Res" have had less dynamic range compression applied during mixing, and this of course would likely make it sound better especially if the original release was made during the loudness wars. So in that case the Hi-Res version of the song definitely sounds better, but not due to its sample rate or bit depth but simply because it was mastered better, and a version of that downsampled to CD quality would likewise sound better as well if it had been available.

But if all that is available is the original CD mastering with more dynamic range compression and the Hi-Res remastered version that has less dynamic range compression, then I say listen to and enjoy the Hi-Res version, just don't necessarily assume that it sounds better due to higher bitrate or sample depth!

I noticed these differences in mastering in for example a Moody Blues album where there are at least 3 different versions of the same album on Qobuz. One of the CD quality versions sounded just like the original LP I remember from years ago with a slightly strident and forward midrange on the vocals and a little bit of edginess on strings.

The Hi-Res version sounded better tonally balanced and slightly more laid back and easier on the ears especially in the midrange - clear but clean and much easier to listen to. Clearly the remastering engineer has gone in and applied different EQ to the individual channels and rebalanced the mix and in my opinion it sounds much better and more like a (good) modern recording.

Anyway, after two weeks I'm super impressed with the sound quality of Qobuz, especially to a full size stereo system with a good quality DAC that can output the "Hi-Res" tracks at their native sample rates. The CD quality tracks sound exactly like FLAC rips and there are a surprising number of albums in Hi-Res formats which all sounded at least as good or slightly better than the CD quality versions. I'm finding myself enjoying the music much more again and have spent quite a lot of time listening during the trial period.


Next up is Tidal. They also have CD quality 44.1Khz 16bit FLAC available for most songs, however I have come across a few albums that are only available in MP3.

From what I have listened to so far Tidal's CD quality FLAC sounds identical to Qobuz's CD quality FLAC, perhaps not surprisingly, except in cases where a different master (remastered vs original etc) has obviously been used, and as the labelling of some of the releases between the two services can be a bit inconsistent you can sometimes end up listening to a different release of the album especially with albums that have been remastered more than once.

Where they differ is that for "Hi-Res" Tidal uses something called MQA, and that's where a can of worms starts to open up. 🙄

Instead of the open, lossless FLAC codec used by Qobuz for Hi-Res, MQA is a weird Frankenstein proprietary codec that is partly lossless and partly lossy. For the full gory details of what I think is wrong with it and why I don't like it on a conceptual basis, I direct you to the following article:

MQA: A Review of controversies, concerns, and cautions - Reviews - Audiophile Style

To boil it down to it's simplest, it has a lossless PCM base encoded in the most significant bits at 44.1Khz or 48Khz onto which additional "detail" for higher (than 20Khz) frequencies is encoded using a lossy codec whose data is encrypted and encoded into the least significant bits. Figure 7A of that article explains it.

To play an MQA file at its full quality you need a hardware DAC that can "unfold" the file and process it back into its original state.

If you play it on a regular player then this additional "detail" is not retrieved. Because the least significant bits of each sample are effectively pseudo random noise (encrypted data looks like noise) that undecoded data would manifest as a slightly higher than normal noise floor - the article suggests that an MQA file played without a decoder would be equivalent to about 44.1Khz 13 bit...so not quite as good as Redbook CD without a decoder.

When fully decoded the lossy part of the data is combined with the PCM part to increase the bit depth and sample rate to 96Khz 24bit, but remembering that frequencies above 20Khz and bits below 13 bits have been processed through a lossy codec.

In short its a clever way to create a hybrid lossless/lossy codec that can "play" on anything but only play at full quality with the right MQA licensed hardware. Supposedly the file size is around 1/2 to 1/4 of FLAC at the same sample rates.

My biggest issue with MQA is simply that I want to move away from lossy codecs back to lossless, so why would I want to jump onto a weird hybrid proprietary codec that needs special hardware support (which I don't have) for full performance and probably sounds worse than Redbook without suitable hardware... ?

Five years ago when bandwidth was less than today I could see the point but today there is enough bandwidth to simply stream FLAC at high bitrates - and Qobuz proves that. So I think MQA is a solution looking for a problem that no longer exists, and in the process throws the baby (lossless audio) out with the bathwater!

I've played a number of Tidal "masters" as they call them with MQA, however I don't have a DAC that can decode them, so my only comment is that they sound "fine" but don't sound any better than the CD quality version of the tracks. So in my situation if I was to subscribe to Tidal I would stick to the CD quality FLAC streams as I'm not interested in buying an MQA compatible DAC when I don't believe in the philosophy or rationale behind it. If there is enough bandwidth available, just stick to FLAC in my opinion.

In terms of numbers of "Hi-Res" tracks, Qobuz seems to have far more Hi-Res FLAC tracks available than Tidal has MQA masters.

In the next post I'll briefly compare catalogue, iOS/desktop apps and pricing and then my conclusions, as well as a quick mention of Amazon Music HD.

New member trying to maintain old equipment and save money.

For source,

Oppo BD-105
PCs running Asus Xonar card (2 flavours of connectors, PCIe and the other one_
The main one of interest and why I joined, Ah! Njoe Tjoeb 4000 which has suddenly gone inop

Also have this Audo Nemesis VLE-1 LE DAC which I will have to use now as the Ah! is not working (will try to get it to work as a transport, no digital out due to the upsampler board).

For pre amp
Counterpoint SA 7 (.1?) updated by Counterpoint
PSVane T-417 would like a pair of monoblocks to go with it


Power amp
2 Alta Vista NP-100s which were COunterpoint SA-12s in an earlier life

Magnepan II, IIA. The IIs need regluing but question whether it is worth the effort.

Oh yeah, also an Elipson t/t

I'm assembling a parts list to restore an integrated amplifier from the 1960's, I haven't tested the preamp & driver transistors yet but I would like to have backups on hand to avoid starting and stopping in the event one is bad.

With the exception of the outputs, all others will need to be substitutes since the originals are now obsolete.

The outputs are 2N3055 and have a fT of .8 MHz (min) to 6.0 MHz (max). Do the replacement transistors in the preamplifier and driver stages need to fall in this range?

I’ve been a member for probably 20 years but have never posted until now

I am about to finish my first speaker build (conversion) and would like to be sure of the integrity of my cabinet. Lack of leakage, specifically.

Is there a simple (or any) way of checking for that, at DIY level?

Do you have a favorite driver that you would like to use in a mass loaded transmission line (MLTL) like a TABAQ, Pensil, or Metronome, but no one has yet run a simulation for it with MJK's software? Sometimes, folks on this forum can be very generous with their time and provide you a simulation if you ask. I have seen Bjohanessen do this many times for me and others. But in case you do not have the fortune of a full-blown simulation in MJK, but would like to hear what your driver sounds like in a MLTL, this thread is for you.

Let me first say that, this is not meant to replace a real simulation, but is rather an observation I have noticed that has worked a few times. I imagine that there are drivers where this will fail. For MLTL's where you want to push the tuning frequency below the natural free space frequency of the driver, choose a driver with a moderately high Qts (> 0.5).

This idea arose out of something I noticed happening on several occasions with a good result. I call it the 'accidental MLTL' technique because it may have been luck or chance but it seems to work a few times now for me (and at least once by others). I originally posted this in Bjohanessens's TABAQ thread but figured it was enough to spin off a new thread. I will repeat that post here...

If you happen to have MJK's MLTL worksheet, give this a try by running a simulation with your driver and running WinISD and calculating the CSA, length, and vent and compare the two. It would be interesting to hear your results.

----

Use WinISD bass reflex software (free) and plug in your driver's T/S params and design a vented bass reflex speaker enclosure using the default optimal case. That will give you the volume of the box and the vent cross sectional area and length.

Now choose a maximum length you are willing to have based on practical size constraints, typically 30 inches to 40 inches long. This will correspond to a quarter wave length that is probably higher than the tuning freq of the bass reflex design that came out of WinISD (circa 60 to 75 Hz).

Calculate the cross sectional area (csa) of the transmission line using volume from WinISD and the length you set. If the csa is bigger than you prefer, you can go off the optimal case by going back to WinISD and adjusting the box volume manually to a smaller value and tweaking the frequency even. You can play with the vent to get it to a diam and length you like. Typically, I do this because the default vent is not very practical. Then use new box volume and vent dimensions to calculate the new csa based in length of the line again.

Now build the enclosure with the csa and length and put a vent with dimensions from WinISD at the distal end. On the closed end, measure 1/3 of distance down and make cutout for driver and mount it there. Put polyfill stuffing from closed end to about half to 2/3 of the way down the line. Adjust this to taste, less stuffing gets more boomy bass. More stuffing gets tighter bass at expense of amplitude. I have found that the position of the driver need not be exactly at 2/3 and indeed can even be at the closed end.

Doing this approach will guarantee that at worst, the speaker will perform as a bass reflex optimized for volume and vent size. But if physics of a MLTL and quarter wave theory kick in, you will end up with a speaker that has deeper bass extension than predicted by WinISD. The speakers that I have built following this recipe have measured very well when I look at what is coming from the bass port - typically, I get the low frequency extended or 'pulled down' by an additional 15 Hz from the plain bass reflex prediction. The speakers also sound great - very balanced when the bass port output integrates with the direct radiation from the driver.

If you want to experiment without investing too much in wood, use foam core to build quick and dirty speaker. Even cardboard can be used as a test speaker for this.

----

Here are the two examples that have worked for me:

http://www.diyaudio.com/forums/full-range/223313-foam-core-board-speaker-enclosures-148.html



http://www.diyaudio.com/forums/full-range/223313-foam-core-board-speaker-enclosures-109.html



A third example not by me might be Cogitech's "nano tower" speaker with the ubiquitous TB W3-881si. Which I think was also designed as a bass reflex with WinISD but due to its slender aspect ratio and length, happens to have some very nice bass deeper than one would expect based on accounts from builders.

http://www.diyaudio.com/forums/full-range/200912-nanotower-tang-band-w3-881si.html




Update : 4th example is a quad driver bipole with W3-881si http://www.diyaudio.com/forums/full-range/234535-tangband-w3-881-mltl-build.html



Fifth example is a folded AMLTL with a Vifa as wall or bookshelf speaker:
http://www.diyaudio.com/forums/full-range/231951-accidental-mltl-technique-16.html#post3466655

If anyone has these transistors and could offer a matching quad with IDSS 10 to 15mA, ...write to me PM

Payment PayPal

Sold.

Up for sale is an Iron Pre Single-Ended Essentials kit. This is the early-release version as shown in the photo. The parts bag has been verified complete, and the kit includes two Cinemag CMOQ-4HPC, main board and a twister board. $160 USD + shipping. I prefer to use Paypal F&F since we are all friends here at diyaudio. Will ship double boxed. US/Canada only, please. Thanks!

This is a builders thread for the "long and skinny" J2 clone boards.

Please post hints, tips, discoveries, etc. And of course pictures. 🙂

I picked up this amp (Alpine T707)and all of the outputs are missing. Does anyone have this information? Or a service manual. Any help would be greatly appreciated.

For sale a pair of tweeters
220 euro plus shipping.
i am in greeece so europe
would be easie for shipping.

I need a new preamp, and I want to to build a DIY preamp like Mike Freda did. Have collected parts like output transformers from Edcor, tubes...Trying to buy the HLMP-6000 LED, for the ccs, but it is difficult to obtain. Are there any equivalents to the HLMP-6000? Data for the LED are 1,6v and 10 mA. Kan I use any other LED, but with the same data?

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I would like to explain my novel compensation I used lately in all may amps, CFA or VFA with excellent result.
This is a kind of TPC (Two Pole Compensation) but in my opinion much improved with excellent bandwidth Phase Margin (PM) and Gain Margin (GM).
To start with is simple TPC. Choose components give good result but not good enough.


Second step is improved TPC by adding a capacitor parallel to VAS emitter resistor. With the simulation I selected the capacitor value to get good LG (Loop Gain) plot.
Both PM and GM are improved.


Third set is splitting input leg capacitor and connect half in input leg coming from VAS and half directly from output. By simulation choose values could be different to get best compensation. PM and GM increased enormously.


Fourth step is to get it even better decreasing TPC resistor (R1), ULGF decreased to acceptable value.

I hope this can help if someone decided to use this kind of amp compensation.
Intentionally I did not use any math, as TPC calculation was showed before even in this forum. Sorry for my simple common language.
Damir

This is very simple amp, enhanced VAS was not used, with more elaborated amp the result is even better (higher Loop Gain an so on).

As suggested by Zen Mod, I have opened a new thread for discussing logic solutions to replace the original rotary switch and the plain volume pot.

Whether you want to control the relays remotely (Lazy Boyz) or from the front panel using Arduino, ATtiny, ESP32, or any other MCU, this is where to discuss it and ask questions.

Attenuators, whether it's remotely controlled or not, can also be discussed here. In fact, any deviation from the standard rotary input channel switch and/or the standard pot qualifies for discussion here.


Links to important information

The original Iron Pre threads:
https://www.diyaudio.com/community/threads/iron-pre-essentials-kits-for-the-diya-store-register-your-interest.390509
https://www.diyaudio.com/community/threads/whats-wrong-with-the-kiss-boy.293169/

Updated schematics and BOM of the Iron Pre kits: post #1 of the Iron Pre Essentials Kits thread

Modification of current boards (first batch) to allow switching of the relays via microcontroller.
SE version: Post Post #13
BAL version: Post Post #14

There is some valuable information about driving the relays using logic pins on MCU's in the Iron Pre Essentials Kits thread, starting with post #1,232 and maybe even before that.

PS: This post will be updated regularly to include important information.

Sorry for a very long first post but I hope it gives all the relevant information needed without having to drip feed it.

TLDR; player won’t read CDs after being stored in attic for about 20 years; I changed all electrolytic capacitors, and resoldered most joints, on the power, decoder and servo boards. Player will now read TOC of a couple of discs sometimes but no others. It won’t play any tracks or discs.

A few months ago I decided to get my late 1980s B&O system from the attic where it had been stored for about 20 years and set it up in a spare room. Unfortunately the time spent in the attic has been detrimental to its health.

The CD player wouldn’t read disks properly giving a “?” on the display.

I’ve done a number of things to try to fix it based on the advice I’ve found on the internet.

I changed the electrolytic capacitors on the 5886 servo board with the recommended pack from BeoParts including the all-important blue axial one. After that it played for a couple of minutes but then powered down. I then resoldered the voltage regulator connections on the decoder board. Success I thought as it now played CDs. To check it out I left it on repeat play for a few hours just to make sure I’d fixed it and it worked fine. But, the next day it wouldn’t read disks at all again.

Since then I’ve;

changed the voltage regulators on the decoder board in case they were performing below spec

checked the caps on the power board. I did find a faulty one although that didn’t actually relate to CD playback. It’s one that’s controls the remote controlled standby system. But, I changed them all just in case.

changed all the caps on the decoder board. I first changed the ones relating to power despite none showing as faulty. I measured the DC voltage getting to the servo board from the decoder board and it is close to specification, but not perfect. The two feeds that should be + and - 10v are between 11v and 12v (taking into account if they’re + or -). The other power feeds should be -6v and +5v; I’m reading about + and - 6v on both. Do these numbers look OK?

Still no joy, so I checked and changed all the remaining capacitors, both radial and blue axial, on the decoder board. Am I right in thinking they relate to sound processing? All the Philips blue axial ones were out of spec, so not a wasted effort.

After all that, I now have a player that occasionally reads the TOC on a couple of CDs correctly as it shows number of tracks, CD total length, track length and lets me advance to individual tracks. But it won’t play an tracks. If it tries to play them it shuts down after a few seconds. Other CDs it just shuts down before it can read the TOC.

Finally, when I listen to it trying to play a CD I’m sure the speed the CD rotates at seems to change as though it’s hunting for the correct speed and then it shuts down. There is also a sound of something rubbing when the CD turns, but there is no evidence on a CD as to where it might be rubbing.

Apart from messing about with the laser power potentiometer, which I’m not keen to do as all I have is a standard multimeter, I’ve run out of ideas so am looking for any guidance you guys can give me. Do I need to test/replace the other components on the servo board; ceramic capacitors, diodes and transistors? Or, is it drive motor related? Or, maybe find a non B&O working player with a CDM 2/10 drive and try that in my B&O?

The top end on some drivers drives me nuts. Jordan JX92s/Eikona, 1st gen Alpair 10/6 i found quite fatiguing. The second gen Markaudio 7.3/10.2 were greatly improved, A10.2 still having a bit of edge. Alpair 10.3 got even closer to the A7.3.

Note that the Markaudio drivers were fully EnABLed.

This is about my annoyance with the top of the Alpair 11ms. And to a lessor extent the A7ms.

Taking into consideration the many complaints and measurements on the Alpair 10.3 of some significant issues up top. Something i did not hear in the A10.3eN (they did have a bit of edge that went away with many hours). @waxx @wchang

There have been comments about the edge on the A7/11ms by at least a few members.

So i thot it would be worthwhile to try the simple mod lifted from EnABL that was most likely the cause of the ringing. So i did the penultimate step of the EnABL treatment, 2 coats of acrylic gloss (diluted 50%) on the cone and one on the dustcap. Sort of analogous to a dirt cheap MAOP but instead of electoning the topfew molecules of the cone, we add a layer many molecules think on top. The purpose of what i have is to paint over decals on model planes/trains/cars.



it is difficult, but possible, to reverse the mod.

Bernie put them in these Compact Floorstander Mar-Ken11ms. These are high quality solid wood build like the Sibelius.



I had heard these stock (and also his Pensil/A11ms & FH3/A7ms builds). When they came back with these drivers installed it wasn’t very long before i thot i had something. A month+ in they have not driven me up the wall.

But i’d like to see if some members would be brave enuff to try it on their drivers. And some who have measured and added notches, does it obviate (or reduce the magnitude) of the notch.

A10.3, A7/11ms in particular but if is working, probably on things like the little Founteks too.

Not great light, but some alround shots.



Questions? Volunteers?

dave

@hobbers

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.....

One each AD1862 & AD1865 both with brand new N-J grade chips.

Both boards have header pins inserted for separate supply to op-amps. I recommend using dupont 2.54mm single pin female prototyping leads.

The DACs will need an I2S input module, either Amanero type or XMOS. It is worth trying the new York Pico by Eclipse as discussed in the vendors forum, this looks a very good design. For sale now on Tindie.

AD1862N-J completed DAC - SOLD

AD1865N-J + OPA604 op-amps completed DAC- £65 plus shipping

Payment with Paypal friends or add fees.



Also for sale are DAC boards with shift registers mounted, included are DAC and op-amp sockets. AD1865 board comes with tube IV PCBs and SOIC to DIP adapter board for mounting DAC chips

AD1862 DAC boards with shift registers - £10 each

AD1865 DAC board with shift registers plus tube IV PCB kit & tube holder - £15

Hello from Hannover/ Germany. Looking forward for interesting information regarding DIY Audio.

Hello everyone...greetings from Malaysia...My Name is Hafiz...just call me hafiz...i own front loading turntable Aiwa AP D50 and still searching info to restore it back to fully function...i hope i can gather all info and sharing here...best regards

Hey, my name is Raanan. I haven't built any speaker yet. I am new to this hobby and very exited about it. I am working on building a clone of the Orangutan O\93. I also want to upgrade my center speaker, and I am still wondering if it would be better to buy new SVS Ultra center or to build my own center.

I've restored an Aiwa AP-D50 turntable and have even sourced a replacement end sensor. Everything runs perfectly (the usual fault of the tray not ejecting unless pulled) but one very annoying fault that I haven't been able to fix is that the tonearm lifts and returns to the rest. Sometimes as soon as the stylus drops sometimes towards the end of the last track, sometimes halfway through the first track. Occasionally it will play through without happening. I've studied the manual and schematics. Can anyone offer advice?
Could S3 the load play/cut switch be ghost triggering? I've cleaned it with deoxit. If I turn the switch to manual the fault still occurs. I've spent a long time on this restoration and this final fault is driving me crazy.

On pins 4 and 5 of the ADAU1467 there is an input and output of SPDIF and I would like to use them in my own preamp project by installing Toslink or RCA or preferably both, could someone show a basic diagram of such an interface?? I've been looking everywhere and somehow I haven't found it anywhere :-(

bonjour quelqu’un peut m’expliquer comment fonctionne ce filtre bass 12 db mid 6 db high 6db

As the title says this is module will keep the output of your preamp shorted to gnd with the condition this output is ac coupled.

Remember for when used for turn on thump protection! if you power off and don`t wait that the relay disengages and power on again you will get the power on thump or if the network drops and suddenly comes back then turn on thump can be present. It is more a protection for our bad memory.. 🙂)

It can be used for turn on or turn off thump.
It can be used up to 36vdc(probably more) and when assembled should look like in the image below.



One module can be used for 2 channels in a stereo setup if same power supply powers both channels. If you have a dual mono setup then one module goes for each channel.

To use it is simple, connect IN to the output of your preamp that has the turn on/off thump, gnd goes to gnd of your pre and V+ goes to V+ of your pre.

I have attached the schematic, bom and gerbers.


Edit.
Thanks to JP for his suggestion! I made a r1 version that includes this diode and updated all the files.

Later edit: Because of the added diode the cap that does the turn on delay discharges more quickly and if you turn off and back on(or when network power is lost) most probably you won`t get the turn on thump. JP is an amp saver this time 🙂 so thanks again 🙂

I was digging through my stuff recently and I found a few Dayton Epique 5 1/2" drivers and wanted to do something fun with them. I was thinking of pairing them with some sort of bending wave speaker so it would have really big sound in a small package. Could I get a recommendation on speaker selection? It seems tectonic and Dayton are the only acts in town for BMRs, and I think the manger transducer is a bit outside my price range. I was also thinking of maybe an exciter on some sort of planar substrate, but I would imagine that would be slightly more engineering intensive to implement. This will most likely be an active configuration with DSP.

Assembled

Modulus-86 10th Anniversary Edition​

299USD$ or 400CAD$

hi all, here I am starting my journey into diy for improving my hi-fi system.
looking forward to engage on streamer/dac stuff and will be starting a new Ian Canada streamer/dac build soon.

Hello, I am a big fan of DIY audio, and am trying my best to build a great sound system for relatively cheap, with decent used equipment.

I have 2@ B&W 604 S2 speakers,
fed by 2@ NAD 218 THX amplifiers in mono-block mode/vertical bi-amped to the speakers' passive crossovers, 1 amp for each speaker,
fed by a Marantz SR7011 pre-amplifier in Stereo mode.

It was all working nicely, until my kid pulled out the Mini-RCA input cable from my iPod to the Marantz amp. That made the dreaded big POP noise; after that, both midrange tweeters are entirely dead. No scratching or humming, just 0 midrange. It sounds like the high tweeters and the low woofers are working, but at this point, I'd like to somehow test all woofers/tweeters on both speakers and each speaker's internal crossover, to make sure it's all working. Since the problem is on both speakers, I can't tell whether it's a 2-speaker problem, or my amp/s has a problem. I have basic electronics knowledge and am very willing to learn. I have documentation for each component, if anyone needs me to upload them.

Is it the speakers, the woofers, the crossovers? or both power amps? or the Marantz pre-amp? How do I test, how do I proceeed, what do I do, where do I start?
I have several NAD 214 amps, that I can use instead of the 218's, to test if it's the 218 power amps. My fear is that the POP killed something in the Marantz pre-amp.

For later:
-I have collected 6@ NAD 214 amplifiers, which I will assign 1 amp in mono-mode to each surround sound channel for a 5.2.4 Atmos setup. I have all the speakers, subs, amps, I just never set it all up. The 2@ NAD 218 THX will remain in mono for the front 2 channels. I know that is overdoing it. I wanted to make an all mono-amp setup for fun, for relatively cheap.
-I'm not sure if there's a way to make the SR7011 mono-amp its 9 amp channels into 1 amp channel full force; I was hoping to use 1 mono channel for Center channel, but I can't tell from reading the manual if there is a way to make it work in mono-mode at 125 watts RMS.
-Subwoofer is 2@ B&W ASW610XP. I quite like them.
-I have 5@ ButtKicker LFE for the 5 seats for watching movies, and 5@ old Crown amps to power 1 to each ButtKicker; they want 1,500 Watts RMS each.
-Just for fun, I no longer measure my Franken-stereo in watts, but in horsepower. As of today, we are at 9,675 watts RMS = 13HP. Huzzah! 3/4 of the total HP goes to the ButtKickers, so it doesn't really count towards the sound system.

For those of us that don't have access to an Audio Precision set, what are folks out there doing for measuring THD, IMD, SID, whatever? Anything clever for helping to set bias points or actually measuring impacts of changing ground connections, etc?

Here are some excerpts on the topic from another thread ( the Leach amp design thread, I beleive); comments welcome (hint, hint!):

From Damon:

...As for other designs and tweaks, I can only point to Doug Self
and Randy Sloan and their books on amplifier design for hints
as to what might be possible. But we're already into a range of
low distortion that my geriatric collection of Heathkit test
equipment can't possibly measure, and I can't afford an Audio
Precision test set.

Unless I can come up with a "cheap" Audio Precision equivalent,
I'd be shooting completely in the dark with new designs. And
that's daunting....

From me (mlloyd1):

..... I wrestle with the same problem as far as distortion measurements. I usually look at distortion waveforms on my scope. I think we need a group discussion about how to work around this issue for DIYers. There was a REALLY nice project in Audio magazine (RIP) some years ago for a very serious analyzer that could be reworked with current tech (using OPA604 op amps and current buffers instead of TL071, for example) and probably have much more performance than we would need. I've also seen some writings in past issues of The Audio Amateur (now Audio Electronics) by Erno Borbely ( I think) for a distortion analyzer. There was a project in Radio Electronics some years back also. I actually made a PCB and partly built this one. In my opinion, it worked very well - it gave consistent measurements with a Audio Precision I had access to at the time. However, it was VERY tough to control the electronical noise in my DIY environment though - halogen lights, misc dimmers, hair dryers, etc. (it's tough being a married DIYer!) don't make for a clean test environment. Finally, reading through the service manual for some of the HP distortion analyzers (I think the guy that did the Audio magazine project referred to doding this also) suggest a few ideas as well. Alas, I can't seem to find my copies of this anymore :-( I can't recommend enough times that service manuals from GOOD test equipment makers like HP and Tek are EXCELLENT sources of material for study!

I'd be happy with a distortion test box that could spot check with high resolution at about 4 frequencies: 50Hz, 1KHz, 20KHz, 75KHz.

Who's game?

Nelson, what do you do when you have something to test (say an idea at home late at night) and the Audio Precision is nowhere around? Wait until later? 🙂

Maybe we could even ask an analog Guru like Jim Williams at Linear Technology to design a simple, high peformance THD analyzer circuit (I single him out because I remember and oscillator circuit he designed that was claimed to have THD specs in the single digit parts per million. This is incredible!). They might already have such a design sitting around somewhere; I haven't checked their web site and app notes lately. Hmmmmm ....

from Grey:
.... Give the distortion analyzer project its own thread so people will be able to find it more easily.

I'm interested.

I'd like to reiterate--for those who haven't understood what I've said on these matters--that I'm not against low distortion, per se, just the use of massive quantities of negative feedback to achieve the distortion figures. Just don't pursue low distortion as an end in itself, as you'll usually find that once you reach a certain point sound quality suffers. But up to that point it can be a useful tool....

Hi. I am an electronic engineer and audio enthusiast. I love to squeeze more out of devices, modify and improve devices. I am constantly improving and trying to move with the times. Every first step can be the beginning of an exciting journey. Who knows what I can do tomorrow?

For sale a pair of scanspeak passive radiators.

They work well, but I going for a different approach. They are currently in use.

Price is 200 for the pair plus shipping costs

I am in Greece 🇬🇷

Happy to ship world wide

Up for sale an excellent high end signal processor from Helix. It is the Helix dsp PRO mk2 upgraded by the Amp Doctor english / Gordon Taylor (designers Genesis UK). Excellent sound quality and tons of parameters. The Helix dsp usb module is included and a programmable controller is also included (I use it for master volume and sub volume, also switch between two savefiles )

Price is 800 euros shipped within EU

I am starting a new project to assemble an amp that will use two LM1875 TO220 packages in a small, lightweight board. The large heatsink I have is more than adequate for thermal dissipation for the amp, but it is not drilled and I don't have the tools for drilling and tapping to be able to screw the LM1875s to the sinks.

I see thermally conductive adhesive glues and even adhesive tape advertised. If I supported the weight of the boards in the case with standoffs, would the thermal adhesives do the job of conducting heat effectively to the sinks?

thanks!

was fooled when purchasing this Yamaha CA 1000, the two internal metal covers are missing, I'm looking for help, thank you

Hi everyone, newbie here!

I play the Cavaquinho, a sibling of the Ukulele that uses steel strings and has a bright sound. We use piezo transducers to pick up the sound, normally two small discs underneath the instrument's top board.

It works well, but it has a high impedance, and we need to put a lot of gain in the mixer. So I would like to make an all-in-one pedal to make it better. The chain would be something like this:



The Digital Tuner is something I worry about later.

For the Preamp with tone controls, I thought about using one described by Rod Elliot at https://sound-au.com/project202.htm



Then I need to create the boost and DI box output. Can I use the OPA2134 to achieve this as well (I bought 5 of them)? I would like to power this either using a 9V battery or a rechargeable 18350 lithium battery.

Is my project feasible? Am I missing something?

Cheers!

I'm looking at fixing my Akai Am-A70, problem is the relay keeps going on on off randomly,

Power supply voltages seem to line up with the schematic

+/- 54vdc
+/- 16vdc
+ 12vdc

Offset at emitters on one channel under 10mv the other 100mv, no adjustments that I can see

Where to start from here?

Long time amateur speaker builder here, retired and having fun. My last project was these ported 3-ways, very successful and running right now. I promised a friend I'd build him a fairly compact system, so this will be my usual 2 way but with electronic crossover and bi-amped. Here's the component list:

PRV Audio 15W1000v2 15" Professional Woofer 8 Ohm
PRV Audio D260My-B 1" Mylar Horn Compression Driver 8 Ohm
JBL Selenium HC23-25 1" Exponential Horn 100x40
Extron XPA 1002 Power Amplifiers
Behringer SUPER-X PRO CX3400 High-Precision Stereo 2-Way/3-Way Crossover

I'm using three of the Extron amps, I found a very favorable review and decided to try them. I have one in stereo for the highs (60w/ch), and one each for the lows, bridged (200w/ch). These are Class D, but the tech seems to have reached a point where that is practical. Definitely a different sound than the tubes I prefer, but very clean.

I'll be using 3cu ft enclosures, undecided whether sealed or ported. Comments and calculations welcome!

Hello,

I would really like to build a 100+ watt Exicon ECX10N20 & ECX10P20 or Exicon ECW20N20 & ECW20P20 Lateral Mosfet based amplifier.

Existing examples that I can find,

->From the Build Audio Amps web site, Project #5,

Project #5 100W MOSFET Audio Power Amplifier

But is uses Hitachi 2SA872, 2SB647 and 2SD667 transistors that are no longer available

->Pee Cee Bee's V4 lateral mosfet amplifier,

PeeCeeBee V4 Rev2

But it only uses one pair of output transistors for 50 watts.

->SusyJ has her 50 and 100 watt version of the AEM6000 lateral mosfet amplifier,

Susy J 50/100 watt version of AEM6000 amplifier

But is uses the SST404 which I cannot find anymore. I would also prefer through hole to all SMD.

->Apex FX-12 lateral mosfet amplifier,

Apex FX-12 gerbers by Prasi

Only one pair for 50 watts.

(BTW, has anyone done a BOM for dummies for the APEX FX-12)

-> Elliott Sound Products Project 101

ESP Project 101

This one actually has most of what I am looking for, except Rod is half way around the world and having purchased from him in the past, post from Australia is ssllooww.... I would prefer to be able to have double sided boards made without jumper wires.

Is there a published two(+) pair lateral mosfet amplifier with an easily source-able parts list?

Don't necessarily need gerbers, a PDF of the circuit board would be fine.

Plan B will be ordering ESP Project 101 boards...

Thank you,

David.

Study of using Fourier transforms to implement digital servo control in XANADU based systems.

This is a test thread, please do not reply (even though you know you want to 😉)

This thread is for discussions about the diyAudio F5 Build Guide we have created for the First Watt F5 Amplifier by Nelson Pass. For more information on the boards available from the diyAudio Store, please see the information below.

Please note: The V2 and V3 boards have different parts placements.

• New V3 build guide here: Firstwatt F5 amplifier v3 - diyAudio Guides
• Old V2 build guide here: diyAudio F5 Build Guide
• Product Page
• Store support area thread for discussion the boards themselves specifically
• Schematic
• Bill of Materials
• Dimensions

Threads on diyAudio that relate to this product (If we have missed one, please post it in this thread and we will add it to the list):

• Build Thread: http://www.diyaudio.com/forums/pass-labs/121228-f5-power-amplifier.html
• Build Thread: http://www.diyaudio.com/forums/pass-labs/182723-how-build-f5.html
• Build Thread: http://www.diyaudio.com/forums/pass-labs/188691-illustrated-guide-building-f5.html
• Build Thread: http://www.diyaudio.com/forums/pass-labs/185154-initial-setting-f5-bias-help-please.html
• Listening Impressions: http://www.diyaudio.com/forums/pass-labs/155599-f5-listening-impressions-discussion.html
• Modifications: http://www.diyaudio.com/forums/pass...building-quartet-amps-search-ultimate-f5.html
• Modifications: http://www.diyaudio.com/forums/pass-labs/168040-f5-2sk2013-2sj313.html
• Modifications: http://www.diyaudio.com/forums/pass-labs/182966-f5-low-z-loads.html
• Review: 6moons.com - industry features: Swansong for an ugly duckling