Tiny tweeters used in the Wilson Audio Alexandria?

I've been looking around the higher end spectrum of larger scale, cost-no-object speaker systems available and had some serious questions to those who have known any owners of these systems. I'm trying to understand why someone would pay the 6 figure price for some of these "end game" speaker systems.

Here's the big rub for me -
How do these companies get away with building very large, (supposed) high output capable systems using 1" tweeters crossed fairly low?

There's no way a 1" dome can hang with that much band specific output capability without going up in smoke or being significantly choked with some means of a dynamic range restricting protection mechanism? Its just a huge imbalance of cone surface area while having the entire HF come from a small soft dome tweeter. Even with a WG, it would still be left in the dust with the other drivers having significantly more reserve in their dedicated bandwidth.

I find it quite shady the crossover frequencies aren't disclosed in a Stereophile review of the big Wilson Alexandria. I feel that any sain person wouldn't buy a speaker system which had all that LF capability, yet only used one or a second rear mounted soft dome - (guys who know me well enough have heard my views on rear mounted HF drivers).

I've heard the older version of this speaker in question and didn't feel it sounded like it was worth the extreme asking price. I feel like I'm expecting to find the highest caliber HF driver available on a speaker like this. I'd expect at least a high sensitivity 30mm textreme dome or even something like a beryllium T34B.
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Talia - Line preamp complete and updated project

This project is dedicated to people who think the line preamplifier is an important component of a HiFi chain.

Allow me a warm suggestion to those 5 members who have no respect for enthusiasts sharing their knowledge: please avoid reading this post. Time is precious. Do not waste yours.
Please do not share your “know-it-all” fake knowledge (zero distortion gain stage, Carnot engine kit, sunglasses, and other nonsense). Again: time is precious. Do not waste ours.

I posted some documents about this project in the past, in a nonexclusive thread. That wasn't very clear!
Since many of you have been reaching out to me via private messages, I wanted to share Talia's latest updated version with all of you by collecting all the documents in one place. Forgive me if you read something you already knew.

Forgive me also for the length of this post, it deals with the whole project. The mechanical section is to read only if you want to finalize the project with HiFi2000's Slim-Line 1U/350mm cabinet.

This update was made possible with contributions from some of you, for which I am grateful.

This line preamplifier was born as a running-in bench for the "Shared Gain Stage" (SGS) module I designed mainly for my RIAA preamp/Eq Vera, and I needed to verify and validate it. It would have been hard doing this job on the Vera, which uses 4 SGS modules, so I started Talia's project, which needs only 2.

The SGS module has its roots in an old project of mine, “WOTS Tre” (2008), a cascode of BF244B JFETs with a follower, created to please my friend Gianfranco, who appreciated the sound of my hybrid WOTS LeGrand (2004), but rejected working with tubes. The Tre turned out very well, sporting the same sound as its older brother. I had published it on my website when I had one, and it was built by several enthusiasts, who said they were happy with the realization.

Let's get to the core of the project.

ELECTRONIC SECTION

Objectives set for Talia:

  • simplicity and effectiveness: a single gain stage;
  • use of the SGS module configured for a “line” gain of about 20db;
  • dual-mono construction, only the mass is in common between left and right channels, and should not be separated;
  • slim aesthetics (19" 1U) because I like it, and it does not weigh down the area of the audio system;
  • possibility of deactivating the BJT of the CFP, leaving only the JFET cascode active;
  • no loop feedback, only local feedback;
  • stabilized power supply without feedback;
  • this point was an old obsession of mine: the possibility of replacing the SGS module with the HDP01 module by Eng. Maltese (Italian magazine Costruire HiFi #61), configuring the motherboard for a voltage of 24V;
  • expected sound: I will deal with it later.

Before proceeding further into the project, I would like to point out that, like the Conrad Johnson GAT, and other prestigious projects, the Talia is inverting. This means that to restore the original phase of the signal, provided your power amplifier is non-inverting, like most of them, the conductors of the speaker cables must be swapped. A non-inverting gain stage could be realized in various ways, which inevitably all lead to non-negligible circuit complications and lengthen the signal path.

Functional blocks:
  • SGSa or HDP01 (SGSb) module for signal amplification;
  • low-pass filter at the input, inserted in the volume potentiometer board;
  • Virtual Battery type power supply;
  • delay circuit that keeps the output short for about 120 seconds from power on;
  • input selector (optional) made with miniature relays, and a small power supply.
SGS module (schematic in picture 1, PCB in pictures 6 and 7).
I will not describe the tuning work on the SGS module; from the schematic, you can see a JFET cascode connected with a BJT in CFP configuration (developed by engineer George Clifford Sziklai in the 50s). The BJT increases the JFET cascode's transconductance and its linearity. In this case, we are not looking for a particularly high gain, but we want the amplification stage to be linear across the entire audio band, a fact guaranteed first by the cascode, and then improved by the BJT in CFP.
Which JFETs should I use for the SGS module intended for the Talia? Even if I started with the LSK489A dual JFET to set up the Vera, ending up with excellent results, when I came to set up the Talia, I preferred the LSK489 selection B (LSK489B), which allows an easy configuration for the gain we are interested in. Selection B has a dispersion of the characteristics that is decidedly less wide than selection A. I verified that there is a good chance of being able to match two LSK489Bs acceptably by purchasing only three.
The gain stage is followed by a source follower loaded by a CCS set to 10mA. The depletion MOSFETs seem to be made specifically for the CCS configuration, so I used an IXTP01N100D, which is expensive indeed. According to the opinion of most audiophiles present on various forums, it is one of the best devices for this use. I had some, so I also used them as followers with excellent results. I believe that the cheaper pairs STF3LN80K5/DN2540, and DN2540/DN2540, which do not show any lack of availability, can guarantee the same qualitative results. In other projects, I used the DN2540 with excellent results, but on the net, it is not much appreciated. Sometimes, it happens that a component gets a bad name just because someone has used it badly. A voltage of 60V keeps the JFETs safe, without particular precautions, keeps saturation away, and allows the gain stage to remain linear up to over 10Vrms output signal. Unluckly I lost the LTSpice simulations, and I don't know how to use it fully to replicate the same simulation conditions; I hate its CAD for drawing circuits, but having received the schematic .asc files from a gentle forumer, who is passionate about it, I was able to run several simulations and determine the values of the resistors to use.
A side note: bootstrapping the CFP seems a good idea, but it is not. I tried: the gain increases way too much, and the hiss too. The same behavior is expected if you load the CFP with a CCS.

HDP01 module (SGSb - schematic in picture 2, PCB in picture 8).
This is the project by Eng. Maltese presented on the Italian magazine Costruire HiFi #61, which I refer to for the complete discussion of the project. I copied it as is, without changing anything. I have always been attracted to its circuitry. I designed its PCB to be mechanically compatible with that of the SGS.

Low pass filter (schematic in picture 3 - motherboard PCB in picture 10). It prevents high-frequency disturbances from reaching the amplification circuits. It is a simple RC circuit housed in the same PCB that connects the hot side of the volume potentiometer to the center pole of the selector and the potentiometer slider to the input of the motherboard.

Virtual Battery power supply (schematic in picture 3, PCB in picture 9). It is the main power supply of the Talia, its circuitry has been known for a long time. I believe it comes from Technics; in my versions, there is always a CCS to drive the Zener diode, preceded by an RC filter. The voltage thus obtained is filtered and sent to a further RC filter, before driving the gate of the output MOSFET. Not being a supporter of feedback, and not requiring a lab instrument precision, my preference for solutions intended for HiFi has always gone to the Virtual Battery circuitry, which has never disappointed me.
60V configuration for the SGS module: set Jp231 open (2&3) and Jp232 closed (2&3). We obtain a 60V output to power the SGS module. Some may wonder why Dz233 is present. Here's the explanation: given the long time constant imposed on the charge of C235, C236 (resistors, capacitors, and the charging current of 1mA), waiting for the voltage to stabilize would become exhausting. Dz233 brings the gate drive voltage almost instantly to the Vin-Vz value, about 40V. Then the voltage continues to rise due to the time constant and the value of the main Zener diode; Dz233 ceases to exist because the voltage at its ends goes below the Zener threshold, and the rise time of the last 20V is entrusted to the time constant mentioned above. This fact is not very decisive in this configuration, which only supplies 60V, but it has its weight when the voltage to be supplied is 300V.
24V configuration for the HDP01 module: to obtain 24V, a voltage suitable for driving the HDP01 module, Jp231 (1&2) must be closed, which puts Dz232 into the circuit at 27V, and consequently Jp232 must be open (1&2), since the “accelerator” function is no longer necessary, indeed, it would overdrive Dz232.
The stabilizer is preceded by a depletion MOSFET connected to the rectifier bridge, configured as CCS, and used as a damper; it limits the turning-on peak and the capacitor charge peaks, and greatly attenuates the disturbances on the power line due to diodes' switching (thanks to Eng. Chiappetta - project Valvolinea). Pay attention to the "Vgs off" of this mosfet (T233), since it determines the value of the resistor connected to its source (R236). The value on the schematic (15Ω) may vary depending on the Vgs off, the goal is to obtain around 72V with load on the + pole of the first capacitor (C231), so vary R236's value according to this.

Delay circuit (schematic in picture 3, PCB in picture 9). It ensures that the output is shorted for approximately 120 seconds from the moment it is turned on. It consists of a timer made with an NE555 and a relay for small currents, whose coil is driven by the NE555 after the time set by the constant R171/C174.

Relay input selector (schematic in picture 4, 2 PCBs in pictures 10 and 11).
It is optional and uses 6 miniature relays with contacts in an inert atmosphere. Their coils are driven by the rotary switch placed on the frame, 1 way, cheap, but reliable. The switch sends to the single relay to be activated the voltage of a power supply similar to that of the delay circuit.

The power supplies of the delay circuit and the input selector have “tube” origin; they use the heater secondary (6.3Vac) by duplicating its voltage to serve the relays. From the schematic, the two power supplies are not evident, it seems to be only one. Being a dual-mono project, the selector power supply is mounted on the PCB of the left channel, and the whole delay block is on the PCB of the right channel. The silkscreens help to understand which components to mount where.

Input and output wiring.
Three solutions are possible for connecting the inputs:
• wired connections, as per the schematic in picture 5, which involves the installation of 12 shielded cables whose braid must be connected only to the ground of the RCA pins, while the opposite end, connected to the selector, does not have the connection of the braid, so a heat-shrink tube must be placed both to isolate the crown of the braid and to mechanically reinforce the central conductor; in this case the selector must be two-way, and of excellent quality; the schematic guarantees a correct path of the signals and the grounds;
• connections made with miniature relays, schematic in picture 4; for the wiring, again consider the schematic in picture 5, starting from the central contact of the selector, which can be cheap, but must be reliable;
• use 12 pieces of unshielded copper wire, of excellent quality, to connect all the inputs to the selector, which in this case must be of excellent quality again.

I chose the first solution, even if the 8 minutes of work for each wire make this activity very boring, which I had to spread out over 3 days. I also see the second solution as a good one, but I have never tried the alternative of unshielded wires; it is used by many DIYers, but I cannot say if it is a good solution.
The last connection to make regarding the input signal is between the volume PCB and the motherboard, also made with a shielded cable, following the wiring schematic.
For the outputs, use the same shielded cable already used for the input signal.

Other wiring.
• connection between IEC socket/fuse holder to transformers: this is made out of a short piece of 3x0.5mm2 mains cable inserted into an antenna cable braiding, connected with the earth conductor only at the end connected with the contact block; everything must be sheathed in a heat-shrink tube; all the conductors are connected to one side of the contact block, the transformer primaries are connected to the opposite side of the contact block;
• connection between the right channel delay circuit and the delay relay on the left channel: this is done with a pair of twisted conductors connected to the appropriate PCB points.
• optional connection between the selector power supply and the relay selector, if you choose this solution.

Printed circuit boards. The PCB of the SGS module (pictures 6 and 7, quantity required: 2) is the result of a unique, reversible design to give a symmetrical appearance to the internal assembly of the preamplifier.
IMPORTANT - sometimes the components do not match between the two faces (see T211, R224, and R225); it is not a silkscreen error. Near T213, two jumpers must be made between the two pairs of small SMD pads that are very close to each other; these jumpers swap the G/S of T213, and are made with a drop of tin on the face opposite to the one where the components are mounted. One pair of pads is behind the drain, the other pair is in correspondence with the gate or the source, depending on the face considered. If you forget to activate these jumpers, the module does not work. Thanks to this shrewdness, it was possible to keep the PCB unique, reversible, and symmetrical.

The PCB of the HDP01 module (picture 8, quantity required: 2) is also reversible in symmetry and allows the use of most JFETs and BJTs available on the market.

The PCB of the motherboard (picture 9, quantity required: 2) also derives from a unique, reversible, and symmetrical design: to assemble the two channels, simply mount the components on one face for the left, and on the other face for the right. Since everything is arranged reversibly, except for the NE555 socket, the delay must be mounted on the PCB of the right channel, and the power supply for the selector must be mounted on the PCB of the left channel.
Nothing prevents that, for needs or opinions that I cannot imagine, one uses the same face of the aforementioned PCBs to create the whole assembly, which will then not be symmetrical. In this case, the right faces must be used for both channels, due to the non-reversibility of the NE555 pinout.

Volume PCB (picture 10, quantity required: 1), solder it directly to the potentiometer; it contains a low-pass filter to eliminate radio frequency disturbances.

RCA input and output pin PCB for the relay selector solution (picture 11, quantity required: 1): connects the RCA pin grounds and supports the relay PCB.

Relay and RCA pin hot sides PCB (picture 12, quantity required: 1): carries the miniature relays, it is to be soldered to all the RCA pin center connections, and attaches to the PCB below with a screw bringing the mass connection.

RCA input and output pin PCB (picture 13, quantity required: 1): optional, makes it easier to connect to the jack pins in the case of wired assembly.

If you want to build the Talia in a 1U "Slim Line" cabinet by HiFi2000, please read the mechanical notes about it.

Final remarks.
The point “Expected sound” still needs to be developed.
Can I get by with two words?
The best.
Did I get by? No, right?
So I will resume the teachings generously given by the eminent engineer Fulvio Chiappetta on some Costruire HiFi issues, and during his seminars.
Given that distortion cannot be avoided (!) in any circuit, we might as well shape it in a way that is acceptable to our perception. This is a concept that struggles to enter the brains of many pseudo-audiophiles present in significant numbers in the various forums, even here (5 at least). A resilient concept, I would say stubborn, against which there is no logical explanation that holds up. I do not dispense education, I gave up, and I stopped answering those silly questions.
How do you shape distortion? It is not difficult, it is like playing the stock market: buy at the minimum and sell at the maximum. Right? Do not worry, it is not that difficult, it is a matter of finding the polarization of the gain stage that produces the lowest harmonic distortion rate while configuring a monotonically and "appreciably" (I would add) decreasing trend of the amplitude of the harmonics. A system of 2 equations. I will not go into detail here, I do not dare, just search for Engineer Chiappetta's articles, or Google "monotonically decreasing harmonic distortion".
When I calculate a gain stage, sometimes it happens that the total harmonic distortion is very low, that the second harmonic is also very low, but that the third harmonic is slightly higher. I discard the calculation, and go and look for another working point so that the total harmonic distortion could even be higher than that identified before, but the harmonic content presents a third harmonic of amplitude lower than that of the second, the fourth lower than the third, and so on. The term “appreciably” I used in the previous sentence means that I prefer the third harmonic to be significantly below the second, and not a tad below, even if this means a higher THD.
What happens in the Talia?
In the pure cascode configuration, without the contribution of the BJT to form the CFP, LTSpice highlights a THD of 0.16%, which shows a harmonic content mainly of the second harmonic, and monotonically decreasing, in perfect compliance with what was expected. A very “determined” sound, pleasant, thick, never banal, descriptive, at times just flattering.
In the CFP configuration, we have an almost non-existent distortion (0.0043%), whose trend is still favorable. A pure sound, very appreciable, precise and clear, very clean, never abundant, never harsh. But the soundstage "seems" to be slightly less coherent over time compared to that offered by the pure cascode, the instruments move a bit. These are my impressions. We are entering the field of psychoacoustics, and I stop here.
I am going to trivialize, and I venture a summary of the listening experience: we like this kind of distortion! The instruments are placed without hesitation, and they remain there. And I knew this, from the many positive reviews that had come to me from those who had built the WOTS Le Grand and the Tre, which by their nature dispense that very welcome harmonic content. Hey, 5 fellas! I told you where to stop reading. What are you still doing here? Trying to learn something? Don't say, don't waste our time, inform yourself, please. Even Google's AI knows!
The two configurations, pure cascode and CFP, are obtained by moving the JP211 jumper; the pure cascode has a slightly lower gain than the CFP configuration.
I'm sorry for the enthusiasts (mainly Italian), I have not yet conducted an exhaustive listening test with the HDP01 module. When tested by switching it on and listening for a very short time, it works perfectly, and I'm sure that it maintains without difficulty what Eng. Maltese said in his presentation. To me, it is a very interesting solution.

The text was too long, so I had to split it. The following posts will end the description of the project.

Feedback for the hybrid amp

Hi there all.


I have for a change sim a schematic who I want to try on mine hybrid who do play without feedback for a long time.

Problem is that I need high open loop, and that is with tubes less easy as with transistors.

But the cascaded 12AX7 do quite nice. Are there more ways to get a preamp with high gain? Did try also cascode
but not such a succes as this one.

see the schematics and sim. output leven 4 amps, so looks very good that distortion.

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universal speaker

hello everyone.
I suppose that besides myself there are many other people in this forum who over time
have set up a laboratory dedicated to audio measurements and the construction of speakers (containers) with the help of software.
it can happen that you are assailed by a considerable amount of curiosity and desire to try a loading
that we have immediately been strongly attracted to and after having built it move on to designing a different one.
there are simulators but the proof of the pudding is obtained only by putting the speakers to work
but every time you do this it becomes expensive to change them every time because their resale is not guaranteed.
so I come to the point, is there a type of speaker (not definitive) to be used only for experiments?.
many or all of you will answer me no because each loading requires a speaker that has suitable characteristics (Thiele-Small).
a speaker that can work for some loadings (BR,TL,HORN) would not be a universal driver but I could settle for that. in this period for example I am working on a hybrid MLTL-karlson. an old field coil could be good?.
bye thanks

Making small DAP, looking for your wisdom

Hey folks, newcomer here after being a reader for a few years 👋

I’m working on small DIY DAP (imagine if iPod Shuffle and Shanling M0 had a baby) and after 1st prototype I’m trying to squeeze the best possible performance from the limited PCB space / orientation and will appreciate your help

I did split design into multiple PCBs and arranged them into C-shape - you could see high freq. and audio parts + dimensions marked on the images attached:
current design.jpgpcbs photo.jpg

I don’t have any experience designing top-notch audio PCBs (or low noise analog in general), so all extra opinions/thoughts are very welcome, but my main questions are:

  1. I’ve used PCM5102A to just have I2S sound output - but does it make sense to use smth better? I’ve heard that biggest contributor to noise will be the rest of the design and getting better DAC with lower noise floor is not really worth it since the rest of the design won’t be able to go that low.
  2. What is the best amplification option to squeeze in here considering it will be for headphones only? I don’t drive anything demanding, but having some margin for the future will be useful.
  3. Any recommendation on analog parts (i.e. using smth else than mlcc caps, any specific manufacturer e.t.c.)? Any examples of best in class PCBs of similar size to learn from?
  4. If I will route audio signal from amp to jack on another PCB via headers - could that introduce significant extra noise?
  5. Is common PCB stackup S-G-G-S is okay or I could get extra shielding by using G-S-S-G? Considering that on DAC PCB below analog signal it will be mostly G-G-S-G since I doubt I would need 2 layers for analog part and will just pour 2nd signal as well.
  6. Will making PCB 0.8mm introduce some extra coupling that will be critical compared to 1.2mm stackup?
  7. Should I put some shielding on PCB with MCU? Like the ones that could be usually seen on ESP32 modules or on XIAO boards - thin metal enclosure wrapping top layer components to limit noise coupling to DAC PCB. Or it’s just way too far to influence another PCB?
  8. I plan to use connected ground planes - i.e. no separation between analog and digital. Or it’s a good idea to split them and then connect using ferrite beads and other voodoo magic?
  9. LDOs are the best choice for portable analog power source, right? Since buck & switching will introduce more noise.
  10. Using same LDO for DAC and MCU DVDD is okay, right? Only AVDD should have separate LDO to decouple from digital noise.
  11. Should I do vias stitching if I don't have any wireless onboard? Could that help shield any external noise or it's not an issue unless I add bluetooth.
  12. I have another option of stackup where PCBs will be on top of each other (”Parallel Stackup” image) - will that couple way too much noise from digital PCB?
parallel stackup.jpg

In overall I want to create end game portable DAP with buttons 😄 So any drops of wisdom from you folks will really help to make it happen! 🙌

My ADCOM GFA 5400 needs new power switch

My ADCOM GFA 5400 was made in 1995 and I was using it until about five years ago. Yesterday I hooked it up to a preamp, CD player and speakers. When I pushed the power button nothing happened, no clicking, no LED lighting up and no sound. Today I opened it up and the fuse is good, and power is going to the power switch. So I think I need a new power switch. I went on the ADCOM parts site and the switch is not available. Does anyone know where I could get one? And if has been discontinued is there an alternate switch that would work?

Thanks,
henrylr

Racer 8

This is the racetrack 8" driver that I introduced in the Bass Shelter thread. It is an unbranded driver sold as a replacement for a name brand boombox. It's 130mm slim across the narrow, and I can think of some fun applications if the driver is good. Build quality is visually superior to Dayton Reference passive radiators on hand. The driver is sold on Ali and is fairly expensive at $104 AUD for a no name. It's proportions and the promise of a decent 40hz low end as touted by the name brand makes it interesting enough to take a punt on. It will also suit the intended beach environment

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Comes very well-packed
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On the Datsv3, driver not yet broken in
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Will stick to plan and take power from the sub section of the current Logitech z623 for breaking in and initial listening. The driver doesn't come with radius info for the cutout, so I will have to do some trial and error to get it right. It does come with a template so I can hack out a cutout for breaking in purpose

Will use the laser cutter CNC router thingy to get the cutout and recess just right for a proper cab in due time. The driver also appears to require a dress trim. The screw holes are very close to the edge of the rubber roll surround glue line. I hate the idea of screw heads crushing this. What do you guys do in this situation? The best I can think of is an alloy plate dress ring in CNC and with matching screw holes to clamp down over the driver's folded up edge. Mounting from behind the baffle will solve this too and have the same clamping effect. Actually, this may not be a bad thing and the material thickness can contain the excursion and just allow a simple grill over the driver

Welcome your thoughts on this unit
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Rotel RB-870BX renovation, C609 question?

I am fixing one of these which includes changing most of the caps, the C609 highlighted in the picture is a 1uF 50v elyt which I was intending to change for a Nichicon gold tune but it turns out I don't have the correct value.

I have a 1uF MKT film cap and larger values 47uF/50v gold tunes but as I am not into details of amp circuits I don't know what C609 is doing or what to consider from a cap point of view?


nxMyydz.jpg

3-Way Studio Monitor System

Hello,

I would like to show you a project I'm just working on. It started with an idea for bass improvement and eventually turned into a design of an entire system.
The subs are already made but not tested for possible improvements. Firstly I have to build the monitor units which contain the amps (Hypex FA 253).
The monitor driver will be Sica 5.5 C 1.5 CP. The sub driver is Peerless by Tymphany SLS-P830669

The concept aims to improve my current monitoring capabilities in my studio of 5x5 meter in near field range by moderate room level.
Usually I use monitors on stands with a distance of 1m to the front wall. So I will build the speakers into stands. The stand will also contain the amp, which makes a free positioning of the subwoofers more convenient. The subwoofers can also stand within the bottom area of the stand for listening in midfield and to blend into the design of pseudo single speaker.
Bild 13.04.25 um 23.20.JPG

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Introduction

Hi , @6L6 I’m a student from india and am here because I’m currently working on my first DIY amplifier project and am really interested in learning more about audio electronics. I’m exploring different amplifier topologies and wanted to make a aleph j ,but didn't able to the pcb because of high shipping cost and customs , if u could please share the gerber file of the pcb then it will be a huge help, in my first diy amp project,I will be eagerly waiting for your reply

Ground lift switch wiring

Hi.

I've recently done some work on my valve amp.
I needed to remove the main circuit board so I could install new valve bases. I (thought I) took pics of all the wiring etc however I'm left with one unidentified wire which comes from the ground lift switch.
The other wires from the switch go; one to the chassis and one to the volume pot (case).

I'm guessing the remaining rogue wire should go to the negative side of the main PCB ?

PXL_20250101_153237269.jpg

Transylvania GR-18 Cassette Deck. Left ch. dead

Hi guys,

Here’s a nightmare repair which seems not to have an end. To date, I’ve been struggling with this device for months.

It’s Transylvania (?) Cassette Deck, Model GR-18,I think is dated somewhere between mid/late 70s not beyond early 80s, sure hasn't been used for a long time. Searching the web, I found out that Transylvania was part of Asahi Corp. further information are nowhere to be found. I only found a single picture of this model, and of course no schematics at all.

The problem is: dead left ch. I mean totally dead, no Rec., no Play, not even on the phones. The other channel is ok.

Things I already did:
1. Cleaned and demagnetized the heads
2. Thoroughly cleaned the rec/play multiple switch by disassembling it and cleaning every single contact.
4. Checked all the channel components on board and all the semiconductors also off board.
3. Replaced the Dolby IC, based on what someone said to be “common fault”
4. On Mooly’s suggestion I also verified continuity between the cables coming from the heads and their respective connection on the PCB.

I'm sure someone has already put his hands on it, the signal cable coming from that channel head was soldered on the same pad of the ground. So it could be the same thing for one or more of the many cables attached under the PCB.

The problem with this unit is the top side of the PCB is hard to access, unless desoldering 20+ wires attached to random zones of the PCB itself (top & bottom side). Any move of the board can lead to wires disconnection, it actually happened several times, and you can go crazy finding out where they came from. Very bad design and construction: cable mess, no silkscreen etc.

All in all, quite a challenge for a hobbyist like myself. Hope you guys can help, thank you.

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Hi everyone, I'm here to learn FM stereo alignment

Hi everyone,

Long time lurker, new member. I've been repairing and rebuilding vintage stereo gear and test equipment for about five years now, but I won't be satisfied with my skillset until I can confidently and properly repair and align FM stereos. Over the years I've read a ton, I've got a good grasp of what's going on in the various stages of a stereo receiver and I've gone down the rabbit hole of buying decent (but not always working) test gear. Is this the right place to find people who can help me improve my skillset?

New Stasis front end

Not too long ago I received a request for some help from an owner of
an old Threshold Stasis amp, whose amp was broken and where some
previous work had left some issues with the circuit board.

As many of you will know, I have lots of time on my hands 😉 and a
sentimental desire to revise old product in light of the wisdom I have
acquired in 50 years.

So.

I revised the circuit, making it more simple, and laying out a circuit board
which should be usable for all known versions with minor adaptation.

Attached are various schematics and graphics and also the gerber files
for the circuit board, which was made my pcbway.com, who will ask
what dimensions are, which is 5.5" x 3.5"

This thread will also serve as the location for asking additional questions
and so on.

:santa2:


EDIT --

Gerbers and info for Nelson's front end that fits the old Stasis chassis and has mounting holes (post #281) - https://www.diyaudio.com/community/threads/new-stasis-front-end.363701/page-15#post-6447824


@Zen Mod made PCB for this project, information and gerbers can be found here :

Front End (post #282) - https://www.diyaudio.com/community/threads/new-stasis-front-end.363701/page-15#post-6447826

6-deep output stage (post #283) - https://www.diyaudio.com/community/threads/new-stasis-front-end.363701/page-15#post-6447831

8-deep output stage (post #284) - https://www.diyaudio.com/community/threads/new-stasis-front-end.363701/page-15#post-6447835




--

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For Sale SPECTRAL SRD-1000SL

Impeccable exterior. Doesn’t play, most likely defective laser. Philips swing beam variety. Built like something going to the moon.

I’d like €500, + shipping, but I’m open to offers. Priced to go.

Regards
RM

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  • Poll Poll
Krell 400xi / 300iL amplifier amplification

Is transistors overlodead and not correct now or just caps?

  • Transistors couse sinusoidal distorsions

    Votes: 0 0.0%
  • Capacitors couse sinusoidal distortions

    Votes: 0 0.0%

PLEASE ADVISE ME ...I bought Krell 400xi I am considering to hold it or give up and send back.. I notice 30Hz 4dB lower sinusoidal signal compare to 1 KHz and more on 20 and 15 Hz.
Channels difference at 1khz is at preamp section 0.0dB but on speakees posts 0.2dB but at 30 Hz both channel difference is 0.7 dB . power section non polar input 47uf capacitor right is 43uf left is 46uf . MAIN PROBLEM IS : ON OSCILOSCOP AT AVERAGE VOLUME LEVEL 1kHz sinusoidal wave is fine but below 100Hz looks like cut out tops and buttoms 30hz worse. . IS SOME SEMICONDUCTORS BROKEN OR INPUT NON POLAR CAPACITORS COUSE SINUSOIDAL WAVE TOPS AND BOTTOM WITH DOUBLE BUMPS PROBLEM? I AM NOT TOO EXPERIENCED WITH OSCILOSCOP . AMP SOUND WITHOUT STRONG BAS BUT JUST REGULAR.

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Maeshowe Build

As noted in a few other related threads, I’ve been working on building a pair of Woden Maeshowes. I’ve lusted after Silburys for years, but I’ve got some Pluvia 7.2HDs so Maeshowe it is.

I’m including the optional baffle braces, because I think they are essential. Material choice was very limited in my location so it is just decent quality 15mm ply. Not void free, sadly, but not really full of voids either. It’ll do.

They sell it in (roughly) 3’x6’ sheets here, so none of the cut sheets were appropriate. A quick google search got me to https://www.opticutter.com/cut-list-optimizer#form which proved to be the perfect tool for the job.

I invested in an inexpensive table saw to do the bulk of the cutting and I am glad I did. So easy to get all pieces with identical width, in order to ensure a perfect fit when that final side panel gets glued.

I was intimidated by the curve on the back of this design, and brainstormed a bit about how to draw that curve at the correct radius. Another google search brought me to https://www.blocklayer.com/arc-layout , which again worked perfectly for the task at hand. It was a bit tedious, but I only had to do it once and then use the first piece as a template. Jig saw did a decent, if not perfect, job.

IMG_4775.jpeg
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IMG_4799.jpeg


Progress is slow due to “never too many clamps” and I am allowing extra clamp time because it is pretty chilly in the house.

Have some Radian Neo745PB CD's + Ciare PR614 Horns to use in a 2-way build.

I have a pair of Radian Neo745PB compression drivers and a pair of Ciare PR614 Horns to mount them on and I am looking to building a 2 way system to complement them and finally give them a proper home.

I am in search of a woofer to start the design process and since I am located in Italy it makes sense to limit the field to the great speakers made here. 18 sound, RCF, B&C, Ciare, would be easy to source here. Here is my active list in a google sheet which your invited to add anything that might be of interest. LINK

How to go about marching a woofer to these compression drivers? I am open to suggestions but I was envisioning a 10" or 12" woofer. How does the response curve, the efficiency, the power handling guide the woofer selection?

The setup will be BI amped with some Powersoft Q3002's and I plan on doing active crossovers with a miniDSP or a rack mounted DSP such as a EV DC-ONE

The intended use is for a mix of home stereo, portable DJ setup, band practice, and just general PA usage. The plan is to add a sub at some point.

The budget is around 160EUR for each woofer.

Volume wise I dont have a hard constraint but something in the 40-55 liters sounds reasonable.

What other requirements or constraints am I missing to nail down a driver before moving to the cabinate design fase?

Thanks

Adding a Korg NuTube buffer to solid-state chain – bias & polarity tips?

My current front-channel chain is Apple TV 4K → Rotel RSP-1576 Mk II (Dirac/DSPBypass) → Odyssey Kismet monos. The system is clean (<0.3 % THD at 80 dB SPL, measured with REW), but I’d like a touch of “tube-ish” H₂ warmth without changing the frequency response or bass control.

The First Watt B1 Korg NuTube looks perfect: unity gain, low Z-out, user-set bias.
  1. Bias sweet spot – People quote anywhere from 10 V to 18 V on the NuTube plates. Where does H₂ sit ~0.4 – 0.6 % (with H₃ at least -20 dB) and which side of the 12 V “phase flip” do you prefer for stereo depth?
  2. Absolute polarity – The buffer inverts; do you guys flip speaker leads or add a second inversion stage? Any audible downside to leaving it 180° out?
Goal is “season to taste”: be able to dial the bias or bypass the buffer and land back on the original Dirac calibration unchanged.

ALSO: If anyone with a finished build wanting to sale please DM me!

Neurochrome LM3886DR Build

Hey guys,

a few months ago now after a failed attempt at buying a chipamp.com kit, Tom mentioned he would be making the LM3886DR Kits, so I immediately put my name down for the first one. It turned up quite a while ago, however I've had a busy few months and haven't had a chance to start putting it together until now.

Cutting the box open, everything was individually bagged and labelled from Mouser - so much more convenient than trying to figure out a handful of components. Thanks to Tom for helping me out with getting all components along with my boards.

20160818_132124 (Custom).jpg

20160818_132254 (Custom).jpg

Printed off the design documentation - super comprehensive and walks you through every step of the build (including about as much information on the chip as you're going to find anywhere). Easy enough to follow that providing you can solder, it's pretty hard to go wrong building this kit.

20160818_134657 (Custom).jpg

Started populating the boards with resistors. Again, having everything bagged and tagged makes this process super easy.

20160818_142212 (Custom).jpg

Power resistors in. Left 5mm gap for airflow around the resistors.

20160818_143059 (Custom).jpg

All caps in.

20160818_145653 (Custom).jpg

Connectors and inductor in.

20160818_153546 (Custom).jpg

LM3886 in (easy to solder, was annoying trying to lift it a pinch off the board for clearance - more due to my fat hands than any issues with the kit).

20160818_154820 (Custom).jpg

Chip attached to temporary heatsink for testing. Definitely overkill, but I had it floating around. Checked DC offset at the output - 3.5mv; perfect.

20160818_161411 (Custom).jpg

Attached to a test speaker and lab power supply. At the start of testing I was hearing some terrible things - the cone flapping around, continual cracks and pops, etc. Figuring I had something wrong I went through and verified all components with no success - turns out I had a faulty signal generator! Switched over to another one and it was perfect.

-----TBC-----

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Please wake me up from this PCB nightmare [Krell KSP-7B Power Supply]

I am working on this power supply PCB. It's double layer, 2.5 mm thick, and pretty much cooked to a crisp after decades in an inadequately ventilated case.

bottom layer.jpg


top view.jpg


First cap that I desoldered, I ripped a trace off the top side of the PCB. I figured the solder wick method wasn't going to cut it this time, so, not wanting to cause more damage, I ordered a Hakko FM-301 tool which I had been wanting anyway (it's a fantastic tool, by the way).

Second cap I desoldered (with the Hakko this time)... I ripped a trace off the board again. But I noticed that the trace was pulled up by a repair eyelet that was stuck to the cap pin. Then I saw that the exact same thing happened with the first cap.

Next I desoldered the DB9 power connector, and 9 out of 9 pins came out with an eyelet attached. I realized the eyelets were on the pins pretty tight, and that they were being pulled up not necessarily by leftover solder but by mechanical friction.

Now, I'm pretty sure those Roedensteins and DB9 connector I pulled out are original, yet every single hole had an eyelet in it, so it looks like the factory built the PCBs that way instead of plating the through holes? Isn't that unusual?

Anyway it is what it is, and I have to deal with it. These eyelets and pads are incredibly fragile, so the less work the better. But, seeing how baked this thing is I think it's reasonable to do a full recap, test all components, put fresh thermal grease on the transistors, etc.

Where I have access both top and bottom sides I can apply heat to both sides at the same time, so I think pulling out diodes, transistors and resistors without breaking anything else will be OK.

Not sure what to do about the caps though. They're mounted flush with the PCB, so there's no room to cut the pins off. Maybe I can just tear them apart to expose the top of the pins

The PCB traces are .35 mm thick and they are super strong, so it shouldn't be too hard to epoxy them back down. Then I can replace all the eyelets and pads. Which will leave me with a severely cooked board that looks like absolute crap, but at least it will be in better shape than now.

I also thought up two other (creative!) ways out of this mess:
  • Have the PCB cloned by one of those shops in China
  • Toss the PCB and rewire the thing point-to-point

I don't know what else is there? What would you guys do??

Thanks in advance, all thoughts are welcome! 🙂

SoundStream TXP4.3500D parts ID

I have an amp with some issues needs addressing…the PIC controller is malfunctioning resulting in the power pwm keeps pulsing on and off, I’ve modded a direct drive circuit thus keeping the power supply on (module was taken from a JBL amp riser pwm card) IRF1404 was used with 10ohms resistors. That worked but my power led flashes alternating with clip led and protect led (it played regardless as I assumed the malfunctioning PIC.
The customer brought back the amps after many months of continued flawless working, this time two outputs are bad thus taking out the power FETs. I don’t know what sub to use in the output side as they are scruffed and etched with C2 on them (currently I’m trying some IRF640’s which seems to play and seems cool) can anyone make any other suggestions for output selection for this amp? The audio drivers are IRS2092S scruffed as C7.
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Adhesive for joining 3D printed baffles to wood

The drawing in the attachment shows the baffle with the rabbet joint. The wood side panels will be adhered to the baffle on the end grain and to 20mm of the inside face. The wood I'm using for the first pair is 1/2 inch prefinished BB.

Construction adhesive should be strong enough but I would prefer something thinner. Epoxy glue? Maybe there's something I'm not aware of?

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Krell KAV 280P Schematic

Hello,
Looking for electrical schematic for Krell KAV 280P.
Thanks.

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BP1048 DSP modules - issues with software

Hello guys,

Recently I learned about a Chinese DSP BP1048 chip and thought that it could be a cheaper alternative to ADAU chips by Analog Devices.

To test it, I bought two modules containing the chip:

1. BDM3P by Wondom
2. some generic no-name module (see attached photo)

module 1 BDM3P.png
module 2.png


To operate it software called ACPWorkbench is required. Initially I downloaded version 2.24.2 from 2021 provided by aliexpress seller, later I found newer version, 2.43.3 from 2024 I think.

I watched a video tutorial about this software, link below:

Login to view embedded media
And what is strange to me that first of all, I don't have all the options that they are showing in the video (for example: my boards don't have any GPIO options available, also there is no possiblity to adjust L and R channels separately).

1748361934215.png


1748361977158.png


Also even though the two modules I have use the same exact chip, for each of them I get different options in the software!

1748361816301.png


1748361721898.png


I thought that perhaps FX list contents depend on on which peripherials are activated? But I can't seem to find relevant settings. For example, module 1 uses 3 regular potentiometers connected to ADC to control volume, bass and treble, but I don't even see how it is set up in the software.

I don't think it's software revision issue, as the video was uploaded in March 2021, so software I have is most likely newer than theirs.
Does anybody have some experience with these chips and perhaps know about my problem?

No sound from new McIntosh C22-MKV preamp

Just purchased a new in the box McIntosh c22-MK V tube preamp. When I hooked up my new Sonos Port streamer to the line inputs of the c22, still no sound out of all 3 of the line inputs. I even checked to make sure the mute on c22 is off, as well as connecting a different source to the input of the c22 and still no sound at all.
Has anyone who owns one of these preamps ever experienced anything such as this? This is a first with me from a McIntosh product, as I own many other Mac pieces I purchased new. Please let me know. I believe this could be a warranty issue. Sincerely

New build problem

Can one of you wonderful people please help a relative newbie with a strange problem.
I have recently built this tube amp from the attached schematic and while it is giving me full range audio it always sounds like listening to well worn records at any volume.
Running on Quad esl 53’s.
My 303 sounds fine so not the speakers.

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More or less overlapping

Consider traditional two-way speaker system comprising a woofer and a tweeter. We're going to build a crossover for these drivers. Assume the measurement has been done and the crossover point of 2.5kHz was chosen. There are two choices here for an educational purpose: 12dB/octave with reversing polarity of the tweeter and 24dB/octave without reversing polarity. Goal for these two choices is to imply ASSUMING two drivers are perfectly "in-phase" with each other--any phase issues will not be taken into account. As a result the frequency response of the speaker system will also be assumed FLAT. The only difference presented on the amplitude curve are the slope rates of all drivers between 12dB/octave and 24dB/octave. The most important part is it's obvious that the 12dB/octave will create larger overlapping area and vice versa for the 24dB/octave. Despite flat frequency response curve and theoretically correct phase behavior, which way do you prefer between having large overlap (12dB/octave) or less overlap (24dB/octave)? And please give some supporting reasons.

1748672181695.png


Cr. for picture: RANE Corporation

Denafrips Terminator II WIMA Caps

Looking for opinions. I'm a new owner of a Denafrips T2 DAC. Interesting piece, what is also interesting is their choice of WIMA caps. MKS from what I can tell are used in the filtering section, and I think FKP on the output. I've had mixed to poor results with WIMA polyester in the past, maybe in the filter section they might be ok, augment/bypass the electrolytic bank, but on the output and/or coupling the DAC resistor ladder, I would have chosen different if I were designing maybe.

Interested in opinions. I was looking at some Charcroft Audio Silver Mica as a possible FKP replacement, fitting MKP will be tough and wouldn't want to make a mess of it visually. Some polys in the filter section would have been nice, what are the thoughts on this? It's as good as it gets with WIMA MKS/FKP?

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Thoughts on the old Schade PP series feedback?

Just curious if anyone has built/analyzed anything like the original push-pull series feedback IT circuit.
schade_pp_orig.png


Doing a bunch of spice modelling with some different pentodes, the surface level analysis looks good, but two troublesome issues stand out:

1) The 0.1uF capacitors in the feedback path cause a huge increase in gain at low frequency as the feedback trails off. If the interstage can pass frequencies below 10 hz, feels like things could be problematic. I had to go up as high as 3.3uF in some cases to kill the hump down in the 3-5 Hz range. Certainly you can get caps that size but it feels a little ugly to have both an interstage and also caps large enough that they probably have to be metallized.

2) I have some bifilar interstages I was thinking about using if I did try to build it, but it looks like this circuit is not going to be ideal for inter-winding capacitance no matter what, as the lowest AC voltage you could put on the wires that are physically at the center of the winding would be the voltage grid-to-grid for the drivers. The voltage swing on the feedback dividers is very large so you'd essentially setup the secondary to 'drive' the feedback path while the grids are the ground side of the windings.

Probably the answer is 'Don't do it, this is not worth it.' Since you have really high drive voltage through the IT, you'd basically have to build a driver setup powerful enough to drive an IT at low distortion and enough voltage for big power triodes anyway. You'll have basically built a monster PP driver that could just drive PP 300B's anyway. In theory the series connection doesn't present much load to the IT, but Rp of the driver is still going to dominate distortion through the IT, plus whatever the parasitics I mentioned above are doing.

Darlingtons MJ3001 and MJ2501 replacement

Hi guys.
I have a question for you.
I have a class A power amplifier built many years ago by an electronic engineer friend of mine that uses a single pair of complementary MJ3001 and MJ2501 darlingtons for a power of 32W per channel, powered by 32V. It worked perfectly and with a nice sound. Unfortunately, one channel no longer works. I would take advantage of this to update some components such as filter capacitors and other things. I need to know which darlingtons I can buy as a replacement since the originals are no longer available and also to use something more modern and superior, since all the work is entrusted to a single pair.
I would have thought of the BDV66D and BDV67D pairs or, even better, 2SD2449 and 2SB1594, but I need your authoritative opinion.
Thanks.

Introduction

Hi,

I am new here but have been a passive reader for a long time - I did recently design a RIAA amp for my dad based on an old Erno Borbely article, but using AD797 opamps. Even if they are not the best for the application the result was really good.

Now I am looking into building a new power amp - and I have my eyes on the Wolverine.

Summing L/R signals, then sending to a distant subwoofer

I'm looking to use my preamps 2nd set of outputs to send a signal to a sub that's across the room. There's already a single coax wire in the walls to accomplish this. It's maybe 30ft of cable. The sub has its own amp, and can accept a single line level input.

I plan on using the textbook circuit for summing 2 inputs using an opamp (maybe TL072).

I'm not sure if the length of the output wire needs special considerations. If it does, there must be a simple device that handles this, but i don't know what that might be.

Thanks!

LM3886 Diy PCB starting picking up CB or conversations?

I have an amplifier with several LM3886 boards I made. I knew them picking up radio might be possible, so I added a low pass filter on the positive input terminal. I have a 10uf blocking capacitor, and then a 20k input resistor. This goes to the LPF of 1k resistor plus 680pf capacitor. An online calculator says this is a 3db cutoff of around 234khz. This was noise free for the last year, but it suddenly started picking up garbled conversations of some kind. Just like once or a twice a day it'll pick up what sounds like someone talking into a tin can, sort of like that teacher from Peanuts.

Anything else I can do to harden this up a bit? Here is a rough schematic of the input and the feedback loop. I have the output recommended components and included inductor.

Thanks.

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For Sale JohnAudioTech JAT501 Class AB Amplifier PCB pair -- LAST ONE

JohnAudioTech JAT501 Class AB Amplifier PCBs from JLCPCB

4 of 5 pairs sold. Last final set remaining.

$10 Canadian plus shipping.

Made from JohnAudioTech's publicly available JAT501 gerber files.
No modifications to John's design were made.
Inline layout, board revision 1.1

High quality PCBs. Standard 1.6 mm thickness, and completely flat.
Two layer, double sided, blue solder mask.

This sale is not for profit. It is being offered as a service to the community. And to help promote John's channel.

These are just the bare PCB boards. You have to source your own components, assemble, and test.
Bill of Materials (BOM) including part numbers is also provided.

Includes two hand made and tested Thiele network coils (labeled L1)

Shipping within Canada is $3.00 using Canada Post oversized letter mail (sorry no tracking).

Shipping to USA is $10.00 using Canada Post Tracked Packet - USA. Includes a tracking number.

Known issue, the board stencil has some typos.
Transistors labeled 2N5551C should read 2N5551.
Transistors labeled 2N5401C should read 2N5401.

The "C" variant has a different pinout.
2N5551 / 2N5401 (1. Emitter 2. Base 3. Collector)
2N5551C / 2N5401C (1. Emitter 2. Collector 3. Base)

One board has scratched solder mask that happened at the factory. The exposed copper was tinned. The traces are ok. See attached photos.

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Pick-Up amplifier design adapted to 80 VDC

This is my favorite amplifier design for now that I am listening for 3 months and it still is good for me. In the past I build perfect clone of an Unitra Stereo Hit but it was desined with RIA correction directcly in the amplifier and for some reason I did not like it. The schematic I am posting now could have some further modifications and I have to say that I like simple designs more, I did complex stuf but to hard to repair.

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Spica TC-50 crossover refresh

I have a pair of what I believe are the original version Spica TC-50 speakers. They are a matched pair with matching crossover codes. I have read that the codes indicate specifics of the exact measurements of the parts (i.e. they were matched tighter than the tolerance of the parts). I don't think they sound as coherent as they did 30 years ago when I was running them regularly, perhaps the electrolytics have drifted. I suppose I should measure them.

Here's my question: would it be possible to trim the capacitor values using just a decent quality condenser mic (I have AKG perception 220), a computer and audio interface? I remember that the speakers were designed to be particularly flat in frequency and phase response at the crossover region (somewhere around 1k or 2k if I remember right). In other words, can I measure the crossover accuracy without buying more equipment? (No problem acquiring / compiling open source software as needed.) Thanks for any help.

Output transformers for DACs

Hello everyone who looking for Sound!
:wave:

After years of expirience the output transformers for current and voltage output DACs have been developed. Transformers tested with following DACs: ES90x8S/K2M/PRO/Q2M, AK4x9x, PCM56/58/63, PCM1702/04, AD1862/65, PCM1792, PCM1794A, TDA1541/43, DSC dac and others. Speaking in general, transformers can be suited for any current or voltage DAC both in SE or PP modes.

These transformers are made on high permeable cores, have the low DC resistance at desirable inductive impedance to achieve the best frequency response of the DAC. Transformers provides significant sound quality improvement compared to other output versions. Your music will begin to live a new life, it will sparkle with a naturalness without plastic nervousness 🙂

25GHrhC



D60 trafos.
Transformers I/V stage for iancanada DM9038Q2MPi DAC
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Transformers I/V (V/I) stage Twisted Pear Audio Buffalo-IIISE PRO DAC
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D46 trafos:
1675169440982.png1675169176721.png
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The simplest way to connect my transformers to your circuit without soldering directly to pins (sometime DIYers wants to try different types in quick manner for example) is to use receptacles from Mill-Max:

1-st way is using receptacles with the soldered to them wires. PN is ED90595-ND
receptacles.jpg

2-nd way is using through-hole type of receptacles for PCB (PN is ED10169-ND):
PCB_receptacle.jpg


Here are some impressions from iancanada IanCanada's Latest RPi GB Goodies Impressions... and your tweaks, mods and hints...

And here is the page from Andrea Ciuffoli: DSC Output transformers test

Dimensions D46 type: OD=46mm, H=24mm
Dimensions D60 type: OD=62mm, H=30mm

Frequency Responce of transformer with ES9038q2m at 1.4VRMS (~8kOhm in secondary loaded on 100pF cable): 20Hz=-0.3dB, 20kHz=-0.2dB (rel.1kHz):
ES9038_4Vpp_1m_cable.png

Same, but with 470pF in parallel with secondary:
ES9038_4Vpp_1m_cable+470pF.png

The possible output level and the LF losses are depends on the output current of the DAC (and I/V resistance).

Here is an example of connection:
common_circuit.JPG

Here you can find prices and all the contact info: https://ivxformers.com

Lead time depends on existing orders. Thanks.

The relevant current documentation for my transformers available here: Output transformers for DACs

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For Sale Fluke Multimeters (EU): 27, 27FM, 85, 87

Selling some "vintage" multimeters.

The legendary Fluke 27/27FM were made for demanding environments: rugged, 1m waterproof, quite bulky and heavy and with an incredibly long battery life. Not easy to find on this side of the pond.

The Fluke 85/87 are first-series models. I guess no introduction needed.

I tested them at 300mV DC, 3V DC, 5V DC, and 1V AC. I forgot the resistance test, but I can do it later this week. I have never used the current measurement.

Price list:
27 – 95 EUR
27FM – 115 EUR
85 – 130 EUR
87 – 160 EUR

If you buy a pair, we can share the shipping cost. If you take all four, I will cover the shipping.

PayPal in EUR via Friends & Family or + PayPal fee.

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IB driver in sealed enclosure

I’m planning to start a new project of large bookshelf speakers recently. I have a pair of old-school 10” MB Quart woofers (model QM250TX). It seems they’re infinite baffle type drivers as they are also used in a car audio three-way system (QM340CX). I would like to know if there is anyone using an infinite baffle woofer in a SEALED enclosure? How about the result? Is it good or bad?

I have gathered information about the enclosure sizes of home audio 10” bookshelf speakers, this configuration is so old school as it was found in the 80s-90s speakers such as ADS L880, Braun LS100, MB Quart 390, etc. I found the volume of the sealed enclosures, in the market, varies in the range of 1-2.25 cubic foot. I wonder which side we should use between small or large volumetric, i.e. lean towards 1 or 2.25 cubic foot, for the IB drivers.

I don’t have the T/S parameters of these drivers (QM250TX), I have its successor’s (QM250SUB) though, hope they should get closer. Here is the data:

Impedanz 4 Ohm
Re 3.4 Ohm
fs 44.2 Hz
Qms 8.51
Qes 0.77
Qts 0.71
Le 0.89
Vas 55.8 Liter

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Low budget choke for single ended class A amp

Hi, I’ve seen many suggest to use microwave oven transformer as single-ended class-A mosfet amp.

Do you have suggestions on the characteristics that I need to look at?

What power can be expected to apply and which bandwidth?

I want to build a SE of around 20 Wrms o test one idea I has time ago, and then build a 100W version.

Expected efficiency is around 40%.

Thank you in advance

ATMOS volume controller

Good day,

I am dyving into ATMOS (7.1.4 in my case) and are struggling through the whole process. As a live time sound engineering hobby, now being at my pension age, I am building my home studio to do audio mixing, having around 4000 studio recording multitracks as material to work with.
After a lot of reading and watching Youtube on ATMOS and immersive mixing for days, I now am at the stage where:

  • I run everything on a Mac mini Studio M2 Pro which has sufficient horsepower to do it all
  • I use both Ableton Live and ProTools (Presonus Studio One as the easy alternative) to turn stereo mixes into ATMOS
  • I have a Motu 16A as my speaker interface (line level signals to active speakers with input via USB from my Mac mini).
  • I have 7 Genelec 1030A (bought second hand) for "the bed"
  • I have 1 Genelec 7050CPM as the "LFE"
  • 4 Top speaker as a DIY, they run passive with 1 Faital Pro 4" full range driver, light weight to mount to the ceiling, just 1 cable per cabinet.
  • I have SSL UF1, UF8 and UC1 to work with Ableton Live and ProTools almost without needing a mouse, which is VERY nice. A mouse is a bit of a stupid device to turn knobs (volume, EQ, panning, etc.), so these are hardware controllers for software (plugins).
  • I can run ATMOS "in the box" as they call it, so all on the Mac mini Studio with Ableton Live or ProTools / Studio One (to be decided)
  • I am the still happy owner of a Mackie 8-bus 48 channel analog mixing desk. It gets its input from D/A convertors from the Mac mini. It is setup to do ATMOS (12 separate output signals back to the ATMOS renderer) based on the ideas and setup of an audio engineer called Bob Clearmountain, very famous guy who is in immersive / ATMOS mixing for years and has produced loads of records, a lot of them in ATMOS. The only draw back on the analog console is that "flying around objects" is a bit of a night mare. Ohhh well, music mixing for ATMOS does not really call for massive object fouling around. When you hear a mix where a synthesizer is swirling like a bee around your head for minutes kind of makes you stop listening. So no game breaker here.

- As a last one, I have Personus Central Station monitoring controller, very nice and not costing a fortune. It enables me to do stereo monitoring.

All that's OK, but I hope you can see it coming: HOW TO MONITOR 7.1.4 ...?? I can do that using a dedicated channel in Ableton live and ProTools, using a mouse. Don't like that, it works, but using that one "volume control" channel becomes a bit of an issue. When I have a lot of tracks to mix (between 24 and 48 is kind of normal for me) go figure where that volume track is, go there and use the mouse to turn the volume to desired level. You get the picture..

Of course there are ATMOS speaker monitoring solutions, where the cheapest, and very good one, is Audient Oria that does it all, it still sets me back some 3000 euro again and I am running out of money here. The other solutions are from 4000 to 15000 euro, just to manage the sound level of an ATMOS mix, kind of stupidly expensive volume knob..... Personally I think that is ridiculous to keep it polite.

Can some one help me with a hardware volume controller that can do both Stereo (which is a subset of the ATMOS speaker setup (just Left & Right) and ATMOS (from 5.1.2 to 7.1.4 in my case). Of course would be nice to have a mute and dim function.

To be a bit more specific:

  • Inputs for the volume controller are the outputs of the Motu 16A that has line level output for the Genelec 1030A active speakers and the Genelec 7050CPM LFE, also active.
  • Outputs are 12 line level signals for each of the 7.1.4 channels.
  • Just ONE rotary knob
  • A mute button (on / off)
  • A dim button, fixed to a set level of gain reduction.

I am OK with electronics a bit, so if needed soldering components on an experimental print circuit board is no issue.

I am OK to pay you if you can provide me with a working solution, that is simple, works at all times, requires no maintenance or configuring and uses easy to get components. In other words rock solid an does what it should. So I can just grab the volume knob and turn to my liking in seconds with the mute / dim function to help me out when for what ever reason those 12 speakers start blasting at full level in my nice bedroom sized studio without getting hearing damage, AND recovering from a heart attack......... 🤐

Preferred NO added noice by this device, not audible at least.

ANY one wanted to give it a go / already knows such a DIY ATMOS volume controller?

Please help me out, very much appreciated.

kind regards,

Frans.

Capacitor choice

Hi
I am currently rebuilding a Rotel RB 870 BX power amplifier
there are 3 capacitors i would like to change to a better spec labeled A B C on the schematic
I have attached the schematic and a pic of the original and the possible replacement parts
which would be the best to use ?
Regards Glenn

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NJM8830 – A New Challenger in the Hi-Fi Op-Amp Arena

Hi all,

I finally had the chance to evaluate the NJM8830 and compare it with some well-established operational amplifiers like the OPA1612A and OPA1642A, which have long been considered golden standards in the audio domain for their performance and cost-effectiveness—covering both bipolar and FET input segments.

The NJM8830 is a compact, rail-to-rail output op-amp designed to operate from 4V to 10.5V (±2.0 to ±5.25 ), delivering up to 100mA (150mA max) of output current even at low impedances. It boasts impressive specifications for its class, including an ultra-low THD+N of 0.000012% at 3.3V, an input voltage noise density of just 2.5nV/√Hz @1kHz, a wide gain-bandwidth product of 90MHz, and a slew rate of 30V/µs.

I received the NJM8830 some time ago, but due to intense research obligations, my audio hobby had been sidelined for the past two years. For this test, I configured a two-stage +37 dB preamp (using both op-amps from the same IC in series for a single channel), terminated with 50 ohms at the input, to accurately measure the amplifier's intrinsic noise floor.

Other op-amps such as the NE5532A, LM4562NA, and the OPA1652/1656/1662 series performed notably worse in this context, so I chose not to include them in this comparison.

What stands out immediately is that the NJM8830 clearly outperforms the OPA1642A across the entire range, although it does not ultimately surpass the OPA1612A—which, to be fair, was expected.

NJM8830 is represented with a green trace, while OPA1612A and OPA1642A are shown in red.

p.s. In a separate test focusing on transient response and settling time, I applied a 40 mV(p-p) square wave (@1kHz, Tr/Tf <10ns) input with a single-stage, 13 dB gain configuration (one op-amp per channel). The NJM8830 demonstrated comparable rise and fall times but showed significantly reduced overshoot and faster stabilization. This implies lower ringing, suggesting that its internal design provides tangible improvements in responsiveness and stability over previous generation devices.

Additionally, to better illustrate this, here is an oscilloscope capture showing the response to a 50 mV input signal. The light blue trace represents the recorded output of the NJM8830, the dark blue trace corresponds to a previous capture from the OPA1642A, while the yellow “live” trace shows the performance of the OPA1612A in real-time.

Best regards,
Dr. Jovan Ivković

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Dual linear pot recommendations?

Hi everyone,

I want to build an Ambler "tilt" tone control (in Baxandall's improved version), which requires a 10k linear pot. Since I want it to be stereo, I would need a dual gang pot. Looking at various pots available at Mouser, most linears seem to have rather poor matching between the gangs. I believe the best I found was Bourns 91R2A-R22-B15 (16mm conductive plastic). So the question is, is there anything better for similar amount of money? (~14€ a piece)

For Sale 211² components

Hi everyone,

I'm selling a batch of components for DIY construction of the late Bruno Plouvier's 211² amplifier. Amplifier inspired by Sakuma San and also designed with the help of Bruno's friend Jean Hiraga.

Included in the sale are:

  • Custom-wound power supply transformers by Étienne Tessier
  • Custom-wound self by Étienne Tessier
  • Lundhal coupling transformers
  • Degawa power supply components
  • Various pre-assembled modules
  • Resistors, ampmeters, heatsinks, capacitors
  • A few other components not pictured

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IMG_1728.jpg
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IMG_1730.jpg
IMG_1731.jpg
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More details on this project here:

https://unepassionaudiophile.fr/laventure-du-set-211²/


I'd like $3800 USD for it, the same price I paid for the components five years ago. A possible exchange for a Nagra IV-S and the remaining balance.

Thanks,

Sébastien

Using 4612 JBL drivers in new cabinet. Firing one driver rearward

I am considering making new cabinets for some very knocked around JBL 4612Bs those are the Baby cheek 2404 tweeter with a pair of 2118H 8" 16 ohm drivers in series.

My use is home theatre/ music and my idea for the cabinet is to make it taller but significantly narrower by firing one driver out the back and one forward. Crossover will be Marchand Electronic 4th order.

This would result in an easier to live with cabinet and I am not certain how the dipolar nature will sound. Will I need a rear firing tweeter too? I will be integrating these with a pair of Subs that I am yet to source.

How will this work? Thoughts?
Regards Onslo

Replace an OPA1678 with an OPA1642 or an OPA1656

Hello everyone,
I'd like to share with you an experience replacing one AOP with another.
In a DAC, I have a R2R volume control that ends with an OPA1678 AOP.
This 1678 is an entry-level AOP (less than €1 at Mouser) with a JFET input.
I replaced it with an OPA 1642, a JFET with higher performance on paper.
The result was less good, particularly with less detail.

In my test piece, there was a spectator applauding in the very front row. With the 1678, I could clearly hear it stand out from the other applause. With the 1642, it was barely noticeable over the other applause.

I replaced the OPA1642 with an OPA1656. The result was the same as with the 1642.

So I went back to the original 1678 and rediscovered the spectator in the front row (and all the other details, of course).

My electronics skills aren't developed enough to provide a clear explanation, and I don't have the circuit diagram.

But it proves that a component, even a cheap one, when well implemented, can achieve high performance.

OPA1678/9 is it really "not a good choice."?

In https://audioxpress.com/article/mod...-part-2-a-few-more-bjt-types-and-jfet-devices it is stated that the OPA1678 family is inferior to NE5532.
But is it really for DIY?
Had a go with a 2,5kHz 4th order buffered LP and HP filter, sallen key on EASYEDA. 1-2k feedback resistors. All melf and NP0, about 0.1$ a piece.
Really good measuring results with a ADCiso. Really low spureous, but of course some, about -110 dB, thermal and voltage noise.
"Cheap" 1mV noise +-15volt power supply cirquit.
The cirquit showed really at it best at 1 volt. Just noise left.
To me it looks like the drive capability and UHF attuantion that TI advertice really are functional?
Saying that because of the flat noisefloor and really low spurea frequency peaks.
Sallen key will be circuit c or d, dependent on frequency. Unity gain negative feedback + positive feedback filter

Contemplating "high end" fullrange build (Cube audio, AER etc.) - speaker selection, looking for cabinet options

Good day to you all,

I’ve done some research and a lot of lurking, wanted to do my homework before asking for qualified opinions … There are other forums out there, but this one seems to have more informed members …

I recently acquired a Sibelius clones, and the sound quality quite impressed me. I initially embarked on wanting to build an SB acoustics Sasandu with upgraded filters components, but the sound of these clones changed my mind.

Considering the cost of actual Sibelius and the cost of what goes into it (speaker cost and cabinet), and the fact that the price of these units has shot up dramatically, I was wondering if I can do (a lot) better. As I'm abroad, I'm also in a position to have cabinets made professionally for a low cost. My max budget for drivers is around 9-10K EUR (the present cost of a Sibelius).

My criterion for "good" speakers is hearing the same thing from speakers which I hear at a concert (mostly classical with a preference for piano concertos) - I'm well aware there is a dependency on sources there, my preference for sources is quality streaming.I have a Ambrahamsen V20 UP but will probably upgrade to the Eversolo Amp-F10 or an Advance Paris amp (model to be determined). My Cambridge Audio CXN I will probably upgrade to a an Eversolo DMPA10.

I’m far from a speaker building expert, so please indulge me if I’ve misread some information.

I've ruled out Voxativ because their prices are exorbitant. I've also ruled out electromagnetic speakers (Supravox, EMS) as overall opinion seems to be that "normal" speakers work just fine.

This has left me with the following short list, from “cheap” to “expensive”:

1. Supravox 215 2000 1519 EUR/unit
https://loudspeakerdatabase.com/SUPRAVOX/215_EXC

Nothing below 200, dip around 8K. Presumed to require a subwoofer.

2. Cube audio F10 Neo V2, 2400 EUR/unit

This is the start of a different price range - quite pricey.

https://www.cubeaudio.eu/cube-audio-f10-neo

Fairly good at the low end, high end dips a bit. Variations seem to be up to 20 dB (diagram is hard to read), things go up and down quite a bit.
In the forums these speaker have mixed reviews …

3. AER AER BD3 and BD3B (pair) for around 9500 EUR, BD2 and BD2B (pair) around 7200.

Different price range but narrowly within the budget ..

https://aer-loudspeakers.com/wp-content/uploads/2018/12/AER-Breitbandtreiber-BD-3-BD-3B.pdf

https://aer-loudspeakers.com/wp-content/uploads/2018/12/AER-Breitbandtreiber-BD-2-BD-2B.pdf

Pretty flat response, better in the lower frequencies (the B models are said to be better for this). Variations seem to be within the 10 dB range.

The “Jazz” cabinet looks nice, but I can’t find cabinet details - and how the transparent horn can be built/obtained is unkown.

So, with the goal of doing (a lot) better than Sibelius, my questions are:

1. Which speaker do you recommend I go for, and why.
2. Which cabinet design can you recommend/provide (I require a detailed plan so I can have it built) - in the case of AER I’m also interested in recommendations on how to build/obtain the horn.
3. Several speaker builders recommend woofers (including AER!). I’m therefore interested in woofer recommendations for the days I listen to 60s - 80s pop and disco (speaker, cabinet plans, plate amp recommendations - e.g. Hypex, Dayton, MiniDSP, Monacor …).

Thanks in advance,

Gunn
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Fully 21st Century Tube Preamplifier

All,

Here is the initial post (more will follow) for a project I wish to share that I have been working on for a while (hence my absence from this forum). It is a vacuum tube preamplifier that essentially marries 1950s technology with that of the 21st century. Here is a list of its primary features:

1. Bluetooth input and audio streaming capability.
2. Smartphone control with Bluetooth.
3. Dot matrix RGB display for digital volume control and menu screens.
4. Protection mode for output overload, open tube heaters, and power loss.
5. Software-defined programmable gain (default gain is 12dBV).
6. Tape input, tape monitor loop, and buffered tape output.
7. Four 20k ohm line level (-10dBV) inputs.
8. Switchable high-impedance buffer (used for high-impedance sources such as ceramic cartridges) makes any line input 1M ohm.
9. 45dBV 47k input impedance RIAA magnetic phono input.
10. Two switchable 6922 direct-coupled line-level outputs can drive 600 ohm loads with 0.01% or less THD. 6N1P tubes produce slightly more distortion.

Audio circuitry:
1. All tube circuitry in audio path for line and phono stages.
2. Bluetooth audio signal processing, tape output buffer, and high-z buffer circuits are solid-state.
3. Four 6C45P tubes are used for phono and four 6N1P or 6922 for line.

The software is not complete. I have at least a month of C coding ahead of me to get the Bluetooth streaming/pairing functionality (along with some bug fixes) to work correctly.

Before anyone panics, I must note that the digital circuitry, display, and Bluetooth radio cause no measureable or audably noticable interference (even with the phono input at full volume without the input shorted) in this design.

Attached are several pictures of the unit. What do you all think? Once I see some initial opinions, I will post some test results.

Thanks

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Altec Lansing H110 (1940s A5) cabinet plans

Hello,

I’m trying to find the blueprints of the original H110 cabinet on net, without any luck…

Does anyone have the plans for the H110 cabinet?

It is an early model of the Altec A5, REF: attached screenshot.

Considering building one.

—I already have the 515 RWB Hollywood woofer and a 288 Gold labeled driver with original diaphragm, and a tar-filled 1005 multi-cell….

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Younger brother of the LM3886 - LM2876 in parallel

Hi all,

Because LM3886 is nowhere to find I got two LM2876s and paralleled them. These are the same Overture series as LM3886 and sound equally good, except not as powerful.

I was able to get 40W 50W into 8ohm and 60W 76W into 4ohm with 24V-0V-24V transformer secondaries.

Updated measurements:

Power supply
24V-0V-24V AC secondaries
33.4V DC at idle

Gain: 25.8dB
SNR: 101.39dB
Sensitivity: 0.835mV

8ohm
19.9Vrms max output
49W into 8 ohms
THD+N: 0.0011% or -98.96dB at 1kHz
THD+N: 0.0023% or -92.58dB at 10kHz
Heatsink barely got hot.

4ohm
17.5 Vrms max output
76W into 4 ohms
THD+N: 0.0024% or -92.4dB at 1kHz
THD+N: 0.0058% or -84.6dB at 10kHz
Heatsink got warm.

4ohm 5W output
THD+N: 0.0013% or -97.61dB at 1kHz

Listening tests:
This board sounds amazing!
Great resolution. Clear highs, warm mids, and powerful low end.
Had no problem playing low sensitivity ELAC speakers.
Very dynamic sound. At max volume got so loud!
I would say this chip sounds 95% of the LM3886.

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Sub preamp/filter - Am I on the right track?

I'm designing my own active subwoofer. This is the preamp/filter. It looks OK in the sim. Am I doing anything terribly wrong?

The first stage is just a preamp and a level control. Then there's a lowpass filter. If r_cutoff=22k, I get about 100Hz corner frequency. I'm thinking of using a 50k stereo pot for R33 and R32. Next is a selectable inverter. Nothing fancy here. R49 is really a switch. Finally an allpass filter to allow the user to adjust the phase. The transistors and zeners are just part of a voltage dropper.

Anything extraordinarily idiotic that stands out?
1748743109336.png

ICEpower 50ASX2 or 125ASX2, and input buffer

I would like to try an ICEpower amplifier.
My speakers are MarkAudio fullrange, and I listen at low-to-moderate volume levels.
My DAC is a Quanghao ES9018 with Lundahl LL1684 transformer output stage, which is pin-configurable for unbalanced or balanced output. This DAC also has software-controlled attenuation.

Q1. Even if I don't need the extra power of the 125ASX2, is it sonically any better than the 50ASX2?

Q2. Although I could feed balanced outputs from my DAC to dual BTL ICEpower amps, I'm thinking just to go unbalanced into a single amp board, unless anyone can say that dual BTL amps will sound better?

3. Compatibility of DAC output with ICEpower input:
I understand the 50ASX2 and 125ASX2 have relatively low input impedance - 8k or so.
My DAC's output transformer apparently does not like being connected to low impedance amps. But the DAC's circuit board has provision for an optional R-C network at the transformer secondaries, intended to suitably dampen the transformer's resonance in cases such as this.

Q3a. will the addition of this R-C network be a (good) solution to connecting my DAC directly to a 50ASX2 or 125ASX2 ?

Q3b. or should I use a buffer between the DAC and ICEpower? If op-amp buffer, are we talking about the often-discussed NE5532 / LME49860 / OPA1622 / MUSES02 etc, or some other?

Q3c. or a solid state buffer, such as -
JFET cascode buffer
JFET input Cascode buffer board assembled ! | eBay
or diamond buffer
Diamond buffer board stereo broadband strong driving capability assembled ! | eBay
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