Intro

Hello everyone! Mi name is Enrique Spinelli. I have been working for many years on analog instrumentation circuits, mainly focusing on biopotential amplifiers. Currently, as a hobby, I am starting with low-noise balanced microphone preamps. I don’t have much experience in this area, but I hope to take it through this forum... I believe that some of my circuit designs could be adapted for audio applications. Best regards!

Three Way Speaker Question

Hi I am contemplating committing to a three way speaker design despite my preference for my current two way speakers compared with various compression drivers horn combinations in both two way and three way configuration. The current speaker is comprised of a Beyma 15G40 15" woofer and CP755ND 1.4" compression drivers mounted to wave guides actively crossed crossed at 700hz per channel. I was originally considering supplementing the bass section of the speaker with a second 15" woofer in the hopes of achieving a lower bass response however after some suggestions from forum members I am now considering an alternative design.

These are the three way designs I am considering. Any suggestions are appreciated.

-A Ciare 15.00SW 15" subwoofer driver crossed at 100hz, a single or pair of Beyma 10MI100 10" drivers in an MTM or TMM configuration operating from 100hz to 900hz, and a Beyma CP755ND from 900hz to 20khz
-A Ciare 15.00SW crossed at 100hz, a Beyma 15MI100 crossed from 100hz to 900hz, a CP755ND from 900hz to 20khz
-A Beyma 15G40 crossed at 250hz, a Beyma 15MI100 crossed from 250hz to 900hz, a CP755ND from 900hz to 20khz

The primary question questions are, is there an advantage to using a single 10" driver versus multiple 10" drivers in an MTM or TMM configuration? And is there an advantage to incorporating to a single 10" or multiple 10" drivers for midrange duties, compared with a single 15" driver? Any help is appreciated. Thanks.

How much does cooling fan air volume affect overall heatsink temps?

I wasn’t entirely sure of where to post this thread. I have a stack of Hafler pro amps, and I need to replace a couple of the fans in them. Amps use lateral MOSFETs that are hard to find and very expensive when you do. So I want to make sure that I have enough cooling for them. The fans are not overly loud, but it would be nice to make them a bit lower. I have a bunch of the Yamaha PC-9501n amplifiers, and they are dead quiet.

This is the current fan that is in them, as you can see the one in my hand is broken.

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It is a 5 blade fan that draws 350 mA, I found that is spins at about 2550 rpm. I cannot find a spec for air volume on it. I did find another fan from the same company that has eight blades and has a speed of 2800 rpm. It draws 550 mA and moves a volume of 115 cfm. Clearly the fan in my amps will move less volume, but I don’t know how much less.

I purchased this van last night, it is referred to as one of the best performing fans if you want ultra quiet.
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It’s spins at a speed of 2000 RPM and draws only 140mA. I found it moves a volume of 60 CFM. Do any of you think that this fan will be up to the task of keeping these MOSFETs cool?

You can see there is some space above the original fan. I was reading that there is another fan that may be even better, but it is 30 mm thick and not all scenarios can support that. I ordered a three pack of them, Phantek T-30.

https://www.phanteks.store/products/phanteks-t30-120-fans?variant=39439636856887

It spins at a higher speed of 3000 RPM, draws like 360 mA which is closer to the original and it moves over 100 CFM. I could run it at the lower speed of 2000 RPM and it will move 67 CFM, of course faster spin, speed and air movement equals more noise. So I was hoping to get your input on this, do you think that the Noctua up to the task? If it is, I will buy nine more and all of the Haflers with it. I’m just not sure how many degrees of difference on the heat sink there will be if you cut the air volume movement down. With the original fans, the amp do run warm. The amplifiers are biased to about 100 mA per output device.

Thank you.

Dan

Bose Acoustimass 10 IV Subwoofer No Sound. Help with burnt component identification?

I picked up this used Bose Acoustimass 10 IV speaker system. Speakers seem to work fine, but the subwoofer does not make any noise. So I took apart the unit and found some obviously failed (burnt) components:
  • 1x 3.14 A @ 250V main circuit fuse
  • 1x bridge rectifier GBU601. there's actually two of these on the board, one is obviously burnt and the other does not seem to have any anomalies like shorts. However I figure may as well replace both.
  • 2x diode. This is tricky because the only markings I see on it are: "617 402". No current rating or anything else. How do I figure what to replace this with?
There are also transistors SGSD200 (PNP) and SGSD100 (NPN). Couldn't seem to do the typical diode test on these, maybe due to how they are wired on the board. There's no obvious shorts, but I wouldn't mind replacing these just to be safe. One of these does seem slightly darkened. The part numbers show up as obsolete on Digikey. What do I look for in order to source a suitable replacement?

So plan is to just replace the components listed above, help identifying what diode and transistor to use appreciated. Everything else on the board otherwise looks fine. If there's other thoughts, let me know? Thanks!

See attached picture.

EDIT: I also just discovered that the small SMD resistor below the "O" in Bose may also be blow. See my post #15 below.

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Beyma 12P80ND v2 and Faital Pro 10AK in 3 way open baffle?

Hi!
I'd like to build some 3 way open baffle using Beyma 12P80ND v2 as a mid and Faital Pro 10AK compression driver for the high frequency. For the low frequency I will decide later the driver(s).
Do you think it can be a good choice? Any particular horn in mind? I am looking at Faital Pro LTH 102, looks OK.
Any ideas for a good passive crossover is highly-appreciated.
Thank you!

how to connect BPA300 (balanced LM3886) to input and output?

so I have tested both R+ and R- by wiring them single-ended and they are working fine.

I want to try testing them now in balanced mode. And I have connected...
  • hot pin of the XLR (pin2) to R+ PCB (this XLR cable is connected to the XLR output of DAC. Volume can be controlled in the DAC so I set the volume to -90db)
  • cold pin (pin3) to R- PCB
  • positive terminal of speaker was connected to R- PCB output
  • negative terminal of speaker was connected to R+ PCB output
This connection results in massive hum. Do I need to connect the XLR ground to the PCB ground?

Thanks!

Golden Age Diode Bridge Compressor

Hi,
I’m currently tried to repair a pro audio compressor I bought month or so ago. It’s a Golden Age comp 554, it’s in a 500 series module that’s a copy of a Neve diode bridge compressor. Every works fine on the compressor, except the compression circuit doesn’t engage. Signal runs through it fine and output gain works. I’ve checked all of the caps so far and they are all fine. I’m confused as to whats stopping the circuit from engaging.

Stabiline Voltage Regulator Vintage Tube

I picked this up years ago. Employer wanted to toss it but asked to keep it. It is beast and very heavy > 50lbs. I believe it was used to regulate voltage to scientific instruments. I’m guessing it was from the 50-60s.

Others have told me to scrap it and some have said it could be used to regulate power to a tube amp. Looking for any additional thoughts. Moreover, I can no longer keep it and would like to get it to a new home where it can be appreciated.

See pics for details.

Thanks!

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For Sale SissySIT

Moving on a superbly built and excellent sounding SissySIT, purchased here towards the end of last year.

500va toroidy trafo 230Vac
Teabag PSU board
Thf51 matched pair
400 deep amp 3U modushop
Cinemag CMOQ-4 high nickel trafos


This owes me £925.

Collection only please from Edinburgh, Scotland. I could ship in UK at buyers cost and risk - but please note it will take me a while to source proper packaging.

Thanks

Eastone

What type of capacitor is this? (ThuleIA250B)

Got a multi broken Thule IA250B that I will try to restore. There are several issues with one being the preamp voltage regulators 317/337 that as std have a what looks like 1uF 50Volt capacitors. I have attached a picture and ofc googled but can't determine if its a ceramic or tantalum capacitor . I have the schematic but its not the same version, in the schematic it says 3,3uF electrolytic.

Thanks.

IMG_4502.PNG

Grounding scheme for simple Class D amp project

Not sure whether this is the best section for this topic.

I've put together a simple "integrated" amp for listening to my turntable using an off the shelf Class D board and phono preamp (see links and poor quality photo below). These are both powered from the same 19v laptop supply. I have butchered an input selector switch unit to give RCA inputs and allow other (line) inputs. I have also added a volume pot.

Currently it's not mounted in a proper case, but boards are screwed to a plank of wood, and it's all working absolutely fine with (almost) all grounds connected and no shielding of signal wires. For now I've been connecting the turntable ground wire to the RCA plug grounds which has been effective in cancelling any hum. There's a labelled ground point on the phono preamp board which I've not needed to connect.

I'm happy enough with the performance of this that I've decided to mount it on a proper metal chassis. It sounds better than it has any right to driving some quite decent (and large) speakers. Not high end but definitely HiFi.

I'm trying to plan how best to lay it out and wondering how to approach grounding. I'm not overly worried about obvious unwanted hum as it works fine as is, but I'd like to follow best practice so that it works as well as it can.

Below are all the grounding points:
  • signal and power grounds on 2 circuit boards, plus the turntable ground on the preamp board.
  • PSU -ve
  • RCA socket grounds
  • Potentiometer ground
  • the (aluminium) case, and I would assume I would connect the binding post for turntable ground directly to this.

I'm thinking of 3 basic approaches:
1. Connect all signal grounds and power grounds logically to each other. Ground the TT and PSU and RCA sockets to the case. This is roughly the current scheme, and it works. I could potentially use shielded wiring internally with this approach.
2. Only connect positive signal and power wiring, connect grounds to casework at nearest point
3. Only connect positive signal and power wiring, connect grounds to casework at common point (star) in this case I'd probably keep RCA socket grounds insulated from the case and use a bar (solid copper wire) to tie their grounds together.

It's very possible I'm missing something obvious here. Hope it all makes sense. I realise it's quite a basic project but the aim is to do it right - and squeeze as much performance as possible out of these basic components.

I’d welcome any thoughts or suggestions on what’s likely to be the best way to hook it all up.

https://www.ebay.co.uk/itm/265099568381

https://www.ebay.co.uk/itm/186733669961

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WTB Sonus Faber Cremona Midranges for first series

Hello. I recently bought a pair of faulty Sonus faber cremona at a reasonable price.

The seller adviced me that I should change midrange drivers because they were not working and that is what happens.

I’d like to buy a pair of discounted Sonus Faber Cremona midranges labeled WOF030001.

They look like scan speak revelador series as you can see in the pictures.


Does any of you has a pair for sale?


Alex




Linn Klout

Hi all,
I have a linn klout that the right side amp module isn’t working and unless I unplug the right side it just blows fuses, does anyone have a schematic? Seeing as it’s nearly 30yrs old there’s no support anymore from linn and there are no linn dealers in upstate NY where I now live now (originally from the uk where I purchased my entire linn system).

How to properly ground Amplifier chassis?

Hi, I have a Vibe mono 21k. When i use Digital Meter set to DC, One wire to RCA ground ( it doesn't matter if rca is connected) and next to amp body (hole for mounting screw) i see 0,3v+ When the equipment was not in use, and drop to 0,145v. after listening music, i can see 3volts and above, this is droping to 0,150 volts.
Amplifier have DC Offset, 0,520volt output without signal...
Any tips to Fix this error? This cause: Head unit rca voltage loss, weak sound, distortions. Amplifier does not touch extra ground. I checked another amplifier, Hifonics brx 12000 and problem is this same.
Thanks.
amp2.jpg

QUAD 63 (and later) Delay Line Inductors

There was some interest in another thread about the modern Quad inductors that form a portion of the delay line in the 63's and newer Quad electrostatic loudspeakers.

Strangely, the British electronics in these speakers is quite robust. But over the 24+ years I've been fixing quads, i've seen just about every failure you can have. I've only seen a couple damaged inductors and they were the results of massive overdriving of the speaker. Below is my knowledge of these inductors, although I haven't really dug into them very deeply (both metaphorically and physically) which you'd want to do if you were going to reproduce them.

The inductors are air core units with a second shorting winding that increases the inductance dramatically beyond what a single coil of this size and wire gauge is capable of. The inductance is about 2.7H each if my measurements are correct. The attachment below shows my method and results. (NOTE: The DCR is in Kohms not ohms as noted on the sheet. And I went back and re-measured the inductor resistance and got the same numbers as the first two inductors; I beleive all inductors are identical). I used a function generator, a resistor and a voltmeter to measure the inductor by forming an AC voltage divider. Using the fact that and inductor has an impedance of: Z = 2 * PI * f * L, I was able to calculate L because I know, Z, and f.

http://quadesl.com/diyaudio/inductorCalc.png


As seen in the pictures, the copper coil is sitting in a plastic "bobbin" with three leads which are soldered to the delay line circuit board. The entire bobbin, and actually the entire delay line circuit board is encapsulated in bee's wax which makes these things easy to work on.

Two of the leads are connected via a buss bar inside the bobbin, and the third (middle) lead runs from the center of the coil. The shorting coil is attached to both the left and right attachment points on the buss bar (forming a short through that bar. And the main inductor coil is attached to one side of the buss bar, at the outside of the coil and the other end is attached to the center lead coming from the middle of the coil.

I don't have a feel for how the shorting coil is arranged relative to the main coil. I'd think it would be bifilar wound, but I have never completely unpotted one of these to check it out.

The way I've seen these fail is that the very fine wire breaks near the solder connection at the buss bar. It appears that the wire overheads and melts, but on come specimens the wire looks thinned which doesn't make sense with a pure melting failure mechanism.

To fix them, I melt the wax around the buss bar and in the center of the coil and using a short piece of jumper wire, I re-solder the fine magnet wire to the jumper and then the jumper to the buss bar. I use the jumper wire because without un-potting the entire coil, the wire where it breaks is too short to reach the buss bar.

Here's a couple pictures of the inductor including one through my stereo-microscope eyepiece. Taking a picture with my cell phone through the eyepiece is a real kludge but works reasonably well to document things.


http://quadesl.com/diyaudio/IMG_4075.JPG

http://quadesl.com/diyaudio/IMG_4076.JPG


Sheldon
quadesl.com
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Michaelson & Austin's TVA-10 vs. Papworth's TVA10 (Burgundy Range) - Schematic Differences ?

A friend of me receive a device of this model in not working condition - made around the late 70th.

under
https://www.diyaudio.com/community/threads/michaelson-austin-tva-10.14116
was mentioned various deficiencies in the circuit design of one of the old versions under
http://www.papworthaudio.co.uk/tva10spc.html
(image under http://www.papworthaudio.co.uk/other.html)
- consequence is an unwanted oscillation (stability issues) under certain conditions.

Bevor I start replacements and troubleshooting on his device, I want to know, if there descriptions for upgrades in the circuit design so that this unwanted oscillations is no longer present.

I guess that was taken into account accordingly in the circuit of the current version under
http://www.papworthaudio.co.uk/redrange.html

Who can upload this associated circuit diagram ?
How many different versions are exist at whole of the TVA-10 and which of them shows the above mentioned deficiencies in the circuit design ?
Thanks for any advices in advance.

According the attached images there are a version with two capacitors and a version with four capacitors behind the rectifier
Additional a version without PCB and a version with PCB (main board) for input and driver stage
The version with PCB exist with two different input and driver tube configurations (3x2 and 2x3 lines) and thus two different PCB layouts

As I know, all versions have partridge-transformers and the McIntosh Unity coupled circuit in use - very important for a tight and clean bass character.

Already exist and closed threads concerning the Papworth predecessors (all devices actually integrated amps with volume control on back panel, but without input selector):
https://www.diyaudio.com/community/threads/series-heated-tubes.369465/ (Serial heated tubes)
https://www.diyaudio.com/community/threads/michaelson-austin-save-attempt.281302/ (DC resistances transformer winding)
https://www.diyaudio.com/community/threads/capacitor-choice-for-attached-schematic.95266/ (capacitors)

More URLs
https://www.audiogon.com/listings/michaelson-and-austin-tva-10-el34-amp-tim-de-paravicini
https://www.diyaudio.com/community/threads/partridge-transformers-vs-hammond.245112/
https://www.diyaudio.com/community/threads/partridge-transformers.183269/

Output transformer of TVA-10 operates according this: McIntosh Unity coupled circuit
https://www.mennovanderveen.nl/imag...018-11-09_VDV-2100-CFB__CFBH-applications.pdf
https://www.diyaudio.com/community/threads/unity-coupled-circuit-amplifier.111673/
https://audiokarma.org/forums/index...mcintoshs-unity-coupled-circuit-works.569405/
https://www.jogis-roehrenbude.de/Leserbriefe/Alexander-Linden-EL84-Amp/Amp.htm

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For Sale Bruel & Kjaer - Vintage Microphone Amplifier Type 2603

Brüel & Kjaer - Microphone Amplifier Type 2603

From the late 60s!

The 2603 is about 55 years old, it dates back to 1966~1969.

This preamp needs some care, but the overall condition is good, and fair given its age.

Was planning to restore it but never got to it so I'm letting it go.

ASKING PRICE € 1.250,00 + shipping

There are 6 or more tubes in the circuitry, and as far as I can tell most, if not all of these tubes are still available in case you need a replacement one day.

I have no idea of the condition of these tubes. Neither do I know if the capacitors need replacement. I never switched it on; it was sitting in my music studio as an eye catcher, waiting to be restored one day. All parts are there. And the unit looks well preserved, which is a good indicator.

  • It could be non functioning, and require quite some work;
  • It could require just a minimal check up and of you go.

I can give no guarantees. The unit is sold as is.

To know how it sounds, look on Youtube for:"Brüel & Kjaer Type 2603 vacuum tube preamp"

I collected several PDF's of this device and line of equipment, including the schematics etc.
I will happily send them to you (saves you some research)

Please note: this preamp does not work with regular XLR mics - although you could probably mod it to do so.

It does seem to work with line signals as a DI, as you can see from the linked video.

It accepts the following mic capsules from B&K: 4131, 4132, 4133, 4134 ..
Combine the above capsules with a B&K "Cathode Follower" type 2612, 2613, 2614, or 2615.
The above parts are available on Reverb and through other places and sellers.

---

The inputs and outputs require a type of B&K plug that is hard to find. Luckily these connections accept simple banana plugs too, which fit like a glove. The ins and outs are basically (unbalanced) line signals. One could easily fashion a set of adapter cables that go from banana plugs (male) to 1/4" jack (female) for day to day use.

---

Connections are as follows:

  • "Amplifier input" = the main line input, which accepts a banana plug
  • The small hole underneath this and other plugs is ground; accepting a banana plug as well
  • "External Filter Input" = actually an output !! Clean signal from this out
  • "Recorder" = allegedly this is a more fuzzy / slightly distorted output

---

The overall gain is adjustable by the "Input Potentiometer" and "Meter Range" switch.

---

The power connector is a Nema 5-15R.
Adapters are available to regular IEC etc.
Alternatively you can connect a power cord directly using the screws adjacent to the inlet pins.

---

My location is Belgium. Items are shipped safe and secure with tracking etc.

Please contact me for a shipping quote for your specific location.
I'll be happy to make you a personal quote.


Have a nice day!

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For Sale High Quality Speaker cables, with Viablue SC-4 cable and Speakon - 1 pair 1,5 m

Selling this excellent pair of Speaker Cables.

Custom built.

The cables are new, never used.

1,5 meters each.

With Viablue SC-4 speaker cable.

Comes with Neutrik Speakon plugs - these can be removed if you prefer open cable ends.

Including one handy Velcro binder per cable.

Price for the pair: € 70,00 + shipping.

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Added mass method for VAS in DATS, differing results

I recently did some measurements on some B&W 8" drivers from one of their subwoofers. Using the added mass method I kept getting the message stating I hadn't added enough mass to lower Fs enough. I eventually plonked on a tape measure on top of the blue tac that I had run out of. I think this gave me an added mass of about 360g.
Voila, VAS measured and came out at 9.2 with an MMS of 101g.

I forgot to save this so I redid it a day or so later. Didn't add as much mass this time, can't remember exactly how much, but it have me a VAS of approx 12 and Mms in the 70g region.

Quite a change from my initial attemp and makes for different results in WinIsd obviously.

What do I need to for repeatable or more accurate results?

RemR P93 1969 Lithuanian Made Turntable

Well. Here is the inside of a 1969 Soviet era Turntable. It's a doozy😜
Apart from the unidentifiable cylinder that looks like it's been at the bottom of the ocean since 1969, the tone arm that seems to suddenly have a ton of tracking force, there is an unidentifiable contact switch hanging loose.
It's a brick.
Oh the turntable turns......kind of.
But it looks soooooo cool.
I could make this a quiz?

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XTZ A100D3 shutting down after power up.

Hi all,

I acquired this amp three months ago, but with life getting in the way, I literally only powered it up last Monday afternoon. Essentially it runs through a start up procedure, the screen works, a fair bit of relay clicking and then it switches off. My own fault for not checking it sooner. I’m half decent with investigating and repairing, and even managed to get the schematics from XTZ. A lot of Chinese though and could be clearer. One 4700uf 63v cap on the power supply board had leaked, actually leaving a leg on the board. Lots of nasty glue everywhere but I’ve cleaned that away, checked the four main 12000uf caps, and they test within spec. I’ve replaced both 4700uf caps, sorted out a couple if dry joints and thought that obviously bad cap could be the issue. No. Found more dry joints on various voltage regulators and elsewhere. Still no joy. There is no response to any control input apart from power on. The overall build of this thing makes it worth the effort, but if anyone has any ideas they would be much appreciated. I’ve never seen an amp with so many screws, it’s depressing!

thanks
Mat

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Old member of the site, but haven't introduced myself

Hi guys.
I have been a member for a very long time of this site, but haven't got into the introduction of myself.
Anyway, I reside in Denmark close to Copenhagen and enjoy good clean analog audio.
My system consists of a XTZ Class A-100, Dali Mentor 6 speaker and my precious Sony CDP-XA555ES.

Kind regards
Henrik Damtoft
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3 way my version

Hello,
Mostly ignorant of speakers rules ,white hair.
In the picture is what i build for home use.
Three separate boxes because this would allow new variants.
They need to be glued, but I'm not sure even if it doesn't sound bad .

Speakers Ciare HWB200-8 ,Visaton w130s,Monacor DT25N.
Simple filter (first and second order (?)) designed with some help with Xsim and FPGraphTracer.
Bass box internal dimensions 61.6x25x35cm, Vnet=51.5L(WinISD), port Dv7.2cm Lv15cm double front panel and two transverse reinforcements.

Please ignore the high standards you have for your works and give me some tips .
What can be done better,bass box dimensions ratio ,port and speaker location...
Thank you and sorry for any mistakes, it's google translate.

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The onboard bass preamp thread

Over the years I've analized / modded / built a number of onboard bass preamps, for some of which there isn't much info out there, and I thought it would be nice to share all the info I have in a single thread. As I go along I'll update this with an index to the relevant posts. I'll also discuss some general topics such as opamp choice (post #2, post #27 and some suggestions for SMD opamps by @Passinwind in post #23) and pickup loading (post #3). These are some of the preamps I'll cover, in no particular order:

- A DIY 2-band Baxandall (post #4) I designed and made for a friend's Warmoth Jazz Bass

- Mighty Mite MM114 (post #6, no longer produced, see page 3 of this catalogue), it came in a Tobias Toby Deluxe 5 I bought back in 2001. An opamp-based, 2-band Baxandall eq with a couple of unusual twists

- Seymour Duncan STC-2P (post #8) and STC-2ASB (the Steve Bailey version, post #9), used in two Warmoth builds. Both are 2-band eq's that use opamps and transistor-based gyrators that lend themselves to many interesting mods (post #12 ff)
Edit 4-Nov-24: There was a small mistake in the previous posts, corrected schematics and sim files in post #10, plus PCB component layout and wiring diagram.

- An analysis of the 2-band Stingray preamp (post #16) and a DIY version based on it (post #17), significantly modded to get a more uniform response and to be able to use standard, easy to find pots

- Info about the 3-band Stingray preamp (post #20) for completeness (I have no experience with it) and possible mods (post #28)

Comments and additions always welcome.

Edit: the attached onboardbass.txt file contains the opamp and transistor models required for some of the simulation files posted below.

Attachments

Hi, I'm a newbie from Sydney, Australia

It would be greatly appreciated if someone could tell me whether the following well-known audio site from Phoenix AZ is still operating?

alan m. kafton / audio excellence az, inc
940 east cavalier drive
phoenix, AZ 85014-1912 USA
602-277-3737


I've been trying to order his Ground Breaker Adaptor Set for my Audiodharma Cable Cooker but have yet to receive any response despite three emails
(https://www.thecablecooker.com/order/ ) and unable to get through by phone.


Many thanks.

Has anyone built a power conditioner?

I'm wondering if anyone here has actually built/designed a power conditioner. I'm asking because I suspect my main systems have less-than-optimal conditions due to switching power supplies, class D amps, and computers being part of the system.

The prices I see on the ones which look like good quality are sky high. Are they terribly complicated to build?

Removing Loudspeaker Group Delay using reverse-IIR filtering

I'm starting this thread to talk about an interesting and new method for "removing" group delay from a continuous audio signal using reverse-IIR filtering.

Background: When you perform "analog style" filtering on a signal, that filter does not output all frequencies at the same time. Instead, frequencies emerge from the filter at different times. This is a consequence of what the filter is doing in the frequency domain (e.g. to the frequency response) - the filter automatically produces non-linear phase changes to the output signal. The result is a distortion of the time-domain waveform, because the appearance in time of various passed frequency components has been altered compared to the input. This sort of phase-domain distortion does not in any way change the magnitude of the frequency component (the frequency response) but the signal in the time domain may look badly distorted. This effect is called by different names, including phase distortion and group delay distortion. When the distortion grows large enough, it is possible to perceive a change to the "tone" of the signal. At lower levels it becomes imperceptable/inaudible. in general, the steeper the filter the worse the group delay distortion and this puts some limits on how steep of a filter we can use for a loudspeaker crossover. With FIR filtering, the situation is different. Phase and amplitude of the filter are independent of each other and this means it is possible to choose the phase to be linear (meaning the group delay is flat). Until now, FIR processing was essentially (to the best of my knowledge) the only practical way to implement a filter with a linear phase response.

More Background: Forward-Backward IIR processing. Someone realized that if you had some finite quantity of sampled audio you could process it first in the normal "forward" direction, and then also in the "backwards" direction. The filter's magnitude changes are applied twice. But the changes in the phase of the signal are exactly reversed, undone, and set back to zero in the "reverse" pass of the filter. What is happening is that the time shift of a given frequency that occurs in the forward direction is exactly equal to the time shft from the reverse pass, but these are opposite in their time "direction" and so they cancel. The result is zero phase distortion, zilch, nada! Let's do it says the crowd. Ah, but implementing this is not so simple. In the forward pass the ringing tail of the signal requires some time to "die down" This extends the duration of the original signal by the length of the tail. This tail must also be processed in the reverse pass in order for it to be brought back in time to its original "place". But if you have a continuous signal, things get tricky. There have been attempts at overlap-and-add methods to process chunks of a continuous signal and then sort of stitch them together but this typically leads to end effects and other problems, and the algorithm is a bit messy.

To the rescue and something that I belive is completely novel is the work by Martin Vicanek that is described in his whitepaper "A New Reverse IIR Filtering Algorithm":
https://www.vicanek.de/articles/ReverseIIR.pdf
I will not get into the gory details of the algorithm too much here. However, in the whitepaper Martin describes a way to take the "Infinite Impulse Response" of the IIR filter, truncate it to make it an FIR response, and then reverse the FIR as blocks in order to reverse the poles of the filter. It's frankly some brilliant work that completely eliminates the need for block processing. The only inputs to the algorithm are previous INPUT samples. There is no longer the feedback from output samples because of the re-formulation of the IIR as an FIR (which is then reversed). This allows for continuous "forward only" processing of the audio signal, and this can go on indefinitely. Best of all, the number of operations required for the processing is much less than the equivalent FIR and latency is much less than reverse-IIR block processing methods.

So, how does this help with loudspeaker group delay? While reading Martin's whitepaper I recalled that the group delay from the summed outputs of an Nth order crossover (the sum of the lowpass and highpass filters that make up the crossover) is the same as the group delay from an all-pass filter of order N/2. An example of this is a 4th order Linkwitz-Riley crossover, which is constructed from two second order Butterworth filter (each having the same pole frequency and Q=0.707) run in series. The LR4 has the EXACT same group delay as a second order allpass filter with the same Fc and Q. So, to make a linear phase LR4 crossover we can implement the allpass response, "in reverse" via the RIIR filter, upstream of the crossover to "pre-distort" the phase of the signal so that after it passes through the LR4 and undergoes an "in air" summation the phase remains unchanged.

The applicability of this technique is very broad. This is because we only need to find the group delay response of the loudspeaker that results from the crossover filters and other sources of delay, construct approximately the same group delay response using only all-pass filters, and then apply those all-pass filters in reverse using the RIIR filter. It doesn't matter what order the crossover may be or what the source of the group delay, as long as you can model or measure it you can cancel most or all of it depending on how close a match you can get via allpass filters. My recent effort to do some group delaey audibility testing, as well as what has been repoted in the literature about group delay, indicates that group delay does not need to be zero but only below some threshold to remain inaudible because human hearing is just not all that sensitive to it. But what (to me) is interesting as someone interested in new frontiers in crossover design, is that filters that previous had too high of a group delay response because they were "very steep" can now be corrected to have as low of a group delay as you wish. This starts to blur the lines between IIR and FIR processing and what each type can accompllish, and I find that pretty interesting stuff.

So, are there any downsides? Well, not really but there is a consequence to the RIIR algorithm: latency. When I described how the algorithm works, above, I mentioned that the IIR impulse response is truncated to become an FIR filter. This truncation introduces error because you are at some point setting the remaining part of the impulse "tail" to zero. This is a source of error or noise in the output signal. The more of the tail you retain, the lower is the added "noise" signal. Luckily Martin provides a way to determine how much of the tail needs to be retained when given some signal to noise ratio that is desired. Also, because the tails from poles located at lower frequencies last longer than for higher frequencies, RIIR filtering for filters with lower pole frequencies requires increasing latency. But luckily the tail decays exponentially, and so the latency is not too severe. In a test trial I ran a 1kHz 2nd order allpass as an RIIR filter. This required 128 samples of latency at 96kHz, which is a latency of 1.33 milliseconds. It's a small price to pay indeed.

I have created LADSPA plugins that implement the RIIR algorithm for first and second order allpass filters. These can be used for phase and group delay equalization and linearization. As of 27SEPT2024 these are available for download at my LADSPA page:
http://audio.claub.net/LADSPA-plugins.html

I will post some follow up information including some more specific examples soon. For now I thought I would put this out there to start a discussion and raise awareness of this processing algorithm.

Hello from Australia

Hello to all. My name is Koulle. I have been tasked with providing a plan to turn some Volvo Dynaudio 4ohm car speakers into "world class" bookshelf speakers. I have been reading J. D'Appolito's "Testing Loudspeakers", wondering if I could find a handle to create a flow chart to the dark art of speaker building with ones own drivers. Can't get off first base. So here I am posting a short intro and hoping that I may borrow valuable information from you. So, thank you in advance. Suggestions as where to begin in terms of where to post my first question would be appreciated.

Peavey FH-1s

I just picked up a pair, also got the tops. Used to have the SP1 variant of these years ago.

I’ve seen some posts from people stating the Klipsch 33, 43 or the Emminence 15c is the way to go in these bins.

I’m using these in my workshop with the occasional yard parties with a pair of Dynaco Mark 3’s.

Does anyone have any experience with these and can recommend a driver for them as well as a complimenting horn driver(s) for a passive 2 or 3 way setup?

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FS: Ian Canada McFIFO/McDual XO, WM8804 I2S to SPDIF/AES, AK4118 SPDIF to I2S, OPA1612

I originally posted this at ASR -> https://www.audiosciencereview.com/...23-i2s-dacs-diyinhk-es9016-dac-opa1612.47265/ but figured I might have more luck here given the nature of these items. I've also lowered all the prices from the original listing. All prices are shipped to US.

1) Ian Canada McFIFO / McDual XO - $150 shipped
Originally purchased for use as a clock buffer for a miniSHARC based 8 channel DSP AES output project. It accepts multichannel I2S and will buffer / reclock input data using oscillators on the McDual XO. In addition it provides 4 MCLK, BCLK and LRCLK outputs which makes interfacing with stereo I2S input boards very easy. Comes with the stock oscillators and will include 24.576 MHz and 49.152 MHz NDK SDAs on DIP14 adapters. If you are interested in interfacing with the I2S to SPDIF/AES boards I have in this listing and I can thrown in some U.FL to pin adapters that work great.

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2) WM8804 I2S to BNC SPDIF Output Boards - $15 shipped (first one is $15, if you would like more each additional board is $10)
I originally had five of these, I now have three. These convert I2S to SPDIF input and take 5 V power. Made in a form factor that is plug and play with Amanero they also work great for other general projects. They are reasonably sized and work well. If you want to adapter the pin header to U.FL connections these Ian Canada adapters for the Buffalo III DAC work perfectly.

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3) WM8804 I2S to AES Output Boards - $15 shipped (first one is $15, if you would like more each additional board is $10)
I originally had five of these, I now have four. These convert I2S to SPDIF input and take 5 V power. Made in a form factor that is plug and play with Amanero they also work great for other general projects. They are reasonably sized and work well. If you want to adapter the pin header to U.FL connections these Ian Canada adapters for the Buffalo III DAC work perfectly.

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4) AK4118 SPDIF to I2S Board - $20 shipped
This board takes TOSLINK or SPDIF and outputs I2S. It has been discontinued for sometime due to the AKM fire but is a great way to add SPDIF input to I2S devices. It runs on 5 V power.


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5) OPA1612 op amps on DIP 8 adapters - $5 shipped (or buy all of them for $50 shipped)
I have fourteen of these. Not much to say other than these are low noise and distortion and easy to use in DIP 8 sockets.

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Michael

Horn / Waveguide Choice for WAW?

Hey Folks,
I've been hooked on the idea of making a WAW for a while. I've also been heavily inspired by xrk & bushmeisters bookshelf synergy thread and love the look of the results they have gotten from the SB 2.5" fullrange.

As it stands I don't believe I have the technical knowledge or physical skills to create a MEH, so I'd like to buy a horn, print adapters and create a WAW setup to begin with. One day I could move to making a MEH.
I've been considering a number of different fullrange 2"-4" drivers and would like to test multiple over time. I'd start with the SB 2.5" and adapter xrk designed in the bookshelf thread.

Please note, my primary goal from horn/waveguide loading the full range is controlled dispersion. Not increasing the SPL of the driver.

Why a WAW/ Fullranger, instead of a compression driver? I want to cross as low as possible in true WAW/Fullrange philosphy of avoiding crossovers in the main audio spectrum. But I also want more bass than can be found from full rangers (WAW), with better dispersion thus trying to horn load these little guys. Compression drivers that reach 300-400hz and have a good top end are not cheap in the slightest!

I am considering the two following horns after a fair amount of research and would love some external thoughts:

Eighteen Sound XT1464 horn https://www.eighteensound.it/en/products/horn/1-4/0/XT1464
for 60x 50 degree CD with 500hz loading.

RCF HF950 https://en.toutlehautparleur.com/media/catalog/product/datasheet/rcf/HF950.pdf
for 90x 50degree CD with 400hz loading.

Is there a reason you wouldn't choose the HF950 besides size?
Are there any other affordable, CD, low crossed, 1.4" commercially available horns you can think of?
Am I just nuts and should stay with a more traditional WAW design?

The reason I am starting a different thread is that this build isn't constrained by size like xrk & bushmeisters discussion is, it isn't immediately a MEH and I don't want to muddy their waters.

As regards to the overall WAW design, I’d see the primary goals as:
<500hz crossover.
1/4 wavelength distance CtC spacing
Good directivity to 10k+ hz
In room Bass extension to 35hz.

Nice to have goals:
<300hz crossover
85+ db total speaker sensitivity
Passive crossover option - unlikely.
Time aligned
Low THD for both drivers
Flexible in room positioning requirements

Would love to hear your thoughts,

Weird X250.8 Meter Behavior

I just recently noticed a strange behavior on my X250.8 meter. With no input the meter has a small rhythmic "twitch". It literally looks like my amp has a beating heart. In the past the meter has only shown movement when pushing the amp from class A to AB. Does anyone know if this is cause for concern... as I said there is NO input and the wall voltage is a constant 120.6 volts.
Thanks,
Tom

array of micro speakers with shading in ATH4 horn

Hi!

For this project I'll be using 40 pcs of micro speaker (102dB@10cm) spaced at 9.1mm C-t-C to emulate a ribbon.

The "ribbon" will be mounted to the throat of a custom ATH4 horn flare. I will use an 8PR200 for bass.

To improve the vertical wavefront / directivity I will add active shading to each speaker or group of speakers.

Each micro speaker will be driven by its own MAX9718A amplifier.
https://www.mouser.co.uk/ProductDetail/Maxim-Integrated/MAX9718AETB+T?qs=1THa7WoU59FvOsgrnnj7bg==

It will be straight forward to taper the levels vertically as the MAX9718A exposes adjustable gain.

For delay I may try wrapping the appropriate allpass/bandpass around the MAX9718A or alternatively use a few LME49721.

More to follow.


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polars
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horiz at 45deg no shading
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vert at 5deg no shading
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test board
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For Sale YL Acoustic 1800G

Information outside of Japan is scarce, but I believe these were YL’s TOTL high frequency driver / tweeter. Very rare to find one in working condition, let alone a pair.

$2,000 net to me, shipping only in the USA for now.

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NOS transistors shelf life

Hey there,
I cant find a clear answer to this question…. Do transistors (if stored properly) degrade while sitting on the shelf? I often find new transistors to have a too high hfe for repair jobs on old gear. Is it a safe bet to replace them with nos transistors? Or am i better off to buy new production? Im talking silizium transistors - not germanium.

Cheers

Phono stage ground hum, PSU dependent.

Been experimenting with one of these - https://www.ebay.co.uk/itm/265099568381 - it actually sounds pretty reasonable.

A big part of the appeal was it should work with a wide range of PSUs and doesn't need a dual rail supply, I was hoping to power this and a cheap class D amp off the same PSU - I was thinking a generic 18v laptop supply to keep costs down.

I've found that it's hum free with my bench PSU but that it has a slight hum with the laptop supply. Initially I thought this was power supply noise, but using the laptop supply I tried running a wire from ground on the phono board to the (mains earthed) case of the bench supply - the hum went away.

The turntable I'm using is not earthed to the mains either, so with the laptop supply earth is floating.

Any ideas on how to get this working quietly without a mains earth? I wonder if a similar switch mode power supply with a 3 pin (earthed) mains would behave differently?

Krell KSA 300-S suicide watch

I'm hoping this gets someones attention. I still an unable to get right channel to come on. If anyone been following this fiasco then they don't need a recap, but for thoses who just checked in I will try to bring you up to date. A couple of weeks ago I decided to take my Krell KSA300-s out of a 3 year storage. This has been stored in a climate controlled building. I'm not the original owner, but I had used it for 5-6 years with no problem.
Anyway I got it set up and no right channel. No lights (led's for that side were on). Left side fine. So, I took it apart and found several issues with a previous repair. I fixed those and turned it on again, nothing. I decided to go in deep and found a couple of other problems so I fixed those.
To try and keep down the total times I took this apart I check every active component on that side. The only thing I haven't check were some timer ic's that appear to be used in the auto-bias circuit. Unfortunately the auto-bias circuit is the only thing I don't have a schematic for (and it appears no one else does outside of Krell who appears to be out of business).
I'm in the dark now and asking anyone that might have some insight to this problem to please come forward. I'm very close to putting it to the side and moving on. So, there you go.
I have attached some schematics that would be helpful.
Thanks,
John

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Order of Operations: Increasing Bass Response in Sealed 2-way Speakers

I have built a pair of towers using vintage drivers I salvaged from beat EPI-400 speakers. They turned out really nice and sound pretty good overall, but bass is a little weak/undefined at lower frequencies. I suspect they could benefit from a little effort at fine tuning them, but I would like to take semi-educated guesses at what might be best to try. Hoping to pick the low hanging fruit first.

Relevant clues that some modifications are necessary:
  • Woofers bounce a at high volume.
  • There is fairly significant cabinet vibration while they're playing.
Some relevant information:
  • These are 6" woofers and they unfortunately do not have any thiele/small information published about them.
  • EPI didn't build any double 6" woofer speakers so I went through some extremely exact math (wink wink) to determine internal volume. I determined the percentage change in volume between a single 8" woofer EPI two-way to a double 8" woofer EPI two-way, and applied the same percentage change to single 6" woofer model from their line-up. End result was a 3600 cu. in. enclosure before bracing, I think ~3400 after. For reference the quadruple 6" woofer EPI-400 these came out of was ~5000 cu. in.
  • The speakers are built out of 3/4" MDF, have internal bracing front-to-back and side-to-side, and are stuffed with fiberglass.
  • The surrounds are original butyl surrounds. They are not that particularly stiff or cracked.
  • Polarity has been double and triple checked
  • I have messed around with placement some as well. They do better against the wall than they did away from the wall (like in the below picture), I've also found they're fairly directional speakers. Bass is definitely improved when you're dead center and near the back wall.
Things I've thought to try, but am not sure which might be most impactful
  • Decrease internal volume
  • Increase quantity of bracing, though would have to get creative with this since their are fully veneered now
  • increase/decrease quantity of stuffing. Amount/compactness of fiberglass is in line with other sealed enclosures I've seen
  • New surrounds
Does anybody have thoughts as to what would be best to try? Thanks!

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Re-wiring arm

I believe someone here has rewired his Graham 1.5 arm. I want to do this with my Graham 1.5T. I understand that to gain access to the wiring, heat from a heat gun must be applied to loosen the epoxy sealing the arm. My question is, what areas of the arm are epoxied? I do see epoxy in the area where the leads exit the arm near the cartridge. Is the arm epoxied near the end where the arm attaches to the mounting? Thanks.

Totem Hawk - Need Help Determining the Inductor Value

Hello. Hoping you more-educated &/or experienced folks can help me out with figuring out what inductor value are in these Totem Hawks. They're not marked, so I disconnected them & took a reading with an LCR tester (see pic), but the reading doesn't make sense. The woofer is supposed to be 6 ohms, the crossover frequency is supposed to be 2500Hz, but these numbers just aren't adding up. I'm sure I'm missing something & could use some guidance. Measurement shown. Please & thank you in advance.
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Strange (?) voltage spike during capacitance multiplier start up

Hi! I have built a MOSFet cap multiplier to filter valve amp B+. Here's the schematic:
Bildschirmfoto 2024-11-13 um 13.23.02.png
Real life DC behaviour on the bench matches the simulation closely.
I have a meter on the load resistor and a meter on the output of the PSU powering it. It is an old unit with an indirectly heated full wave rectifier, so comes up slowly.
The cap multiplier has the typical slow startup from the time it takes for C1 to charge. At the point where that is nearly finished and Vin starts to drop, there is a click sound to be heard from the cap multiplier and Vin jumps up to 1kV for a moment. Unloaded Vout of the psu is 510V, under load 484.
Do any of you have an idea where that spike is coming from and how to suppress it?

Thanks for any useful input.

Speakers for my project

Hi everyone I'm currently in the process of building myself a sound system for my home. I'm currently in the process of choosing what drivers I'm going to use for this build but I'm not entirely sure what drivers to choose.

I plan on having 2 bookshelf speakers for left and right, 2 surround and a subwoofer. For the bookshelf I wasnt sure whether to make a 2 or 3 way speaker. I have already purchased the epique e180he-44 speakers I saw great reviews for this speaker but I'm not to sure what speakers I should pair with this speaker, I've been told the e180he-44 is best in a 2 way but I think a 3 way must be better to achieve better quality of sound and a wider frequency range.

Please could you give me some advice as im pretty new to this I just want some decent drivers to pair with what I have. For the crossover I'm getting one custom made for when bookshelf speakers once they're made.

Thanks

46 driving a 300B

I am in the final stages of building a pair of Monoblock ( LL7903/1:8 to a 46 to a LL1671/30mA to a 300B to LL1664/80mA)
I've got all the PSU dialed in and was reading some of the articles from Ale Moglia of how he switched to the 47 for a bit more gain given his 2v input so wanted to check my gain by the stages.

I have a 4V input from the DAC (SMSL su-10) so the 46 would see ~30V on the grid. I almost went to turn the amp on and realized that the Rk on the 46 is at 17.7V. Should this be 32V ish? should I just do a B- supply to this or do a Rk with a higher value to give me the 32 ish V on the cathode?

Obviously the B+ would have to change to around 240@30mA if using the -30 curve.

With the 4V into the LL7903 in 1:8 it yields the 30V and the mu of the 46 is 5.6 which should translate to 168V swing. The max per the LL1671 on the secondary is 130V rms. Will this configuration drive the 300B too hard and the 1671 too hard as well? I have highly efficient speakers at 96db/w/m so I probably wont be cranking it all the way up anyway but wanted to check everyone's thoughts here.

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2 x Hypex SMPS1200 dead following power cut

We had a power cut last night, once it came back, everything turned on as normal... except my two DIY Hypex monoblocks.

Specifically, both SMPS1200A700 modules have stopped providing any DC power. There is no relay "click", nothing.

- The fuses are fine
- The current inflow thermistors are fine (I had to replace one of these before)
- The main capacitors are charging
- I've tried letting the main capacitors discharge fully before powering on again - no change
- I have checked on other power outlets - no change
- I have tested with and without NCORE modules hooked up

According to the HYPEX data sheet, the SMPS can be latched off if the NCORE module sees a fault, but this should be unlatched after capacitor discharge.

There is also the following in the datasheet, but no more details of how it works:
"The supply is triggered for normal operation or latched off in case of critical fault via in built-in actuators."

Does anyone have any ideas for common faults/diagnosis? The fact they both failed in the same way seems suspicious, and I really don't want to drop >400Eur on two new SMPS if it's just a fault-protection, then wait for the next power cut...

Making a diy USB pre-amp (need help desperately!)

Hello!
I'm still new to audio electronics and learning a lot as I go.

For context, I’m working on building my own USB microphone from scratch as a starting point for more complex personal projects. My current design is inspired by DIY Perks, but I’m designing my own preamp and power supply circuit. I’m working with a tight budget and limited access to specialized or hard-to-find components, so please keep that in mind.

Right now, I'm running into an issue with my power supply section of the circuit.
1731737446148.png


When the power supply is not connected to the rest of the circuit, it provides a stable +15V and -15V output. However, when I connect the preamp to the power supply, the voltage drops drastically to around 1.2-1.8V.

After testing, I discovered that when I directly supply the preamp with ±15V, it works perfectly. This suggests the issue might be related to the Darlington pair circuit.

Blue is micInput, Green is output (Power input connected to pre-amp) Red is output, Green is micInput (Direct ±15V connection)
1731738037185.png
1731738136656.png


Could you help me identify and resolve this problem? If there is anything else, don't be afrain to rag on me for that as well.

Thank you all in advance!

For Sale Tamura A-875 line output transformers

Selling 1x pair of A-875 line output transformers, excellent condition

17K:600 (150 split), 7ma

Originally designed for the 6SN7, but of course can be implemented for any tube that fits the specs, like a 76 for example

Asking 600€, will ship worldwide

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3-way corner speaker?

hey all, in the never ending adventure to both have nice sound and WAF, i'd like to try to build a speaker for corner placement, which will mostly be behind a chair in the corner. This is for mono playback, so there's no second speaker, just one speaker in one corner. I'm currently listening to a linkwitz pluto, but although i like the sound the omni pattern and it works well with mono-playback, it really wants a closer listening position and i don't want to keep dragging the speaker out from the wall every time i want to listen.

The whole thing will use minidsp for crossovers/eq. Primary seating position is about 16 feet away from the corner (see attached poorly drawn room layout). Right now unless the Pluto is close (<9ft away or so), there's too many reflections going on in a bad way (expected but worse than i had envisioned).

so here's what i'm thinking to try to play well with the corner:
highs (>1250hz): B&C DE250-8 compression driver paired with either a HM17-25 or H6512 horn (horn and xover point to be decided upon testing/listening)
mids(300-1250hz): Dynavox LY302F 3.5" tucked right into the corner as close as i can get it
Lows (40-300hz): Dayton RSS210HO 8" woofer

see attached bad sketch of how i would arrange the drivers. view is at a 45deg section of the corner. I neither enjoy nor excel at woodworking, so i was hoping to just use the denovo knockdown cabinet for the 8" woofer. so all i'd have to build is a simple box to house the midrange. Unclear right now if i'd just mount the midrange box onto the wall, or have legs that go down to the woofer box.

I've seen Wayne's PI corner speakers and if i had the space i might just buy that. but they too big to pass the WAF test.

all thoughts, comments, criticisms, better ideas welcome! Thanks!!!!

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Theta DS Pro Basic 3 DAC blows a fuse

Hi,
Don't think I am a DIY person because I am a member here 🙂 I can do "some" stuff but not too much 🙂

So, I bought this DAC from someone knowing it was not powering up. The seller said he had no time to check and I took my chances thinking it could be an easy fix such as a bad fuse. I was like you know what I had a unit with an issue before and a fuse replacement solved the problem. Maybe this will be the same.

I put the appropriate fuse I hope because that's what it had in it, slow blow 3/8A (375mA).

WELL, THE UNIT POWERED UP, I pushed the buttons, they were all working, etc. two minutes later, I saw the fuse burned like red and then puff! The unit back to dead again 🙂

What could be the reason for working few minutes and then burn a fuse?

I have a friend who has all the tools to fix it (if fixable) but I thought let me ask here maybe it would help him pinpoint the problem.
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