I plan to use a hot wire cutter to cut XPS foam to make speaker horn moulds.
Wire size about 0.5 mm, needs ~5 amps.
A 1 metre cutter in nichrome is ~5 to 6 ohms - so ~25 V and 125 watts.
That is easy and reasonably safe with a transformer.
In fact I even made a hot wire cutter similar to this as a one-off project for a friend's factory to cut broadloom plastic cloth.
It was decades back so I don't even remember the power requirements but commercial units are similar to what I calculate so I don't think I'm too far out.
The idea for the new project is an improved circuit that controls the power to reduce idle losses and allow closer tolerances.
It should be not too hard to monitor both the current and volts thru the wire and calculate the resistance, to use in a feedback loop to keep the temperature constant.
Anyone seen a circuit to do this?
The obvious way is a simple micro-controller but maybe a simple all analog circuit would work.
Or other ideas?

David

Since the beginning of the Circlophone project, I had always toyed with the idea of making an ultimate test of flexibility and versatility: create a vintage, full-germanium version.


I finally had the time and opportunity to build one.

• It is an interesting test: would a modern topology like the C have been viable in the early sixties?

At the time, amplifiers were prone to thermal runaway and had to include a whole range of counter-measures like NTC resistors, compensating diodes and emitter resistors.

• An adjustment-free, totally safe amplifier would have been a considerable improvement, but is it actually possible?

The answer is here, it is the "Steamclophone", an amplifier built using exclusively components and techniques available in the early sixties.

Some adaptations had to be made, for mainly two reasons: one is obvious, it is the different material bandgap, the other being the slowness of Ge transistors compared to Si: they are litterally orders of magnitude slower.

This required some redesign of the compensations too.

Another problem is the lack of good NPN transistors: this mandated the use of P-darlingtons for the OP, instead of CFP's.
The bias currents also had to be corrected, because of the low allowable dissipation in the VAS and driver transistors.

The result however is still very much a true Circlophone, completely self-adjusting and thermally safe.
It worked almost immediately, with the quiescent current establishing itself at 190mA, just like its silicon brothers.

Now that I have the thing working, I will measure its characteristics and post some oscillograms....

Have fun!

Quest for the perfect amp! Ordered TPA3255 from Dr mordor 🙂

First of all, thank you for the answers to my beginner questions in a previous thread!



It helped me decide to go for Dr mordor's amplifier TPA3255 which i'm awaiting in anticipation.


Now i'm researching on where to go from there! Hope some of you can help me with this 🙂



So i have an Akurate DS as my DAC/Streamer. This has inbuilt digital volume control, and XLR outputs that i'm going to feed dr mordors amp with.


However, if i want more inputs on this amplifier i presume i have to add an input selector board in the back. And maybe a volume control pot. How do i wire these things up?


And can i buy some of the already premade boards on ebay without messing up the sound quality? 🙂


Would love to add bluetooth as well, but i know this can bring its own problems!

Thanks guys!

I just obtained a PE 3046 turntable that I would like to repair (ca 1974-1978). The PE 3046 turntable features automatic playing, however, I can only get manual playing mode to work. I have obtained the German PE 3046 manual (which I can't read unfortunately) as well as the manual for the similar Dual 1226.



The problem is this: there is a cam in the turntable that activates a black lever (number 155 in the attached photo from the service manual) which both raises and lowers the tone arm, as well as pushes on another lever down and sideways, using friction to move the tonearm horizontally to move it from the tonearm rest to the edge of the record (number 201). Whatever surface treatment was used to provide this friction has worn away and therefore the black lever simply slips when pushing on the lever sideways and the tonearm does not move horizontally automatically.


If I disconnect the return spring, I can get it so that I can manually place the tonearm and play a record, however, I would like this feature to work. What can be done to restore the friction? It feels like some gritty material was used to provide it. I tried to place a piece of rubber between the two but it doesn't work, it seems to require a lot of force.


If anyone has experience repairing such friction mechanisms please let me know.

I'm trying to salvage a bad first build with a bit of a redesign and numbers I'm getting from my volt meter are not matching what I'm being told should be there by the PSUD software. For example, based on measuring the voltage across my 6L6GC's cathode bias resistor (SE design) I am running them at 65ma. But, if I put that value (x2) into the PSUD software, along with appropriate pre-amp tube draw I'm not getting very accurate B+ results. Now, if I drop the draw to something much lower (on the order of 50ma per channel) then the voltages line up spot on with what I'm recording on the volt meter, given the rest of the parameters being accurate.



Seems simple to me that 21.5 volts divided by 330 ohms (small design change from original) equals 65ma but I'm wondering if I'm missing something. The design (attached) plays incredibly quietly at something a bit north of these B+ voltages so worried my bias is off.........then again, could just be those 6sn7s.

Just repaired one side of the PS, IRFZ48N units. All other parts checking out all OK. Put the amp back in the sync, which takes a minutes, and it powers with 1.4A idle draw. ~98vDC rail. Everything else seems to be checking out OK, but no audio.

Power is Green.
Music is Green, I'm not sure if this is right since there is no audio connected.
Protection light blinking red which I understand may be an RCA shield problem.

RCA shields are directly connected to non-bridging speaker terminals.
RCA shields to amp ground is about 14kohms
Powered, there is no DC across speaker terminals.
Powered, referencing amp B- theres 0.74vDC on speaker terminals.

Is there a SM floating around?

Hello all,

I have a pair of Fostex F200a drivers in good condition that had never seen regular duty in an enclosure. They have been used for testing purposes (by the previous owner) and had been in temperature-controlled storage since I bought them 10+ years ago. These are alnico full-range drivers that had been discontinued by Fostex. They were made famous by the enclose and filter design by Bob Brines known as the FTA-2000 which used to be considered one of the very best single driver speakers.

After owning them for over a decade, unused, I might sell them now but have no idea of the value. I hope that you could help me determine what a fair price would be for the pair.

Thanks.

Can somebody explain active servo speakers? Please explain like I'm a four year-old.

In context: I've had a number of Yamaha sw-P130's. As a result I have two working drivers which I intend to use for a separate project. However, I have utilised a P130 plate amp in a separate project. It is connected to random Sony driver. Does the active servo system require the technology to be present in both amp and driver?

anyone have a current recommendation for replacement of the output fet's?

Hi All, I have a Beolab 7.1 speaker that has issues (doesnt play sound even though I get the green led light) I'm thinking as there is 3 x 125w icepower modules inside maybe I can reuse these (hopefully one or two are still ok) and put them in a Ghent case and make a nice little class d amplifier. First up , does anyone if these icepower modules are mono or stereo? I have the speaker open so I can send pictures of the modules. They seem to have a separate power supply PCB, what PCB would I need to give these modules power?
I've built an icepower Ghent case amplifier before but it had the integrated power supply and it was an easy self build. I think build is within my skill set, but just need some direction.
Thanks for letting me send this post and cant wait to here your answers / opinions.

Hi everyone,
I'm an absolute beginner with the idea to build a small portable bluetooth speaker and I'd appreciate some input on my ideas!

These are my tentative specifications:

• very portable: around 1l net volume
• mono (I assume one bigger driver is better than two small ones?)
• 2-way
• vented
• casual listening to music and watching movies
• bluetooth and Aux input
• battery powered
• USB chargeable
• overall budget up to around A$150. Flexible if there is a good reason.

First of all, I've read in other threads that it's difficult to achieve similar sound quality / range to commercial products, as they use DSPs.
Is that true? I would like to get somewhat comparable results from my speaker.

I was thinking a 2-way, with a mid-/fullrange and a tweeter. I assume higher sensitivity is preferable as it's battery powered? Any suggestions for combos?

As far as I know vented speakers can be smaller and can produce lower frequencies. My personal (unqualified) choice would be a passive radiator, mostly because of looks but also because it can be even smaller than ported. I'm happy to use a port as well though, if it brings advantages.

I'd probably use plywood for the enclosure, how thick would it need to be for a small speaker like this?

That's pretty much where I'm at, I'd love to hear your opinions and ideas!!

Thanks in advance!

I've always been intrigued by these speakers, for what they do vs. what they don't do (loud, lots of bass, free of quirks - plenty of 'good' speakers I can buy for that).

I found Falcon sells a kit for about $1,200 USD. I know this is a TON of money if you just compare specs, but I am looking for something fun.
Falcon Q7 | The Ear

An easy choice would be something like the KEF LS50, but I worry I'd find myself a little bored with how...perfect(?) they are. I know that is strange to say.

Anyway, these would be used mostly as my computer speakers for background music as I work, or anything else that needs audio. This will not be my main system, but of course I could try them on my main system.

Using an ACA right now (I appreciate it because at the low levels and close distance, there is ZERO background noise vs. the Rotel integrated I am using as a preamp) and will upgrade the preamp side of things (like a Schitt Asgard with the Multibit).

Thoughts? If I want something flawed but lovable, good at low levels, think I would like these?

Looking for single RS-7 Grille in good condition.

Thank you!

Hey everyone,

I'm currently working on a small 2.1 sound system for a friend, aiming for a flat response and clean overall sound in as small a package as possible.

For the satelites, I've settled on using a pair of FaitalPro 3FE25 Full Range drivers, as they have a relatively flat response and excellent off-axis sound. For the sub, I am using a Dayton Audio DC160-8 6.5" Classic Woofer and a matching 8" Passive Radiator, also from Dayton. This combination will allow me to cross over nice and low, hopefully somewhere around 90-100 Hz.

I've read elsewhere on the forum about the same woofer-PR combination being used in a sub known as the Bitty Boom V.2, which is a 14" length of Sonotube with one element in either end. The gross volume of this enclosure is about 11.5L, and apparently it can reach frequencies as low as 25Hz pretty comfortably with 25g of mass added to the PR.
So my question is, basically, how much of that LF extension is attributable to the shape of the enclosure? All the other designs I've seen have the PRs mounted to a panel adjecent to the woofer, rather than directly in line with the back wave. Is there a known trade-off for this design? Is there one preferable way of doing it? I have no prior experience with building PR-tuned cabinets, but I'd definitely prefer to build either a cube or rectangular design over a tube.

Another side-note question: Can I make up some of the losses in cabinet volume (from the baskets, motor, and onboard amp) by stuffing the box with wool? Or is it preferable to just add more mass to the PR to maintain the tuning?

Any help would be appreciated!

Thanks

I was given power and output transformer, along with an EZ80 rectifier and EL84, and looking to build a small guitar amp.

The heater secondary is measuring 7.8v, and has no center tap.

If I plan on using with the EZ80, EL84, EF86 and a 12AX7, what would be a good voltage measurement without the load? And how can I lower to that voltage?

Ive seen other posts where adding resistors is suggested, but wonder if this would work for dropping 1.5v.

Is it possible to disassemble these transformers and remove windings?

Thank you

This amp had me scratching my head for a while because the bicolor status LED would only glow faintly on both sides with remote applied after I replaced all the FETs, but as it turns out it’s actually making both rails and has good output drive. Guess the lamp circuit needs work, but my question is:

Is it normal for these to change the PS wave shortly after startup?

When I first apply remote I get pic #1, after a couple seconds it changes to #2 which to me looks terrible but the amp seems to function this way and nothing gets hot.

Hi People,
I am kind of wondering about how to best construct a loudspeaker.
Now given that in the end I get some volume for the Box that I need to build. but the problem is, that when I build in the Chassis I loose some volume (Chassis cone and magnet at the minimum)
Is there any tested technique how to best calculate or measure this volume?

kind regards
huegene

For example I'll use the LM3886 which I'm currently looking at the datasheet for. If you feed it 28v, at 8 ohms you get up to 38 watts output before distortion starts rising. If you feed it 35v you get up to 61 watts at 8 ohms before distortion starts rising. The distortion is effectively identical for both voltages from 0 to 3 watts, but between 3 and 38, distortion is almost half the amount using 28 instead of 35 volt power.

Is there an explanation for why this happens? Is this effect exaggerated in the LM3886 compared to other transistor amp chips, or is this how things always are - and the reason variable rail amplifiers exist? I always thought it was Energy Star efficiency requirements or similar

I'm trying to help a friend, who owns an EMM-Labs DCC2 SE and CDSD SE combo, come up with a solution on how to play DSD files from his PC or laptop using his existing (btw. amazingly good sounding) DAC.
The problem is that the DCC2 SE only accepts DSD via EMM-Labs own Optilink glass connection or via SDIF3/Raw-DSD using BNC connectors.
EMM-Labs streamer would be too big of an investment for the few titles he wants to buy that are just available as DSD downloads...that leaves the exD USB DDC as the only option I could find so far and well I didn't manage to find one of those units. Diy options are ofc very welcome.

Please I need your advice diya with this unusual problem😕

I have confused my self regarding the T/S parameters for Isobaric loading. As I understand things ....

In one version, Clamshell, the woofers are on a common baffle and mounted face to face with the drivers wired electrically out of phase (so the air space between the drivers is isobaric). As I understand it, in this case the Vas is reduced by approximately 1/2; Fs remains the same; Bl doubles; and Mms doubles. The port length would also double (e.g., in a 4th order bandpass). So far so good.

In another version, Push Pull Slot loaded (PPSL), the drivers are again wired electrically out of phase but the arranged "magnet-to-face" into a small volume that is open on one side (the slot). The "other side" would be the back volume of the cabinet. In this case there is a reduction in distortion (via cancellation) that would arise from any non-linearity between the cone moving inward vs outward.

My questions:
A. Are the T/S Parameters for the Clamshell and the PPSL configurations the same?
B. Does the reduction in distortion for the PPSL configuration also hold true for the Clamshell configuration?

My ultimate goal is a clamshell configuration in a sonotube cabinet or a PPSL in to a Tapped Horn design. But first I need to get straight on the T/S parameters.

Please....anyone?

Is this possible? I was going to put together one of Marks tube preamps into one of his low power tube power amp modules (https://frogpedals.com/). Then I got to thinking. Maybe I could also put in one of the little 10 Minute chipamps in there as well (10 Min Amp - AUD $7.00 : diyguitarpedals.com.au, A shop for all diy guitar pedal enthusiasts!). Then I would have a .75-2.5W tube amp and a 7-8W chip amp for testing my pedal builds and such.

Only problem is I'm not sure if it will work. I was going to run the output of the preamp into a switch to choose between the tube/SS sections each with a separate master volume or I could put a switch between the outputs of the power sections and put a dummy load on the tube OPT when switched to the SS side.

Any ideas?

I'm putting together a Raspberry PI to play my FLAC music files. This will output SPDIF to a miniSharc using Volumio.
I can run this through my old Sony TV via HDMI. I am wondering if the TV is just a pass through or is liable to be doing something not bit perfect to the data.
The miniSharc resamples the data with sample rate conversion so I don't think jitter is an issue.

Hi there,

I'm getting set up to experiment with using a hi fi circuit (HF60 for starters) that will be using old output transformers from a Leak Stereo 50. They are UL of course and have a primary impedance of 7K. I thought that was unusually high for a pair of EL34's in push pull but, I'm a little bit new to old hi fi tube amp practices.

What differences would you expect to hear if say, one were to hook up an 8 ohm speaker load to the 16 ohm OT tap, especially in terms of volume?

If things work the same with a UL OT as they work with non-UL I would expect the EL34 pair to see something more near 3.5K ohms as a primary impedance - which I have seen that value more typically used in amps than the 7K that the OT's are speced for.

Just to clarify, this stereo amp I'm building up is to be for a musical instrument preamped at line level that possibly may benefit from a more accurate reproduction than say, a guitar type amp - that is to be seen more clearly later on as I experiment with a few things but, I entertain the possibility that since it is an instrument playing only single notes (80Hz and higher) , it may benefit more from a greater volume output than a super low distortion percentage. We will see.

Thanks!
Best,
Phil

Has anyone ever experimented with adding shapes convex/concave to a woofer cone/dust cap?

The original cap on this woofer is concave and has poor top-end frequency response.

I'm looking to increase FR an octave or more. I understand I'll loose efficiency when adding mass.

Hi, I was shutting down my preamp, after I finished building it and listening to it.
I was left with the doubt on the development of the follower cathode.
I have a CCS (C4S) set with 3.2mA by trimmer.
The valve is a 6n1p gold grid - gold pin.
....but i could also a tesla e88cc as an option to be mounted.
With the voltage values ​​in the drawing I have the 6n1p grid at about -3.5v, I think, if I remember correctly.
I can do better?
I tried to center the tensions as much as possible, 138v is the practically exact goal of 275v.
I did a further research and I see that the 6n1p can also be crossed by 7 / 9mA. Are my 3.2mA few?
According to you the voltages are compatible with the assembly of an e88cc ... then I adjust the ccs to a correct current (?).
I don't want to touch the Ra of the previous stage, this maybe puts me some limits. Unfortunately I don't know how to use simulators.
Is it more important that the Vg of the CF is central to the B + or is the negative value of the grid more important?

PS: in the drawing I wrote ecc88 in brackets ... but the one I can put in it is an e88cc.
The stage preceding the follower cathode is ecc83.
I drive 3,5mt of signal cable and enter an amplifier with 20k input.

Thanks in advance for any advice.

Hello everyone,
I need help to mod a table top telephone (no tel line but sim card instead).
I would like to take the hand free speaker signal to send it in a radio mixing desk.
It works but only for the tone. No voice...
I also would like to send a line level feed to the hand free (build in) microphone.
This is even trickier. I made a divider to attenuate (10kohms and 1kohms) and added a 10microfarads condenser to filter out the DC (needed for the electret mic).
The build in hand free speaker and mic are disconnected.

The goal is to use the hand free button to switch from hand set to the mixer and have the listener "on air".
Do you have any advice?

Thank you.

You might need to tap holes in a plate for standoffs, or into a heatsink to mount transistors. If you are like me and are a clumsy machinist, then you have experienced the frustration of breaking a tap inside a hole that you are trying to cut threads into because you are not holding the tap handle straight. If you are lucky enough to own a mill, you can position the tap in the mill to hold it straight, but if not, you may have to do it by hand.

So I have designed these 3-D printed parts to hold the tap perpendicular to the surface of the part being tapped. The tap is inserted through the top and bottom hole, and the bottom surface is placed flat against the surface being tapped. These might be used also to help drill the holes straight as well.

This is a parametric FreeCAD model so you can change the dimensions to suit the diameter of the tap being used. I included 2 mm, 2.5 mm, 3 mm, 3.5 mm, 4 mm, 4.5 mm, 5 mm, 5.5 mm, 6 mm, 6.5 mm, and 7 mm diameter hole guides already generated with the model. A picture of the 3.5 mm model is included which is usable for 4-40 SAE or M3 taps.

If anyone has one (or two) of the yellow/black Kevlar ScanSpeak 21W/8554-00 drivers I'd be interested in purchasing it/them. UK or EU preferred, but given they're long out of production I'd consider further afield if the seller were willing to do the shipping.

Anyone have the Cerwin Vega HED 90 amplifier service manual or the schematic?

I plan to build a Firstwatt F6 and just ordered the parts and boards. The plan is to build a separate PSU unit where i can swap out different amp sections which is house in another enclosure in future builds. This means I can get smaller enclosures. But is there a problem with smaller casing/heat sink for the amp section? What do I need to be warry of?

I’ve been lurking around here for the last few months and I’ve learned a lot. I think I’m ready to pull the trigger but your personal experiences are encouraged.

About 2 years I started my audio journey. For me, headphones were the gateway drug and still a hobby that I very much enjoy. I have a small room that serves many purposes from recording/editing videos to playing video games to listening to music. The room is roughly 12x15. It’s cluttered with shelves and assorted acoustic panels. I have a listening position setup in the room where I sit about 7 feet from the speakers, the speakers are about 6’ apart from one another and they’re separated by a television.

Current setup is computer> Denafrips Ares > Woo Audio WA22 (tube pre/headphone amp) > Schiit Aegier monoblock amps > KEF R500.

What problem am I trying to solve? Variety is the spice of life. Ideally I would get something very different from the KEF house sound. I would say I fall pretty hard in the 4-5 piece band category ranging from 50’s/60’s jazz quartets to the likes of Puss N Boots and singer song writer focused acoustic music. I’m not a bass junkie. I’d say I’m mostly into imaging, sound stage, and performances. I think the KEF’s do this stuff pretty well, but I’m intrigued by the idea of full range point source speakers. The goal at the moment would be to complement the KEF’s rather than replace them.

To start out, I’ll like be using the Aegir to power the speakers. For those unfamiliar it’s a class A/B amp rated at 20 WPC, the first 8ish watts of which are class A. Not the best amp for the intended purposes, but likely not an awful option either.

So I’ve told you about the gear, about the room, and about my taste. I don’t think the frugal horn is a good fit for my room because of distance from wall, but other than that I don’t have many other rules. I could feasibly do a speaker that sits on a shelf, a speaker that sits on a stand, or a floor stander like my current R500. If you have any speaker + cabinet suggestions that you would recommend I’m all ears. I should also mention I’m in the States, I’m fairly handy with a saw, and I’m also open to pre-made or second hand options.

Thanks in advance.

Captain Aloia, regarding SRPP, said that there is an optimal load (in relation to the output impedance) which minimizes the distortion even if a little at the expense of the maximum output voltage excursion, which, sometimes, it might not even matter.

If I remember correctly, from some readings, this optimal load was equal to five times the circuit's own output impedance.
In addition to the source "Bart" ... it is something that we read very often around, this is the ideal impedance.

I don't know if he is the forerunner of this (applied) "theory".

Another example written by an environmental expert:

"The SRPP is often thought of as a circuit with a particularly low distortion, which is not entirely correct.
It is also said that the SRPP has the strange characteristic, compared to the normal gain stages, of having the distortion decreasing at the decrease of the impedance of the load, and this too is not entirely correct.
What occurs is that as the impedance of the load decreases, the distortion increases regularly, up to a certain impedance value. By further reducing the impedance of the load, the distortion instead begins a decrease and reaches a minimum, beyond which however there is a rather marked increase in the distortion.
This phenomenon has been (relatively) recently analyzed and exploited by Bartolomeo Aloia in his creations.
The knowledge of this phenomenon therefore allows to minimize the distortion of an SRPP stage simply by appropriately choosing the impedance of the load.
This becomes very interesting because the impedances we are talking about are generally quite low: for example, from the simulations I made it would appear that in a case very similar to that of "SIMPRE" (ecc88) the distortion grows inversely proportional to the load when this decreases above 50kohm, then it begins to decrease again up to a minimum placed around 15k ohm of load impedance and below 15kohm it begins to grow again very rapidly. In practice the distortion with an impedance of 50kohm is almost double that with a load of 15 kohm.

Aloia therefore in its creations adopts the technique of minimizing distortion precisely through the careful selection of the load. "

In my srpp that I have just built, I am undecided whether to put an input pot (100k) or a very low value output pot. I use a 6n6p srpp at 200vdc and 14mA.

Opinions?
Thanks 😀

Hello,

I am currently working on an amp that uses a fixed bias supply with a source follower direct coupled to the output tube in a single ended design. The bias supply is shared by both sides of the amplifier.
I originally derived the negative bias voltage from the main B+ winding with reversed diodes, but I noticed that there was a slight hum and I didn't like the amount of heat generated in the large dropping resistor so I instead used a small transformer that i had handy that provides 200VAC/100mA.

When I set everything up, I was happy to notice that the hum was gone however, one of the output tubes was severely over-biased. With the driver and output tubes removed, I rechecked the bias supply and sure enough, instead of the maximum negative -100VDC bias I was reading around -50VDC on one side. Thinking that the mosfet was likely blown, I replaced it, only to then have the other side start also dropping voltage. OK, with both mosfets replaced, I put the tubes back in their sockets and after the warm up-- grrrrr a very loud buzzing sound.

With the tubes removed, I added first protection zeners (which I had forgotten at first) which didn't help. I then added a small break out resistor on the source of the mosfet. This seems to be beneficial for the life of the mosfets as they aren't dying now, however, the drain voltage has now started drooping into negative voltages. I am perplexed as to how this is happening.

I have made some small sketches of the schematics in question (sorry for the hand drawn aspect!) Is there something in what I have done that is obviously wrong? Surely I am missing something here...

Hello,
I am currently building a DIY Bluetooth speaker. I used this video as a reference: Build your own Concrete Bluetooth Speaker (how-to) - YouTube

I followed all the instructions in the video, I used all the same components (except I did my own enclosure design and used similar drivers with the same ratings). Other than that, I followed all the instructions closely, and sure enough, at the first test, everything worked perfectly.

But during the final assembly, I accidentally shorted one of the speaker output wires from the amplifier (disconnected from the drivers) with the battery pack. I saw a little spark, and since then, the speaker is playing at a very low volume.

So I bought a new amplifier (thinking the old one got fried). But after I swapped them, nothing changed. What component is damaged, then?

The speaker works normally, only the volume is low. It isn’t a transmission problem because the opening jingle is also very quiet even at full volume.

Any advice will be very much appreciated, as I am really lost at this point.

____________________
Parts list:

• OEP30Wx2 Audio Module Class D Digital Power Amplifier
• USB Wireless BluetoothAudio Receiver 3.5mm AUX output
• DC to DC Step-down converter
• DC-DC 5V Power Supply Module 4 Pin Isolated converter
• B50K 50K Ohm Dual Switched Potentiometer
• 21V 18650 battery pack with protection
• 2x 18W Speaker drivers

Hi all , trying to upgrade my audio interface and up to the oscillator. but what value is this oscillator?

Thanks you

Hello, I hope you are keeping well.

Just a quick question regarding adding a flare to a recommended port size.

I am designing a 2 way speaker. I've got some rough figures (need to do some tinkering but wanted to get my head round this part first) for the sim.

VB = 9L
Tuning freq = 55hz (do plan to try get it lower later on, just experimenting atm)
Vent diameter 2 inches
Vent length 186mm (again, know this may be a squeeze, haven't finished tinkering)

Unfortunately around the 30 - 50z region, my air port velocity just exceeds 18m/s. I was reading how if you flare a port, you're good aslong as it doesn't exceed something around 36 m/s (cant remember exact figure)?

Therefore, I was just wondering how you go about adding on a flare to the suggested port length winisd has given you ? Do you obtain half the flare radius and subtract from length of port if that make sense?

I know this a nooby question and I may be missing something blatantly obvious. I had a go on flare it with no success. I really do appreciate any response, even if its stating something simple. I'm completely new to speaker designing, not the best at this sort of stuff but always up for trying new things.

All the best
Have a great weekend folks 😀

Hi. I'm trying to establish whether the RCA8000 Power Triode is equivalent to the RCA810 Transmitting Triode. Reason being that I'm looking at purchasing a pair of push-pull RCA8000 mono block amps and I want to make sure that replacement tubes won't be an issue. HT in the amps is probably 800V. Currently trying to locate the schematics but it's probably one of the ones in this thread: 8000 electron tube.

The tubes look the same and the bases and pin-outs are the same but there are differences between some parameters. Question is whether the tubes would be interchangeable in a push-pull amp.

Salient characteristics are shown side-by-side below.

So, the Amplification Factors are different but the grid voltage and plate current aren't specified for the 810.

There are minor differences in interelectrode capacitances.

The AF Power Amplifier Maximum Ratings are the same.

For "Typical Operation" the DC Grid voltages are completely different. The Maximum signal driving power differs by about 35%. Other parameters are very close.

I've also superimposed the anode characteristics. The 810 tube curves initially show a higher current but show a lower current above 800V or so.

>>>>>>>>8000>810
400V/+60V 32mA 34mA
800V/+60V 49mA 45mA

The variation probably isn't more than you'd see between tubes of the same designation.

My guess is that you could replace a pair of 8000s with a pair of 810s but wouldn't want to mix them.

What does the forum think?

Does anyone have the schematics for this 4 channel amp? I can't find anything on the internet. It is identical to another model (king sound eca-504). It was going in protection mode, I repaired a broken solder connection and a blown transistor and seems to work now. However, when connecting the sub in bridged mode (channels 3 and 4, since 1 and 2 have the LPF broken) it's almost inaudible (the other stock speakers cover its sound). This is strange, because if I unplug one channel from the inputs it starts to be louder.
I tested the output channels individually at home with an oscilloscope (but with no load as I don't have any) and I get the same input wave (amplified) in all of them. I also checked the rails where I can see +/- 20V steadily.
What else could I check?

Ok, my timing is bad, but I have only half of the Baltic Birch that I need for my next speaker project. It seems that wood is getting increasingly hard to find 😡

So, which wood should I choose for the balance of my needs? I have several choices at my local supplier:

1) "paint grade" Lauan in 3/4" that appears to be 8-9 ply.
2) "good one side" Duraply in 3/4" that appears to be about 7 ply.
3) back to MDF

The Duraply is a heavier/more dense plywood than the Lauan. Both are "relatively" void free (though not 100% like BB).

I welcome your advice, suggestions, and input!

Thanks!

Hi

Looking for service to be performed on my Nakamichi OMS-5II CD Player.

I believe the laser is good, however I need to have the Alignment process checked with the proper Test Equipment.

The player ran 3 days without missing a track after I did the following:

1-Removed all the Glue on Caps
2-Light lube
3-Proper laser clean
Adjusted the Laser alignment t without scope!!!

I went to turn it on, and it started to play, one channel was out and static noise on both L and Right. After my attempt to adjust the pots, as said, then Player worked for 3 days but now is back to not working.

Help and thanks for any information!

B-

This amp is using MPS6523 in the output section

Anyone know of a good replacement?

Recently acquired one of these and planning to restore it, currently looking for a source of a schematic for it. Can anyone help?
cheers

Hey all,

I'm interested in building a small amp for desktop and headphone use, using the SSE design as a starting point.

I'm thinking currently of 6V6GT/A with the operating point of:
B+ -- 275V
Vp -- 250V
Kr -- 390R
Opt -- 5K

* 1) For plate voltage of 250V with 275V B+, does that occur simply from the voltage drop across the output transformer or is anything else needed?

I've been looking around here for which mods on the SSE are needed to support lower voltages (around 275V B+) and found:
-- Jumper instead of 10K resistor to CCS
-- If using 12AT7, increase current set resistor to 470R
-- Alternatively, use 12BH7A. Although I'm not sure if any further changes are needed except swapping the tube.

* 2) Any other frontend changes needed?

Also, I will attempt to "roll my own" PCB for the SSE minus the power supply. This is to better fit the design I have in mind for a smaller chassis.
It is in early stages, I like the symmetry but not too happy about "weaving" the pentode plate trace next to the signal. I'll probably have to rotate that tube and figure something out.

* 3) Has anyone tried to use a 12AX7 ? I've got alot more of those than 12AT7, and since I'm making the PCB I could incorporate any changes needed to support that.

Thanks in advance.

this is a 2 chassis preamp, power supply & preamp connected via 4 wires umbilical cord using Lumberg LV thread locking connectors.

Few days back troubleshooting a problem (not solved yet) , found this. The yellow wires (ground) on both chassis board end on connectors where gone to no wires, and on the umbilical cord is are a black wire that happen the same (to nowhere) but on opposite pin. Could be a miss from the builder ? None has continuity to chassis or AC ground outlet.

If YES how it works good for years ? 😕

Hi,


I'm building the Seas A26 speaker kit and I'm looking for a place to mount the tweeter's single crossover capacitor, a Mundorf Supreme. The back of the mid-woofer has an extra tab opposite the speaker's terminals (first photo). Is there any reason not to ziptie the Mundorf to this tab? The cap is about as big as a C-cell battery, and I don't want to wire it point-to-point to the tweeter without anchoring it to something. An alternative is to mount the cap using one of these grommet things to either the interior front or interior rear of the speaker cabinet. Any advice?


Thank you!

Hey everyone,

Hope this makes sense:

I had a pair of Pensil 11s commissioned but the speaker holes weren't countersunk and I'm experiencing pretty bad baffle diffraction. I've remedied a lot of it by placing compacted wool around the drivers, however I'm looking for a permanent and better looking solution.

The cabinets are currently at a CNC/laser shop and he's figuring out a way to make the speakers flush with the cabinet.

• Our plan A was to see if we could fit the built cabinets in the laser / CNC machine to accurately cut the 4.6mm required depth-- unfortunately neither machine had adequate height clearance to do this.
  
• Our plan B is to see if he can construct a jig to accurately countersink the existing speaker hole the required depth to flush mount the speakers without damaging the veneer.
  
• Our plan C is, as a last resort, to CNC a custom front panel to overlay on the front of the speaker cabinet to emulate a flush-mounted driver, which will then be veneered to match the existing finish, either with a hole cut in a 5mm panel or a thicker panel with the 5mm countersink CNC'd into it.

Assuming we have to go with C:

1. What risks to audio quality will I be introducing by adding a solid 5mm panel to the front of the speakers to make the speakers flush with the cabinet? (assuming it's secured tightly and there's no gaps between the cabinet and the newly added front panel)
2. If I have to go this route, given we have access to a CNC machine and laser, is there anything specific that would be beneficial to add to this panel, or is a completely flat panel ideal? (example: thickening the panel and CNCing a wave guide vs. just keeping the panel completely flat)

Any help would be appreciated. See picture for a hopefully better clarification here:



Thanks in advance!

Hello Friends,

I want to 2.1 System rather than 2.0 3 Way Tower System.

what should Minimum Size of satellite (tweeter+ mid/woofer)enclosure for faithful reproduction of above 100 Hx.

Below 100 Hz I plan to use Sub-woofer.

Thanks in Advance.

Hi guys, first post in this forum.

I am trying to build a DI Box with an ty 250-p transformer.
This transformer is very linear in his frequency response, and I tested it for core saturation purposes earlier for another project and find out that it accepts 5V without clipping or making resonances .

But now Im having troubles with resonances around 1k/2k when I use it as a DI Box transformer, specially with passive bass. I do not know how to avoid those resonances.
I am comparing it with a palmer passive DI box.

I am doing this type of circuit



And this is the datasheet of the transformer

https://www.google.com/url?sa=t&sou...FjACegQICRAC&usg=AOvVaw2TSbSlK1NplrnO6Shoa6k6

Can someone tell me what to do to avoid these resonances?

Thanks

Hi, I am downsizing my office and have a set of AES Journals from 1989 to 2020 that I would like to re-home. I think that the set is complete, but I haven't checked. Probably too big to post, but I am willing to deliver within 10 miles of the South Coast of England from Poole to Southampton. I'm sure that someone, somewhere will want them - I really don't want to have to put them in the skip!

Hi folks. Hoping someone can give me a clue. The thing keeps cutting out every 20-30 sec or so for just a second. What could it be? I have another tuner without the issue so it's not the broadcast. Good contact on all connections. TIA

I'm familiar with the OTK stamp, but what is this K stamp? Some of my 10µF 600V MBGO caps have that.

OTK caps for military, K caps for something else?

As a curiosity, does anybody know what the bottom line means?

Hi all,

I am interested to know exactly what sources of noise the S/N ratio listed in turntable specs is assessing. Is it all mechanical and electrical noise sources or just one or the other? In particular, is it likely to be affected materially by the standard cartridge provided with the particular model? If so, for me this would mean the figure is less useful as I would soon be upgrading the cart (within reason - obviously -40dB vs -90dB implies some significant differences in the turntable construction other than the cart).

Any info on this matter would be appreciated!

Cheers,

Greg

Want to buy a 4 deck, 20k to 50k stepped potentiometer for balanced solid-state preamp.

Thanks,

Vince

The question here is how to divide up an amplifier that would be too heavy on one chassis.

1. Monobloks. Looks nice, compact, good rectifier choices, but duplicates the mains transformer and chokes.

2. Separate filament supplies. Very useful for DC choke input supplies to DHTs like 300b, 2a3, 26, 10Y, 01A etc. The transformers and chokes can be big if filament bias is used e.g. on 26 or 10Y, and the transformer voltage varies with the operating point. This solution offers flexibility and keeps chokes away from the main chassis

3. Separate PSU. This way you put a good chunk of the weight - mains transformer and first choke - on a separate chassis dividing the total weight neatly. Downside is the heater supplies can be in the mains transformer, so AC leads etc.... or convert to DC....

4. All heater/filament supplies on main chassis (unless chokes are a problem with DHTs). Separate PSU chassis for just the HT. Allows all the valves to be heated before the HT is switched on.

5. Other?

What format do you like to follow when the amplifier is too heavy on one chassis?

Hi folks,

building up an active speaker from existing parts/modules, the question arose how to build a star grounding facility.

In total I need to connect eight modules which are all mounted on a large power amp heat sink:

4 power amps
a module containing differential input amps and analog cross over
+/-15V Regulator board
power amp supply
control circuit board
potentially more...

The star grounding concept as such is completely clear, so my question is about physical detail:
How do you electromechanically build up such a star ground configuration? Which mounting and connection materials do you use?

I'd be very grateful for your shared experience and for actually sharing your project picture(s).

Thanks and regards,
Winfried

"Rundfunk- und Fernsehtechnisches Zentralamt der DDR" called "RFZ"

Hi,

if you need some informations or schematics about amplifiers and other gear from the "Rundfunk- und Fernsehtechnische Zentralamt der DDR" you con look at this site: Home - Rundfunk- und Fernsehtechnisches Zentralamt der DDR. Currently it is only in german but i think you can use google translator.

There you can find a lot of information this company.

If you have some material about this company (schematics, pictures, papers and so on) please send me this material (as PDF or othe file formats possible). Then I can add this material on the website.

The website is continuously updated.


Best regards!

jokeramik

So after a lot of soul searching and going around in circles I finally decided to bite the bullet in a sense in creating a tube based Analogue to DSD amp. This follows the plan with the SMPS with the power being an isolated switching of the DSD output to give the final audio.



I've started with the basics that I'm currently building out by creating simple first order logical analogue to digital DSD converter.

These building blocks are relatively simple to then map into a valve topology.

U2 - difference - this will simply be a differential amp using two triodes.
U1 - integrator - this will again can be two triodes.
U4 - the comparator - this can be.. you guessed two triodes.

D-latch = this can be done with OR-NOT and so 5 triodes.

At this point the output is DSD and so simply I need a gate driver, or simply a push-pull set of tubes (replacing SW2 and SW3). These simply have to be able to produce ON or OFF so they don't need to be very linear.

I still have some more work todo on the logic before I jump into the tube implementation. Specifically making the system biased to operate in 'class A' so it will be tube compatible.

There are some 'additions' in terms of functionality I could do such as make U2 the front end amplification, even making the input differential..

both are good woofer almost clean to 500hz, but what si the big difference except from the size. It seem not many people use l26roy as the bass woofer

Here is an example of a fake capacitor.

I would like to build something like this:
These directional speakers throw sound in focused beams like a spotlight | Yanko Design


But I want it to be quite a bit larger, capable of high SPL. I only need it to produce audio above 5 or 6 kHz, so I'm thinking of using a compression driver for the audio part. I want to use pipes to mix the sound waves.



I was thinking of simply summing the audio and ultrasound by putting a pipe inside a pipe, both possibly having an exit throat at the same point, perhaps a flattened pipe (see attached images). Would this be sufficient to modulate the pressure waves?



I was wondering if anyone here would have any tips for me, or if anyone knows of some online acoustic simulator that could aid the design?


This is a personal project as an exercise. I looked up uni-directional drivers, and found the link.

Hi all.

I have a pair of Goodmans LS3/5A's which were not properly sealed around the terminal posts. Upon opening the speakers up to investigate I found a clump of hardened black goo covering the holes where the speaker terminal cables pass through. I suppose this was rubber which deteriorated and hardened---the substance was brittle but not brittle enough that it was easy to crack and tear away.

I've managed to remove the deposits from both cabinets but now I have a dilemma: how do I go about resealing these units? Clearly the original rubber seal wasn't a good choice: those two holes I'm trying to seal are aprox 10mm in diameter and are just there to allow the speaker cables though (these use the old screw type connectors (photo attached).

The first thing that came to my mind was silicone like that used to seal a bathtub but I really have no idea if that is appropriate or if it will breakdown the same way the original rubber did after a couple of decades.

All advice appreciated. Many thanks.

I, like others before me, have noticed that it's annoying to listen in headphones to recordings with loud sounds panned hard to one side, left or right. For me, it's worst with midrange sounds like lead vocals, electric guitars, solo horns, drum sets play from one side only. It's like someone shouting at you into one ear.

I rigged up a Stereo-Mono switch, which was easy enough and didn't cause any degradation I can detect. I find myself using it on early stereo records from the 1960s, especially Blue Note and Prestige jazz records. (It can hurt when Elvin Jones takes a wild solo in the right channel!)

I'd like to experiment with a passive Blend control, preferably infinitely variable so I can dial in how much the stereo image is reduced towards mono.

Some old receivers and preamps had a Blend control that allowed you to switch from full-width stereo to constricted width stereo, to mono. Some had Mono-L or Mono-R settings, so you could hear just one channel in both speakers, which I don't think is necessary for everyday listening.

I'm surprised I found so few examples on the innerwebs. I've attached a scrawled set of schematics showing three simple examples.

Ex A: Lifted from a harman-kardon TA-5000X tube receiver. It uses a dual-gang 100k potentiometer. I assume linear taper.

Looking at the pots with connections labelled top (1), wiper (2), bottom (3):

I think this would only work if the top (1) connection of the left channel pot goes to the Left channel input, while the (3) connection of the right channel pot goes to the Right channel input.

The bottom (3) connector of the left channel pot and the top (1) connector of the right channel pot are shorted and connected to signal ground.

The two (2) wiper connections are shorted together.

This arrangement would put the two tracks connected opposite from one another, so that when you turn the knob clockwise, the left channel track moves clockwise while the right channel track moves counterclockwise.

When you move the knob all the way clockwise, the two wipers will be grounded, and the two channels will play in full stereo.

When you turn the knob all the way counterclockwise, the left and right channels will be shorted together, with each 100k track loading each channel to ground, so the two channels will be blended together into mono.

Ex B: I found this on an electronics hobbyist forum. Uses a single track 10k ohm pot connected between the channels, preceded by 1k series resistors, one for each channel.

Turn the pot full clockwise and the two channels are shorted together (mono). Turn the pot full counterclockwise and the wiper is shorted to ground, leaving the full pot resistance between the two channels (stereo). The pot is not connected to ground.

Looks nice, but will this inevitably leave some 'bleed' between channels, causing permanent reduction of channel separation? (That might not be a problem with headphones. How much separation do you really need?)

Ex C: Lifted from a cheaper harman-kardon tube receiver (TA-3000). It uses a single 2M pot with its top (1) connector left disconnected, wiper (2) connected to the Left channel signal, and bottom (3) connection to the Right channel signal.

It looks like this one attempts to keep the two channels from blending (bleeding) too much by using a high value of resistance. Does such a high resistance introduce enough capacitance to cause attenuation of high audio frequencies?

Are all of these doomed to failure for one reason or another, and that's why you never see this kind of control?

The source I'm using is a Raspberry Pi with an Allo Boss DAC (output Z probably well below 1k ohms). The load is a tube headphone amp with a 50k volume control pot, or an O2 amp (10k vol control). I'm going to try to wire a box up today and give these controls a try. I'll post results if I find anything noteworthy.

Thoughts?
--

I'm helping my father revive some old cabinets my grandfather had made for speakers in Hobart, Tasmania, circa 1960-something. He's kept them for all these years and we just pulled them out of storage. The timber (teak) is in beautiful shape as are the front grills - I love the retro look. The drivers however are beyond saving. The timber for the baffle will be replaced to suit the replacement drivers.

I've built Frugel Horn mk3's to date but this is my first attempt at a multi-way speaker. The front baffle is 330x610mm, tweeter is an SBA SB29RDC-C000-4 and woofer is an SBA SB23NBACS45-4. Enclosure volume is ~41L once the drivers are in there.

I've simulated the Baffle and enclosure and am fairly happy with where I've got to so far - my concern however is the roll off from ~300Hz. The speakers will be in corners so I expect there to be some room gain... Anything I try only results in the low end dropping or the mid-high dropping, requiring much more power (the amp to be used is fairly low powered). Without fully understanding enclosure effects, my hope is that the total SPL increase indicated in the enclosure chart from 300-50Hz will compensate, but again, this is a gap in my knowledge.

What can I do about this? Additional comments on impedance and phase are welcome too. Target XO is 1750Hz 🙂

Hi all,

I am trying to source driver units and perhaps crossovers for a DIY repair of Wharfedale Jade 1 speakers.

The sorry tale had me playing an Audiolab 8300 cd player directly to twin power amps inadvertantly set near max volume. They have been over driven and it is a question of what isn't broken. No sound is never a good sign as is nothing like the proper sound.

How you may help is in a where to buy context. Here are the driver details of this three-way
Tweeter is a Cambridge Audio D-951
Mid range a Cambridge Audio D-916
Woofer Cambridge Audio D-910

I have just obtained a multimeter to continuity test to prove the damage if you will.

I have no idea on prices or where to buy here in UK. Have written an email to Wharfedale, nothing back yet.

I would like to have a go at even upgrading these with a new build. Advice welcomed. Thanks

Hello, I would greatly appreciate any help solving a distortion problem in the TL12s I am building.
There is a significant sibilance at the HF end of the ATC SM75-150 mid range speaker. It sounds as if the speaker is mounted on a sheet of thin ply rather than the 1" MDF baffle it is securely bolted to, I exagerate as it is only noticeble on quiet passages, particularly piano music. And the distortion is present even when the speaker is on the bench.

Both speakers suffer the same problem. I have no reason to suspect a problem with the amp so I suspect the crossover. It is the standard WA Prestige Xover design but I am using (on the adviceof WA) the Scanspeak R3004/662001 (4 Ohm) rather than the usual D2905/9900 (6 Ohm). I note WA have increased the series resistor to the tweeter to 1R5 rather than 1R. The only other changes they have made are to the mid circuit: 1R rather than 1R2 on the series resistor and 1R5 rather than 1R2 on resistor in series with the 6.8uF cap. I am sorry to say my understanding of Xover electronics is very poor.

Until I can remove the distortion there is no point in seeking further refinements.
If anyone can provide advice on this problem it would be very helpful.
Thank you.

Does anyone have a preferably matched pair of these caps. I've got loads of different K73-16 values but not these, can buy or swap for different values.

A few years ago, I presented a small resonant converter, the Q-conv:
https://www.diyaudio.com/forums/power-supplies/286078-resonant-hv-converter-conv-post4599222.html

This version uses the same principles and has similar features, applications and specifications, but it is somewhat simpler, thanks to the single-ended power section and a minimalist control circuit.

The resonant principle allows a "soft" operation, without large amplitude transients, and does not require custom magnetics: a standard, two-terminal choke is sufficient, and in applications where non-isolated input and output grounds are acceptable, it provides a simple and quiet solution.

This is the circuit:



And this is the output waveform (unloaded) on the tank circuit:



This is the same waveform under full load conditions:



The waveform is essentially a sequence of sinewave cycles, separated by variable-length zero-crossing "pauses".
In addition, when the output is loaded, the rectifying diodes slightly clip the tops, but compared to a conventional, switching converter it remains very benign.
With the values indicated, the operating frequency is comprised between 50 and 80kHz.
The circuit is designed for an output power of 10~20W, but there is no lower limit. Going higher would also be possible, but it is probably not advantageous because of the limited efficiency: 74% @15W.

Here is a pic of my prototype:



The inductor is based on an unholy hybrid between a cross-core and an old potcore, because I didn't have a suitable standard inductor available, but basically, any standard 56µH/6A power choke would work (and would have lower losses than this one)

Hi I bought two ATC means SM75-150 STD 8ohm for my PROAC E.B.S
and I want to replace them brings 16ohm source
I would know someone tell me that I have to change
thanks

Hi all

I purchased a Denon PMA-700v for parts but decided to see if I could get it working. I am a beginner in electronics and have hit my knowledge limit and was hoping someone may be kind enough to offer some suggestions.

The amp had one channel blown and I spent ages removing and testing components. I replaced the output transistors (originally 2sa1104/2sc2579) with 2sa1943/2sc5200 and driver ? Transistors 2sa1306/2sc3298 with mje15033/15032.

I powered it up with a variance and dumb bulb tester not problems. I then kept swapping higher wattage globes until I had a 1.1 amp current limit. Both channels produced great sign waves using dummy loads.

I then thought I would remove the dim bulb tester and ramped up the variance to 240v no problem. I hen turned up the volume u til I reached the previous limit using the dumb bulb tester (9.8vrms, 27vpp) with no issues. When I increase the volume some more to 12vrms R560 burns and all output transistors are killed.

ANy suggestions as to why this may be happening. I did have a relay issues that appeared to be a broken solder joint I fixed. Any help appreciated, it appears the outputs die first ?

Richard

I have a small box that says "Mains interference suppressor", above which there is a label "PANDA - distributed by Pact International Ltd".
On the side of the box it says "This unit is suitable for audio & hi-fi equipment".

There is a mains plug (UK type), and the box has a socket for a standard UK plug.

Does anyone know about this? Is it worth adding it to my system?
How can I test it and see what it does?

Thanks. 🙂

There is an auction on *bay with reasonable price for this part. Have you any idea about what is the actual part number for this mosfet? Seller claims that it is IRFP3710 equivalent but there is no such supporting info.

I have four of these drivers, which I bought from Blue Aran to make a Boominator. I never got around to it, so they've never even been opened. Offers around £30 each plus £10 each for postage (in the UK). International shipping possible, but might be expensive (they're about 3 kilograms each).

I have tweeters as well (see separate listing),

I always wanted a PASS amp, but didn't really have the cash for one. I heard a SIT-3 that was factory and really liked it. I know we can build a 3X, but I wanted to give something back to one who has given us so much, and also have one direct from Papa.

I went looking for a SIT-3 and could not find one in stock, so I gave Kent a call and asked if he could find one. He did the legwork and came back with, what is likely, the last SIT-3 still in the box/wild.

It just so happened Fred Kat/li had one left in a box and was willing to part with it. I jumped on it and it's shipping out of CA and will be here Friday.

I'm excited to have an amp from the shop, I might even build one and do some A/B stuff and measurements. Heretofore, I never had the real deal to test.

I know this isn't a huge money deal and Nelson doesn't really need the cash especially the little from this, but it is still nice to have an original from his shop. 🙂

JT

This is a thread to help me keep track of the 'cheapest speakers' I build. Feel free to add your own.

Rules, parts must be new and available. Prices in USD.

Tweeter = $1.98

https://www.newark.com/mcm-audio-select/53-805/3-x-7-piezo-horn-tweeter/dp/96K1143

Woofer = $9.18

https://www.newark.com/mcm-audio-select/55-3231/70w-rms-4-ohm-rubber-surround/dp/43W7872

MDF box = $10 (30 liters)

Rear Terminal - $2.84 (could go a $0.41 version)

Optional = 3mh inductor (common mode choke for a PSU? ~$3)

Glue = $1.0
Staples = $0.50
Wire = $0.50

It still makes it ~$30 per enclosure. hmm

Don't worry, the dog is deaf.

The graph is 1/6 smoothing


Can it REW? yes

Sound = OK with/without the EQ

How does the size of the driver relate to the width of the baffle in a conventional surface mount configuration with regards to where in frequency the baffle step correction needs to kick in and to which extent?

Is there any relation at all or is it simply a function of wavelength? In other words, will a 6,5" midbass driver get more support from a 600mm wide baffle then its equivalent 15" due to the greater relative baffle area to driver size (or Sd)?

Hello,

the issue is about Audiolab 8000CD, made in England, one of the first production series I guess. One day after switching player on I heard only short rasp and then saw ERROR message on the display. Obviously player was not able to read CDs anymore. I replaced whole CDM12.4 mechanism with only poor result: player started to read CDs, displayed number of tracks on the CDs and total time but after choosing particular tracks ERROR was displayed again. I thought that CDM12.4 is not broken despite brand new (I think this was not original Philips) and replaced it again with new piece - this time laser/optic unit only. Symptoms are exactly the same like after first replacement.

Any ideas ? Could the problem lie in the mechanism / laser itself or should I look somewhere else ? I read that these first production pieces of CD8000 suffered from 'some problems' ?

Best regards and thanks in advance for any hint,
toms789

Hi, new here so finding my way! I run vintage Cambridge Audio R40/50's and Lentek Monitor speakers all of which use various iterations of Kef B139s. I have spare 'later' ones for my Lenteks when needed so no problem there. The Cambridges have the earlier 'racetrack' versions which require a different opening in the baffle to the later ones. I have acquired a pair of racetracks but they are rough and I want to rebuild them. Those of you who know these speakers will realise that 'normal' rebuild procedures do not apply because the 'cone' is a solid lump of 'styrene. They can be rebuilt, I've had one done in the past but the price is exorbitant, anyway I want to do it myself. So has anyone any hints and tips which may help please.

I wonder if one of you guys would be able to help someone barely a newbie on simulation software with a model (info in title and a screenshot of the symbol attached)