I recently built an up firing omni speaker. I'm trying to make sense of klippel bl and kms symmety graphs. Looking at the plots is the slight sag of upfiring working to my advantage?

I'm ready to have our annual meeting again after the new year.
We'll schedule it for the 3rd Saturday in January.


We might as well meet at the same library as the last time
unless we find a sponsor's location.

Hi, included are photos of a 35C5 OPA541 hybrid amplifier. Please only include commentary relating to the thread title.

I have build myself a preamp with 12B4A tubes. About 80v-ish on the plate. Dropping voltage from around 250v over a 11KOhm resistance. Pin 2 and 7(grids) have each a 2,2kOhm connected to the 100K volume pot. A 470 Ohm resistor on the cathode, no capacitor in paralell.
Sometimes the volume starts to increase, then goes "down" again?, witout touching the volumeknob. Can somebody tell me what is wrong. The preamp sounds great if I compare the preamp to other preamps I have12AU7 srpp, 6j5 single stage, but after about 15 hours use the preamp starts to behave this strange.

Recently, I have been trying to put together a fun, simple project to make a headphone amplifier for use at the office. I plan to use this amplifier with a pair of Grado S80 headphones. I already have the PSU drawn up and the simulations look good for producing a smooth 20V through a standard LM317 voltage regulator. However, I wanted to check if my circuit looks correct because it is a modified version of the one from DIYaudioprojects.com found here: DIY IRF610 MOSFET Class-A Headphone Amplifier Project based on some suggestions I saw on this website. I guess my issue is when I run this simulation on LTSpice, it looks good until I get to the voltage being sent to the headphones themselves and I am not sure why or how I can verify this circuit. What kind of simulations should I be running? How can I switch this to reduce PSRR using a negative power supply or perhaps a P-Channel MOSFET? What alternative to the IRF610 would anybody suggest? I have so many questions.

I'm intrigued by hopefully building some speakers with 'Onken' style enclosures)(I'm currently leaning towards 'classic Onken' rather than the Planet10 alternate 'Fonken' -although I have the highest regard for Dave and his work).

I'd be open to either Multi-way or Full-range, but with a definite gladness to go full range if I find suitable drivers within my reach. And I think I have found two potentially-excellent candidate drivers for this: the Tang Band W6-2144 or W8-2145.

I've been using the 'Onken calculator' at HiFi Loudspeaker Design

The W6-2144 looks like it performs very, very nearly as well at lower frequencies as the W8, and the W6 manages to do so in a much more modestly sized enclosure. PDFs of calculations (from the above-mentioned onken calculator website) attached. The results I am getting from the online 'onken calculator' show the W6 an f3 of 33.8hz in a 60.8 liter enclosure vs the W8 with 32.5hz in a 100.7 liter enclosure.

Difference in price between the drivers $91.10 for the W6 vs $96.90 for the W9 is practically a rounding error, so prices of the drivers aren't a reason for me to lean towards one vs. the other.

My questions:

1) Can I reasonably rely on this online 'onken calculator?'

2) I don't yet have real depth of understanding (at least not nearly what I'd like) of how the multiple characteristics of drivers interact with different size and configuration enclosures - so- am I missing the boat here somewhere in what the calculator is telling me, both in relation to:
3) Aside from "graphed frequency response" and a slight gain in db per watt efficiency in the W8 over the W6, is there any other reason I should consider going with the W8 over the W6?

4) similar to "3" but more narrow: if I use a single-ended triode tube amplifier with no negative feedback, would the W8 in a large enclosure present a better load and match to the amp than the W6 in a smaller enclosure?

I realized that I stacked a lot of 'ask' into this question, so thanks for plowing through it. and I appreciate and look forward to input. Thanks in advance!

Hello, you were all so helpful when I was upgrading my stereo speakers a while back and I'm hoping your collective expertise can help again today!

I am hoping to make a mono 'grotbox' speaker for checking my mixes on.

I have a single driver passive speaker to do the job. At the minute I am running a Lepai Class T amp to drive it, but it keeps making loud clicking noises intermittently and scaring the life out of me.

I'm hoping to turn the speaker into an all in one unit with the amp on the inside and a power-switch attached to the outside somehow.

Part of the point is that it doesn't sound great so I'm looking for something cheap and effective rather than 'good'. Any advice is appreciated!

I've attached pics of the speaker and driver.

Audio will be coming from Protools > RME interface > Mackie Big Knob

Thanks in advance 🙂

The Carver ad indicates a five year warranty on amp & tubes, with "Unique Carver Tube Tech" and variable bias. Nothing found on Carver's site and there's little info on-line.

It could be image distortion because the output tubes look like taller KT-120s. Does anyone, here, know the details on this <$5k amp? If an elaborate biasing scheme successfully develops 180 watts from a KT-120 pair, at low distortion, long life, and reasonable cost, what might be the trade-off? Or aren't those tubes KT-120s? Intriguing.

Hello,

Attached is a pic of the board inside my Steve Sank modified Parasound HCA-1000A. I am currently recapping the four large filter caps located in the center and am cleaning the PCB from their electrolyte leakage. It eluded me, but I found another electrolytic cap, which I have circled here in red, which has also leaked some electrolyte onto the board but no bulging at the top at all. It's a 25V 1000uf cap. As you can see the amp is laid out "dual mono" if you will, with a few parts not mirrored on each side and this part is one of them. It's located near that Omron relay. Anyone know what the function of this cap is and if I should swap it out? Wondering how silly it would be if I tried to get by not changing it. Any help would be appreciated. Thanks.

Hi Diyaudio,

While staying at home for COVID reasons, I'm studying different 45/2A3 DC coupled schematics. the following schematic from Shishido San is puzzling me.

What could be the purpose of 22k resistor, between the 300 V B+ supply of the first stage, and the cathode resistor of the power triode ?

Thanks for your assistance. Warm regards,

Jerome

Would the amplifier in a typical plasma tv be able to drive some external speakers, assuming the external speakers were sensitive enough?

I want to disconnect the internal speakers and connect the (same) wires to some external single driver speakers. (I'm guessing the internal amp is anywhere from a few watts to a few dozen watts?) The TV is at my in-laws; one of whom is technicallly challenged, the other has Alzheimer's. I've tried external amp from the line level out, but between the extra remote, and choosing the input on the external amp, etc., they're constantly confused and I just want to make something for them that sounds better (one has hearing problems too) but is seamless from their standpoint. I'd build a pair of bookshelf speakers with some high sensitivity full-range drivers.

Is this likely to work?
I could also feed that internal speaker output signal to a simple (Lepai-type, always on) amplifier and then onto the new speakers, but don't know if the input voltage to the Lepai would be too high, or if there are impedance issues I need to consider.

Thanks for any advice. Watching TV is one of just a few things that make my father-in-law happy.

Hi there everyone.

I have a basic home recording studio setup at home here with some old synths, a bunch of old microphones and a computer with a Lexicon Omega and an Mbox 2 interface.

I'm usually using a mysterious Okay-II dynamic microphone which I inherited off my brother in law, which is good as it isn't very sensitive so it picks up less background noise.

Anyway I found these old Japanese AMI Jorgen SE-25A bookshelf speakers on the side of the road. I want to use these as my monitors for my low budget studio. Obviously they won't be as good as proper monitors, but they should be better than nothing. I am thinking that as they are old, perhaps the manufactures didn't try to colour the sound too much maybe?!

I pulled the drivers out of the boxes as I was painting the front of the boxes and I noticed that whilst the tweeter has a 3.3 uf capacitor on it, there is no inductor on the woofer. I thought it might be better to have an inductor on the woofer. The woofers seem to be 5" and the tweeters seem to be 2"

The Woofers have 481121 E written on them and the tweeters have TWEETER 8Ω 0726Aa Japan 8N22D on them. I looked up on google and there was nothing came up.

Would I get a better sound for mixing with some inductors? I don't have much money to spend as I'm a stay at home father with two children, so there is not much of a budget for this.

By trial and error, I looked up on the following website and found the crossover point which seems to match the installed capacitor.

2-Way Crossover Calculator / Designer

1st Order Butterworth
5500 Hertz

8 Ohm Tweeter / 8 Ohm Woofer

Parts List
Capacitors
C1 = 3.61 uF
Inductors
L1 = 0.23 mH

HI All;

Hope all are safe and well.

After house moves and other financial commitments forcing sale of all my hifi equipment, I now have some time to get back and an opportunity to get back into hifi.

I have my heart set on hi-res streaming (e.g. tidal) and biggest bang for buck, so use of a Raspberry Pi seems logical. I also like the look of Roon.

Speakers i will source locally, used, and the one piece of equipment that survived the clearout is a NAD 3130 integrated Amp.

So for the computer end of things, I am thinking Raspberry Pi 4 + DAC + Roon.

I have a 2018 i Mac that is constantly on, which I think I can run Roon core on.

I am struggling with the options out there for a Raspberry Pi based streamer (i.e. the software options for the Pi, the DAC options etc.) so has anyone any pointers on potential solutions that offer sound quality and reliability?

Thanks

I have been scheming a replacement for my edgar 75 hz horns.

As I have gone back and forth on whether to make another horn or make it simple with a closed box (which, of course, I would obsess over to the point it is almost as complicated as a horn) there is much talk about the stiffness of the baffle with the closed box but I have read nothing concerning whether this is as important with a horn loaded enclosure.

Another question is: If one encloses the horn within a box would it be "bad" to use some of this area as the rear chamber of the horn? I know ALTEC used the inside of the flares of the 825 as some kind of filter but these areas are much smaller than the horn I am considering. Of course, additional space would be required behind the driver but this way it could be made much smaller than it would be with a separate box.

Of course I would use braces along each horn side to the outer cabinet.

You read about using three inches thick baffles with closed boxes but nothing is mentioned how important this rigidity is with a horn. Would the forces acting upon both sides of the driver in a horn make this less important?

Any counsel will be appreciated.

I'm goint to install a fuse (or two. or three.) in my KT88SE amp since I'd hate to fry an output transformer. I see a lot of amps placing the HT fuse on the CT of the power transformer, but would it be a good idea to fuse the output tubes separately in addition to this? Something like 500mA-1A, depending on inrush current, on the CT and around 200-250mA on each tube?

I have got a curve tracer from ebay. But, it neet some repairs. The collector supply seems working. The step generator seems working.

But, the beam is thick. I switched on the tacer there is a large green spot about 3mm in diameter in the screen. I turn the intensity knob to try to make it smaller but little change. I turn the focusing knob the shape of the spot changed from circular to rectanglar.

Grateful if anybody can tell me what is wrong? Is the Z amplifier has got problem??

Any helpful hints will be highly appreciated.

Eric

Hello

I was working on a broken RA 971 amp. It had blown fuses , which despite replacement, did not switch on (on a DBT - 70W halogen bulb).
Here is the schematic : https://www.vintageshifi.com/repertoire-pdf/pdf/telecharge.php?pdf=Rotel-RA-971-Service-Manual.pdf
R
However, I could hear a faint sound from the speakers, so by process of elimination chanced upon the relay RY901 and it associated circuitry. I couldnt find an obvious problem, so reflowed the solders for fuse F901, D902 and couple other passives in that path.

Lo and behold the relay started clicking again upon startup and everything was fine. The amp sounded good in operation, all the while on the DBT. Note that I am not sure what the 'fix' was - a bad solder, or the relay just deciding to come unstuck (it was in storage for a while before that).

The Bulb lights up bright (on switch on) and then doesn't quite go to fully to 0 when it selttles, even when nothing plays.
I have repaired a RA 930 and the bulb goes pretty much 'off' when that amp is in operation, but the 971 having 2x the power i assumed a higher current is normal.

Now when I plug to the socket directly, F902 blows with quite a visible spark and charring inside. For a slow blow 6.3A fuse that does seem like a lot. (note that F901 is associated with the relay)

I am scratching my head about what the issue could be. The caps including the 50v 10mF power caps are all original, and so are other components afaik.

The only other point to note is that where the +/-49V in the schematic shows, I get only about +/-39V (all on the DBT). But then the 50v power caps seem pretty keenly specified at 49v. The RA930 is supposed to have +/-37V and i get slightly less (c34v).

With the power fuses out, and connected directly to mains, the transformer outputs are about 39.5v AC.

The regulated voltages show 16.1v. The regulating transistors q901, q902 do get very warm or even hot at idling.

I did in circuit checks of all the power transistors and the driver/bias transistors that are coupled thermally to the heatsink. I checked all the diodes as well.

I am yet to find an obviously 'bad' component.

Any pointers would be very appreciated.

This afternoon I've been refining the horn in one of my projects*.

I've been doing some measurements as I go, so I thought I'd document the improvements.

Some of this information will repeat from my 'Homster**' thread, so apologies in advance for the redundancy.




The first thing is to put a resistor in series with the compression driver. This does a few things. First, it lowers the noise level. In my setup, with a miniDSP, there was an audible hiss. This is because the compression driver has an efficiency of 109dB. So it's like an Xray for noisy components upstream. On my system, the 10ohm resistor dropped the output of the compression driver by approximately 8.5dB.

In the pics above, I have four measurements. From the top to bottom we have the speaker with no resistor, we have the speaker with a 10ohm resistor in series with the compression driver. Then I repeat the process but I adjusted the output of the compression driver to match the midrange, now with the resistor in series. The difference in noise level was startling.

The second thing that the resistor does is to increase output in the bottom octave of it's response, and in the top end. This is particularly nice since it's challenging to get a lot of output from a compression driver in the octave from 10khz to 20khz. (Due to the driver's mass rolloff.)


Here's how this works: Like most speakers, a compression driver has a rise in impedance at resonance, and due to inductance. When you put a resistor in series, it affects the frequency response *less* at resonance and in the top octave of it's response. (Due to inductance.) For instance the resistor drops output at 2khz by 6dB, but by 8.5dB at 5khz and at 10khz. So once you compensate for the reduction by increasing the voltage to the compression driver, you now have a bit of 'boost' at resonance and in the top octave. (Because the resistor's impact varies by frequency.)

Last but not least, the resistor linearizes the impedance curve to a degree, like a cheap version of a zobel filter, which makes any passive components work closer to 'textbook' values. For instance, if you put a capacitor in series with a compression driver, the affect of the capacitor will be small at the driver's resonance, due to the rise in impedance. But if you use a resistor to linearizes the impedance, then select a capacitor value that's appropriate for the higher overall impedance, the filter slope will be a closer match to the ideal value, be it Butterworth, Linkwitz Riley, etc...

On face value, it might seem like this tweak would reduce the output of the compression driver. But for the most part, it doesn't. The compression drivers power handling is still the same, just a fraction of that power is going into the resistor. And it's excursion is as high as ever. The only real downside is that it's a waste of power... But I probably use about five watts on my compression driver, if that, so the waste of power isn't a big deal to me.

If you want to read up further on this, check out some of the posts on 'LeCleach on Horns' where they discuss putting resistors in series. I broke out a new thread simply to document the results in my project.



More improvements coming up...

* http://www.diyaudio.com/forums/multi-way/252549-edge-no-control.html

** http://www.diyaudio.com/forums/multi-way/151376-homster-how-i-learned-how-fix-horn.html

Hi there,

I think this is kind of an interesting way of doing this. I am installing a Hertz Mille Pro 6.5 (mpk 165.3, 4 ohm) component set-up into my 4runner and I need to pick an amp from my existing stash. I took the backs off all three amps and was surprised at what I saw. I guess largely because I do not know exactly what I am looking at sans the caps and power supplies.

Can you help me pick one on 3 criteria (just for fun)?
1 - Based on looks of the internals?
2 - Based on the specs?
3 - Knowing all what would you yourself actually run?

I will be using the hertz crossovers and may think about going active in the far future since I have the amps.

The three amps are:

Crossfire CFA 142 (35 watts x 2, 4 ohms).

RockFord Punch 100.2 Gold Cap (50 watts x 2, 4 ohms)

MTX Thunder TA404 (35 watts x 4, 4 ohms. Bridged: 140 watts x 2 4ohms.
This is amp is brand new (old stock) btw, I just pulled it out of the plastic, if that matters.

Thanks!

I have an old favorite amp I use for a pair of slightly modified Canton Plus C sub/satt setup in my bedroom.
It is an original Audionics CC-2.
The caps are more than dried up now and causing me to unplug before total failure.

I have a neurological disorder that causes my hands to shake when trying to do precision work. (Cannot even use my turntable anymore)

I am in Southern VA US.

Any of you within a reasonable drive of DC/Roanoke/Raliegh-Durham who is good with a soldering iron interested in the project?
The cap replacement is very straight forward, but I do not want to end up with shaky globs/damage trying it myself.

Hey guys, I just finished a diy ribbon mic and it sounds sweet!!! Compered it to some other brands and I really like its character.
The only downside is I need to cranck up about 8-10db more then the other ribbons I compared with.
I am using a self wired transformer (I will provide with firther info about the core and wiring), 52N magnets and for now I am experiencing with 1.8-2.5u AL foil . I get an avarage number of 11-13 corrugations(maybe to little?) on a 5cm long, 0.5cm wide ribbon .
I guess I just need a clue on what should I be looking next to try and increase the output gain.
Will apreciate any help 🙂😎

I stuffed the Aleph J boards today and did a quick mount in the chassis for a test.

Adjusting the bias for 350mv, then 400mv is no problem on either board.

When I connect a speaker, sometimes LED3 will dim and LED2 will go bright. Then consistently maintain this state if I turn the output of my source up.

If I turn the source output up to the extreme, I'll hear some muffled, highly distorted sound. I may not have heard it at all but my speakers are very efficient. At the same time, LED will blink - seemingly with the amplitude of the input signal.

The behavior is exactly the same on both boards - so whatever mistake I made, it's consistent 🙂

I don't know where to start troubleshooting, and would greatly appreciate help.

The only test tools I have are some inexpensive digital multi-meters.

Thanks in advance!

When I became aware of the F4's ability to be driven by a main amplifier's output, I started wondering if I could use it to drive a wider range of speakers with my SET - and still have the SET character?

I'm not sure how significant this is, but the 300B is a Tubelab SE.

Aside from 300B's any descriptions about the F4 being driven by low power SET amplifiers would be great! Please include your speaker. Thanks!

Hello everybody,

I've jsut got a webinar invitation for some 3M filters:

http://www.filter-equipment.com/uploads/1/2/2/8/122858212/3m-high-flow-hfr-series-data-sheet_1.pdf
Look at bottom right of page 2. That one has the purpose to increase the filtering surface area of a flow passing through the filter, but...
Imagine that the flows doesn't pass through the membranes but tangentially... imagine that is not a membrane but a sheet of Alluminium, imagine it is air and not water and imagine it clamped on the magnet of the cone.

Has anyone ever done a "clampable heat sink" made in the 3M HFR way, to be clamped on the magnet and with the same diameter of the speaker?

The purpose is:
- increase the heat tranfer coefficient (when the cone moves, the convection goes from natural to forced (by the cone moving), so increase the capability to transfer heat;
- increase in the surface area of the magnet, so increase the capability to transfer heat.

This will give two scenarios:
- lower temperatures for the magnet, so higher reliability for the speaker;
- higher allowed power handling.

Has anyone an idea on how (badly) it would impact the sonic performance of the speaker?

By doing the "radiator" in Alluminium and thin, it will give optimal passage for the air, low weight, low costs and less sonic impact, but how can it be calculated?

Thank you all in advance
Roberto

Looking to add four ceiling speakers to my 5.1 to make 5.1.4 for Atmos.
I want to do this on a budget though, since I'm a couple years away from a total home theater build.
It seems the Atmos ceiling speakers don't need to work below 150hz and should be somewhat non-directional.
At $100 for four speakers, any reason not to try the 4FE35's?

I don't have Atmos experience yet, but based on what I've read, I don't understand why some people are going with large speakers for the ceiling.

FaitalPRO 4FE35 4" Professional Full-Range Woofer 8 Ohm

Hello good people!

Thank you for any time you're able to put towards this in advance.

I have a Crown 5000VZ with an unusual problem. It doesn't pass sound, here's the LED situation
ODEP - off - both channels
SIGNAL / IOC - ON, GREEN - both channels
Load / Limit - ON, RED - both channels
DC/LF - ON - both channels

The service manual doesn't cover this combination, but the closest one listed suggests that 'the output stage has been damaged in both channels' and that is it.

Things I have done
Checked the flyback diodes for shorts - D596, 507, 508, 509
Checked the driver and pre-driver transistors for shorts and opens
Checked semi conductors 8188-2 and 8187-4 on all output boards
Repaired numerous dead traces on the output boards (corroded by some obscure green compound)

Would really appreciate any help you good people can offer!

Thanks

Hello everyone!

I develop the reclocker for Raspberry Pi.
I did I2S dividers for BCLK and LRCK.
For testing I use the Raspberry Pi 4 and Volumio OS with selected HiFiBerry DIGI as I2S output.
I connect the Raspi I2S output to my DAC board AK4490, in finally I got corrupted audio.
All wires for I2S connection is very short (not length as 10 cm).
My questions is:
- Which I2S DAC I need select in the Volumio for get native I2S?
- Maybe needed make a configure the I2S driver via I2C or SPI interfaces?

I recently picked up a pair of vintage Weston VU Meters from a Quad Eight.



I am hoping I can add them to an Aleph J that I am building. (I previously built and F5 without issue.)

Has anyone implemented vintage meters of this type? I'm wondering if I can use this PCB board to do it.



I can run 15v from a small transformer or add a regulator and use the 24V from my existing power supply.

One thing I'm unclear on is where this could go in the chain from preamp to amp to speaker outputs. In other words, what is the input to the VU board?

Will it have any negative input on the sound quality?

This is purely for looks.

Hi Everyone,
does anybody happen to know what output devices exposure use in their amp's, specifically the Super XX, i've just shorted the o/p so mine are no more! there are three per channel:
EXP002N EXP002P EXP8N ch1.
EXP002N EXP002P EXP8P ch2.
any info greatly appreciated
Mark

Hi.
I'm building a 4ch ucd400oem amp (driven by two smps400a400) + 4 channels of ucd250lp (one smps400a180).

I made input buffers pcb and all, that part seems to work good (exept one of the ucd400 input buffers that makes the connected module ticking (sound from the speakers) about once a second, no other sound than that), its like some protection is kicking in, cant get whats wrong with it, maybe I'll have to rebuild a new one from scratch.

And I have a problem with one of my ucd400oem modules. it is dead silent, no sound what so ever. I bought 4 of them on ebay and the other ones works as expected. If i connect one working ucd400 oem to the input buffer then it works, switching place with the ucd400oem that not works and there is no sound at all. So there have to be som wrong with the module.


I also have one smps400a400 that has negative rail shorted to gnd. measured when not connected.
I have just sent a question to hypex regarding the smps but I know it will take a while before I get an answer so I thought I could ask hare also. What could be wrong with this one? I have tried to see if there are some metal pieces on the pcb shorting the - and gnd but could not find anything.

This really off putting because I really wanted this to work, already spent a lot on this both time and money. Any help would be appreciated!

some pictures of my input buffers.



Thanks /Joel

Hi,

Just received and built my nc122mp amplifier, in a ghent audio case.
For now, the amp is always on, but I would like to use a trigger input.
I'm not an electonic guru, so I'd like your approbation before doing anything ^^

I undestand that I have to deal with the ncore module H-bus connector.
Need to cut the J6.3 (PS standby output) to J6.9 (PS Enable) bridge, and add kind of relay that will be open/close according to the mono 3.5mm jack input?

My preamp doesn't have trigger output, so I'll use a wifi/IR outlet with a 5V power supply plugged in on it to trigger the amp.

Any recommandations ?
Thanks

I am looking for a cheap simple system for the kids to play around with.

A quick search gives party speaker systems at different prices but you have systems as The box PA 15 DSP as well. A Bluetooth speaker is easy to use but membrane area adds some significant punch in the sound.

Are there better alternatives than the PA system above in that price range?

I think a diy project will be more expensive even if the sound could be better with proper designed speaker cabinets. Only the speaker element will most probably be more expensive than these cheap active systems.

Hey guys,

Hope you’re all staying safe and well.

I’m using a Cyrus One V6 solely as a phono stage in my setup. Naturally it is quieter than let’s say a CD input but it still seems a little on the quiet even as a phono stage. I love the quality of the sound and I’m wondering if any of you guys know of any “mods” to squeeze a little more gain?

Thanks guys

I have used a number of SMPS to power my music server over the years, but none of them has got close to my Paul Hynes SR7 in the high frequencies.

Some computer power supplies, such as high end corsair, and the AX1600i I had have very good voltage regulation and ripple suppression - the AX1600i sounded to me as identical in the lower and mid range, but rolled off / unsatisfactory upper band.

I use Hypex SMPS in my amplifiers and these are completely perfect accross the frequency spectum to my ears.

Has anyone else experience this, and any thoughts reasoning why these computer smps sound rolled off in the high notes, yet the Hypex power supply doesn't seem to do this.

Thanks

Hi,

I found in flea market very large number MOSFET from ex CDMA/GSM BTS or a brand new one from datasheet I know the have 60 to 300w in 1800-2200MHz because the price is very cheap. from 1USD to 10USD I been thinking to use it in Pass design amplifier. but before I regret my idea if some one kind enough to give me idea or suggestion about which scheme and parameter I must use Vmax, I max etc
some of parts I already have is,
MRF6S18060NR1
mrf8s9260hs
MRF8P9300HS

All idea is appreciated.
Anton

Found these in my stash. Are used but measure fine at 435u, 439u and 105u

Make me an offer. Shipping can be with letter (cheap but no tracking I guess) or package (costly)

Collect in Utrecht NL

YouTube

My perspective is based on my feelings, after seeing this video. Loudspeaker design and Mastering go hand in hand. If we take Earl G and lock him in a room with Bob Katz, for a year, though they represent decades of research, both would potentially "master" each others material quickly, because its so closely related.
Mastering has made me a better loudspeaker designer and loudspeaker design has made me a better mastering engineer.

I hope this turns into a discussion about what needs to be done to advance both sides of the fence and how, in 2020. Also mentions of Standout products or Practices.


I named the post as I did because that is the formula, in my mind to move forward....The audience (general public) is less important than is being treated in this video, The audience is "uneducated and incorrect" and we are educated and way more correct. Floyd said it, and I am witnessing, the consumer products move closer to mastering monitors and not visa versa which to me is a slow showing trend of the audiences (general public) audio intelligence increasing causing their expectations to become more like the Mastering engineer. FLoyd shows how they did their double blind test and I see a larger portion of casual listeners than critical listeners ...that can only makes sense because one group has a much bigger pocket book.
I thought.....I thought.....I thought we were moving away from the smiley face voicings and "max bass boost" type systems. The General public will lead you to more non sense if left unchecked....just kick them out of the picture so early in the game. Loudspeaker design, mainly, should be left up to Mastering engineers and Loudspeaker engineers until the later stages, like we have 10 models that we screened out, we'll use the general public to figure out what they like out of what we find acceptable.
So why, why do I say this? Mastering Engineers are the Ultimate Critical Listener! We have the ears! Our job consist of creating the most neutral playback, listening, and adjusting, to create an emotional, dynamic, tonally balanced presentation from scratch. This process is almost identical to the objectives of loudspeaker design but a mastering engineer repeats this process, from scratch, many many times over in a week. How many loudspeakers can a designer create from scratch and go through the hoops to completion in a week? So at the end of the day, Mastering engineers have the experience advantage. If a speaker is perceived well, is a question that should only be answered by mastering engineers since we have the ears. Loudspeaker designers seem to need a group of people, to make decisions. Making these types of decisions are what Mastering engineers do on a daily basis, by their lonesome , and whether they are right or not, is feedback that can be discovered within a short time frame. So not only do we constantly listen and manipulate the signal, several times over, more times in a week, than the loudspeaker designer can be challenged...we are also obsessive with being right, so we are constantly checking our work, within the week, to see if we are "right"......if we are right about our choices voices versus what we hear....we have successful mixes that translate from system to system and successful careers in the field....if the loudspeaker designer is "wrong"....only the mastering engineers and other niches of high critical listening care!!!! How is this possible!? A good production sounds fine on un-perfect speakers! Creating these types of mixes that sound good on every additional system post the studio...cannot be done on consumer level speakers...so not only do Mastering Engineers have the ear training, they have it based on systems that are an attempt at perfection, designed by loudspeaker engineers.

So the advancement of the field of Perfect Speakers should only be pushed forward by the people who are the main consumer of perfect loudspeakers who conveniently happened to be the same people who design the signal in the first place, and thus need perfect speakers regardless of preference, Mastering Engineers....

Go

Hi all,

I'm an IT guy, 37, my father is the big guru (built couple of tube & solid state amps) but we're not living together and he almost doesn't have any more time to teach me on basics of electronics or hi-fi or whatsoever.

- Ca. 15-20 years ago my first speaker set was 2 fullrange boxes made of MDF, I cut the holes, drivers in, glued together - perfect. Then blown with Prodigy at that time 😀 - well, youth .. you know ..

- Ca. 15-10 years ago my 2nd project: 4 boxes, 2 left, 2 right, all mdf. (3 way, separate boxes for Bass and mid+highs). 2x12" (30cm) bass drivers / side, 4 Ohms each, in series, with front-facing bass reflex holes. Mid+high in a separate box on top of it.

- Passive filter for the 3-way was set-up by myself, based on my father's instructions. He calculated me the 2 basic X-over freqs matching my speakers and I had to solder them together and wound the coils also by myself from his copper wire, using a ferrite core (for the bass coils only).. 3-way classic Butterworth design.

Still using these huge boxes, but the speakers need to be changed. Mids are from Visaton, silk dome ferrofluid tweeters from Audio-Ton ( = almost noname, sound excellent however..) .. bass drivers you can forget .. cheap whatever types..


Now I want to sweep the table and start all-new again with a far better project.

What I have:
- precision soldering iron 😀
- contact to a local company building high-end grade extraordinary speakers, also for hobbyists, partly or fully .. I'd have them build my boxes. BIZSÓK Loudspeaker Manufactory
- my ultimate-looking wanna-own speaker design http://www.canhifi.com/Resources/ProductImage/ImageFile/beethoven_concert_grand_1_xtralarge_xtralarge_XtraLarge.jpg
- a 3-way active crossover design I'm going to use from Linkwitz Lab - Loudspeaker Design, fine tuned to my needs with help of my father..

Where I need advice on:
1. High sensitivity, detailing, fast, audiophile-grade speaker driver recommendation (exact models maybe?) - or at least a good store ! For bass, mids and highs.
2. No big magic pls, I'm rather the realistic type of person, not believing in placing spikes below the 230V/AC splitter 😀
3. Some questions regarding tri-amping with different DIY amps

Target music: sadly I'm almost an all-eater but don't want to have 2 separate systems..
- electronic music (Goldfrapp, Air, Morcheeba, Asteroids Galaxy Tour, Jean Michel Jarre)
- Classics (Karajan Gold collection on LP, mint condition)
- Rebecca Pidgeon, Loreena McKennitt, Dire Straits, Mark Knopfler, Diana Krall CDs, SACDs, LPs and high-res DSDs
- Deep house, techno, electronic music (all lossless)


For topic #1 my imagination so far:
- 8 Ohm total. 2-4 bass drivers, 1 or 2 midrange(s), 1 tweeter
- tweeter would be a ribbon type, I LOVE their sound. DIY amp design would be carefully selected to be able to drive such tweeters with transformers.
- mids: no idea. Something from Visaton, Seas, Fostex, Scan Speak, others.. ?
- bass: no idea. Maybe 8-10" ones ? For more power handling I plan 2 pcs here (at least) - does it make sense to have 4 pcs ? (Does it increase overall sensitivity and detailing capability of the bass section while retaining decent power handling ? I hate slow bass..)
- all 3 ways at around 90+ dB sensitivity, I would like to hear details
- 80% stereo music, 20% home cinema use (2.0 stereo, maybe 2.1 but no sub planned yet)

Topic #3 - tri-amping, DIY amps.
- Does it make sense (or is it a completely wrong idea) of using a mixed setup ? MOSFET amps for bass, tube class AB pushpull for mids (Tung-Sol KT150-s), tube OTL for tweeters ? Considering phase coherence mostly.
- Gains matched, in the amp design and also on the active crossover network (matching the parameters of the driver sections of course)
- Where to put the volume controls ? (Stepped attenuators). The more towards last amp stage or right at the beginning at line level in the preamp ? (Before splitting into 3 ways).
- Passive preamp OK ? (with XLR Input from high-res DAC)

Any advice or useful info warm welcome. Please forgive me for my English and try not to laugh if I seem to dream big with so few knowledge about electronics. 🙂 It's always that I dream and my father helps me to put things ready onto the table. 😉 I'll need him for this project too but without your valuable help it would still remain just a simple oldschool design without any traces of quality. I really strive for quality here (combined with decent power), at least as far as a hobbyist I can go..

Many-many thanks !!

Hi All,

I hope all is well out there.

I love my Schiit Saga (R2R), I basically want a networked version of it (I use Roon for my music almost exclusively now), something like the ESP32 controlling an R2R or a LDR based volume.

I would want the firmware to support an interface that allows setting and querying the volume level. Home Assistant automation software has ESPHome which allows you to connect ESP's with automation easily.

I have seen RelaiXedPassive, LDR Pre MkII and some other bits out on the internet but nothing seems to be quite what I need, is anyone else doing this?

I'm happy soldering surface-mount, ordering parts from DigiKey and I have used Gerbers to order boards from PCBWay, so the DIY route is all good, I've never designed a board though.

Thanks for looking.

Cheers
Richard

System: Roon (PC) >> Pro-Ject Pre Box S2 (DAC) >> Schiit Saga (R2R Volume) >> Valvet A1r's (Amp) >> Salk SongTower QWT (Speakers)

helo good folks, I'll get right to it. I don't know much about Audio and such but I was wondering if there is a mixer or something that exists that allows me to change the volume levels of individual programs of my choice, like a knob/slide for Spotify and one for chrome, etc but also have USB inn for my mic and a knob/ slide for that. I have managed to use macros and an external Numpad to use for volume control on Spotify but its kina janky some times and I was wondering if there was something better and I'm kinda interested in learning about stuff like this. Also, I have SteelSeries arctic pro wireless headphones (Amazon.com: SteelSeries Arctis Pro Wireless Gaming Headset - Lossless High Fidelity Wireless Plus Bluetooth for PS4 and PC: Electronics)
and was hoping I could still use them with the mixer or soundcard (again I'm a noob don't know what the ting would be that is why I am asking). I was imagining something like the audio going from my PC into the mixer/soundcard then into my headset and also the mic into the soundcard/mixer or something
Thanks

Hi everyone! This is my first post on this forum 🙂

I have recently been given a pair of quad 99 monoblocks.

Two questions:

1. The speaker terminals on one is kind of broke.

I am thinking of opening it and soldering a new pair of terminals.

I have soldered RCA cables before. That worked great.

Is attaching new speaker terminals doable för a beginner with basic soldering skills? I am kind of assuming it is very straightforward, just soldering wire to the terminal.


2:

Googling this site I found a member stating the following about Quad 99:

"I bought a new Quad 99 as a local shop here is selling for less then 500 USD. I hv solder a direct wire from the RCA input to the amplifier board, and eliminated the messy Quad link circuitry that killed the sound."

That member no longer seems to be active, but the post intrigued me. Just soldering a wire from one place to another doesn't seem that hard, and if it can improve the sound...

Does it sound like good/bad idea? And more importantly - would anyone on this board be willing to help me identify the points to solder when I open monoblocks?

thanks in advance,
Carl
Sweden

Hi there!
I heard a rumor about Alpair 10.2 drivers in droves being spread around Sweden.
Is there anyone in Sweden who is sitting on a pair or two they don't need?

Peace!

/Hilding

hi!

I got an old video recorder, that was stored in a cellar for decades.
A Sony SL-C7 for Betamax tapes - I never had β before! 🙂

I worked with many different tape types as long as I live, but mostly VHS and S-VHS.
And I've been interested in some uncommon formats like "VCR" or the better known "Video-2000" for example.
Unfortunately, I missed Beta! 🙄

Video on youtube: "Sony Betamax C7 - Video Recorder Repair"

YouTube


And here some Betamax video footage.
My 90 years old grandpa and me in the garden, Easter 2020.
Filmed with an attic found Sony Betamovie BMC-100P on Betamax video tapes - just to enjoy the pure analog visual quality!
Much Better than I expected for 40 year old video equipment!

So this video is filmed on the Betamovie, then played back by the C7 recorder and after all digitized with a today's technology DVD recorder.


Video: "Easter 2020 on Betamax video tape / Sony Betamovie BMC-100P footage"

YouTube

I've got 2 NE95W-04 fullrange drivers to give away. I will cover the postage so they find a home.

One of the drivers has a small 2mm dent in the dust cap as can be seen in the pic. These were taken out of a small Dolby Atmos speaker I built using dual drivers and now have changed the design to single driver with PR so I no longer need them.

Suggested box size 2.4 litres Fb=75Hz. For best sound balance a LR network is required to give some BSC. For 2dB use 1R + 0.2mH, for 3dB use 1R5 + 0.3mH.

In the near future I will be giving away a TPA3116D2 module as well as a Dayton DTA-2 module and both work well with this driver.

When I use Wondoes to run my system, I have Equalize APO installed and active. Recently, I began experimenting with its ability to run "VST plug-ins". Particularly searching for a dynamic range compressor that just happens to have expansion capability.

The reason for this is I found a radio stream I really like (Retro Radio California), but by Listening I have to say they compress the ---- out of their audio broadcast.

I realize with all the possible variables that can be used in compression, getting back to the exact signal originally fed into the compressor is a fat chance. But somewhere toward that end is better than nothing.

I found such a beast at REAPER | ReaPlugs. I downloaded the "VST FX Suite" of various VSTs he offers for 64 bit - the system bit width matters - unzipped it and installed the ReaXcomp VST. It works to give the compressed music a little more punch.

Easily overdone, you have to be conservative with the settings and accept the most you can do (before unwieldy) as better than flatter-than-pondwater dynamics. A/B testing; it's definitely doing something.

Anyone here using ReaXcomp inside Equalizer APO for dynamic range restoration? (It'd be nice to share settings, which are a little daunting) And of course, it's be great to get this functionality inside Daphile, so I dont have to boot "Wondoes" just to listen to the radio - with a little dynamic range

Thanks!

I have scoured old threads on dynamics to understand what makes a speaker dynamic and its seems there still isn't a general consensus on the topic. A few points do stand out more than others.

So am curious what would give me more dynamics? Multichannel or PA style speakers?

Of course it would be specific to what multichannel and what PA speakers so lets break it down. But I am up for a general discussion and opinions as well...

Multichannel would be Troel's 3WC which has an 8 inch woofer 3 of these at the front with say a smaller Troel designs with a 5 incher for the rears that are voiced similar. A Marantz pre pro with Hypex NC122 amps and dual subwoofer(pretty descent one RSS315HO with a 900 watt Class T/D amp)

VS

Active crossovers with a Visaton DR45N Compression driver in a horn loading probably something average that goes down to 800hz/900hz with Eminence Delta Pro 12A and the same dual Dayton RSS315HO subs. LEts say for arguments sake the Okto DAC8pro with the same Hypex amps.

Now, what which system would give me that guitar popping out with into my face?

I’m planning a hybrid open baffle build with a 10” full range located about 36” from the floor on a flat board. The board will be held up vertically by attaching it to a box about 1cuft (28 l) that also serves as the case for an 8” sealed woofer.

Each speaker will be located about 4 ft. from the front wall and 18 inches from the side wall in an 11'x12’ carpeted room. I plan to crossover to the woofer at about 200 Hz.

Question is whether locating the woofer driver on the front, or back, or side, or top of the sealed box will matter. I’d prefer not to put in on the front, but would do so if it would make much difference to the sound in the room.

If not on the front, is there a preferred orientation that would provide optimum bass distribution in the room.

I built a new crossover for my Dayton B652 Air bookshelf speakers. At the time it was well known that the treble was hot and it was common to increase the tweeter resistor to compensate.

I used one speaker constantly and the other was used a bit but not much.

Now I measured the frequency response again and the treble is down 5db. The DC resistance has not changed. I measured the unused speaker and it is the same way! I tested my microphone, as far as I can tell it works the same as it did when I made the crossover.

What this means is that either both woofers have increased in sensitivity by 5db (unlikely as sensitivity is set by the mass of the cone, strength of the magnet and number of turns in the coil), or the tweeters have lost 5db of sensitivity.

Now when I change the tweeter resistor back to the original 4ohm value, the sensitivities are matched.

This suggests to me that Dayton already knew these tweeters would lose sensitivity over time and that is why the treble is hot out of the box. Furthermore I don't think they ever explained this to anyone, although they presumably easily could have.

Has anyone who purchased a Dayton mini-8 AMT measured an initial sensitivity higher than what is in the specsheet?

Puget Sound Speaker Contest 2020 - Canceled!

The Puget Sound! DIY Speaker Contest - Canceled for 2020

The Contest will be held at the Mercer Island Congregational Church, 4545 Island Crest Way, Mercer Island, WA on August 8th, 2020.

Doors open at 9:00 a.m., and judging begins between 9:30 and 10:00 a.m.. We may be flexible on start time if a contestant is traveling some distance and may be arriving later than the start time; please let us know in advance so we can schedule accordingly.

Entries
This year there will be two "classes" of entries: original DIY designs and kits. We realize that a lot of effort, experience, skill and sometimes luck are needed to come up with a successful design. Many accomplished designers learned their craft by starting with kits of one sort or another and subsequently progressed toward original designs. For others, the lack of time or experience makes a kit a means towards better, affordable sound.

A Kit is defined as...
...Any pre-designed or published design (publication includes the Internet) that is the product of another person. This may be plans, parts, kit, or both plus cabinets, etc. Any major deviation from the plans may at some point constitute an original design -- however, a club appointed technical committee will have the final determination if this problem presents itself. All kits must have been offered to and available to the general public at some time. We rely on the integrity of the contestant!

The kit class is broken by cost per pair, and excludes the cost of the cabinet and finish. The breakdown for kits is as follows:

$0-$175/pair
$176-$325/pair
$326+/pair

Original Designs:

The breakdown is as follows:

One way ----under $75/over $75
Two way ----under $200/over $200
Three way -- under $300/over $300
Unlimited ---- no limit

NOTE: This is for drivers and crossover parts only; cabinet material and finish are not included. We again rely on the contestants' integrity. Your uncle who owns the HiFi Emporium cannot give you a special deal that isn't or hasn't been available to the general public.

The speakers will be driven by the system the club supplies. Don't bring your own MegaWatt Hyper Triode Amp -- we're not going to use it. To keep it simple, we are not going to allow biamping or plate amps for a bass module.

The entry fee is $5 if you pre-register two weeks in advance or $10 if you register at the door. If you have any questions please post on one of the forums with an official thread on the contest, like this one. Also, it would help us a great deal if you'd let us know or think you're coming. A downloadable format entry form and the address to send it and your entry fee will be provided at a later date.

Judging

The judging will be by a panel of judges in a special room set aside for this. We will have three judges, which in turn plays his personal pre-selected musical selections (2-3 minutes max.) while sitting in the "sweet spot." Each contestant is allowed to specify general guidelines to set-up, that is, distance apart, toe-in, although corner loaded speakers will need to provide their own artificial corners. Remember that the judges will be sitting side by side so that should be taken into consideration.

We intend that judging will be totally blind -- this means that a curtain is utilized to ensure that sound, and not appearance, will be the sole criteria. All speakers will have the loudness adjusted to the same level before judging begins. Set-up of speakers will be by a club crew using notes taken at time of entry. Each judge will be issued a standard evaluation form prior to each session, and will rate each entry by assigning points, after which the form is collected by the Technical Committee member assigned to monitoring the judging (and setting SPL levels prior to each evaluation of an entry, etc). The forms are then scored by the Contest Director, or his designee for total points. After the final entry is judged, the points earned are assigned within each category to determine ranking within that class. The total overall points regardless of category or class will determine Best Sound of Show.

Other Stuff:

General admission of the public is free. We hope to have a garage sale/swap meet type offering for those who wish to sell or buy audio gear for free. However, the owners are responsible for their gear, etc. Don't expect to drop off stuff and return at the end of the day to pick up the money. (Please advise in advance if you wish to sell items).


Awards:

Each entry will receive a certificate of participation and a print out of his entry's frequency response measured by a qualified expert designated by the Contest Director or head of the Technical Committee. Additional certificates will be awarded to class winners, as well as 2nd place .

Finally a "Best Sound of Show" certificate will be presented to the overall winner. All winners of their respective DIY class are eligible for this. Please note that "kit" speakers aren't eligible for the Best Sound of Show.

Unless overruled by the Technical Committee for violation of the rules, the decision (total points) of the judges is final.

Best Regards,
Terry Olson

Can someone please help with circuit or notes for retro-fitting a TAOHS-JP3 with a TAOHS-KP1.
The TAOHS-JP3 is from a Yamaha CDX-810 CD Player and has an extra circuit board on the side, looks like the KP1 replacement has SMD components and this extra board in not needed.

Dear all,



I have a Carver CT-26v A/V premaplifier/tuner that I bought in a broken state. Problem was in two burned/blown resistors with signs on the board R905 and R906 the value of which could not read nor measured and I put two pieces of 2 KOhm. I Please Carver fans to help me with this problem or send a diagram scheme (service manual) of this preamp.

Another problem of the same preamp is that by increasing the treble potentiometer does not have enough treble and also increasing bass potentiometer has a filthy muddy bass. Please help me in solving this problem.

This is a part of the scheme which I read/registered from printed circuit board and by the way measured voltages:

Hi guys. I don't want to do this but the wife says time to sell some drivers I've had for a while that I was planning on using in projects. They are all brand new, never used in the boxes. I will accept payment via PayPal, and ship world wide. PM me if interested.


Pair of Legendary Focal Audiom 7K2 7" mids $300 for the pair.

(notice how bright yellow the cones are. You can tell they been sealed in a box because the Focal Kevlar mids get an orange patina after a while when in open air)



2 pairs of Focal TC120TDX2 tweeters. $300 per pair.


Pair of beautiful PHL 1120 7" mids $300 for the pair



2 pairs of PHL 1240 7" midwoofers. $250 per pair



1 pair of Eton 8/800/37 8" mid woofers $250 per pair



1 pair of Eton 12/680/62 12" woofers. $750 per pair.




I also have some 7" Accuton midbass's. Various other brand new Focal 5.25" midbasses as well. Let me know if interested 🙂

Thanks,
Mike

Hello all, I have some questions involving headphone amplifier power and relation to volume control.

A little background: I am building a tubelab SE-II, which supports 45, 2A3, or 300B output tubes. My goal is to build a headphone amplifier out of this circuit. When gathering opinions on what output tubes I should get, everyone said go 45, because they have less output power, and the wattage of 2A3's and 300B's (about 3.5w and 8w respectively), would blow out my drivers and my ears.

But here's the thing... There are tons of tube amps out there that output 5+ watts of power into headphones, including the ampsandsound Mogwai SE I have, and I've never had a problem at all with "blowing out my headphones". I also confirmed with Justin of ampsandsound that there are no resistive networks, and that the full output power of the taps are put into the headphones, but volume control is used to attenuate to the listener's desired level. So I don't quite understand the worry from folks about blowing out my headphones with more powerful tubes. Maybe they assumed that I wouldn't be using volume control in my amp?

Anyway, Here are my questions:

1. If a headphone amp has the capability to dump 5w into headphones, would the headphones only receive all 5w if the volume pot was set to max volume? That would explain amps that can output many watts, but don't damage headphones.
2. If yes to the previous question... what's the point of having a headphone amp that can put 5w into headphones only to have it reduced significantly by a volume pot?
3. Is there any detriment to sound quality from using only a partial amount of the input signal (from turning down the volume pot)?
4. Does this mean most headphone amps can be made to output MUCH less than 5w and still be able to drive "hard-to-drive" headphones, with the only side effect being that the volume knob has to be turned up higher?

I'd appreciate any input on this. I am new to diy audio projects, and there are still some gaps in my knowledge.

Hi all.

I built a single channel push pull amp from the Dynaco 10W schematic found everywhere.


Also used some simple 15W output transformers designed for guitar amps without UL taps so had to run a SG resistor off B+. That OT claimed primary impedance 8k and output 4ohm. My speakers say 6ohm.

I fed B+ with 300V and the thing sounded fantastic as per that schematic.
I can't remember what the value was but I fed the EL84 screens with b+ across a resistor.

So now I am in a position where I want to build a stereo version of this amp and in a chassis rather than just bread boarding. I have a few questions.

1. I want to feed the screens with B+ across a resistance, do you think the 100ohm is sufficient in this case?
2. 1W in that position or should I be going 5W?
3. I've dropped the resistor feeding the 12AX7s. This is of course because I will be feeding 2x 12AX7s so the increase in current meant I needed to decrease the resistance but I went further to increase the voltage going to them. Is this too much now? I was hoping it may increase my clean volume.
4. Should I also drop the resistance of the 100ohm 10W resistor and use a 20W+ resistor in this position as it is now feed both amps? Unless I separated the feed from the rectifier into two power filter boards, one for each side of the amp?

The power transformer I want to use is the Hammond 260E.
260E 80VA 450V C.T. @ 115ma. 5V @ 2A 6.3V C.T. @ 3A
The 6.3vAC tap should be sufficient to run the EL84 heaters and I plan to rectify the 5VAC into DC which through the solid state rectifier should come out at 6.3VDC? Throw a cap across that and send it straight to the 12AX7 heaters?
Below a sketch of what I am thinking. Any comments in addition to what I have mentioned above also welcome.

The Little Bear B4-x is nice sounding tube hybrid headphone amp. It's powered by a 12V battery with 2 DC-DC converter (Mornsun A1212S) into +/- 12V for each channel. It uses two of the 5899 subminiture tube and I traced out the tube circuit but my knowledge is far too basic to understand this particular pentode design. Since I like the sound of the amp, I would like to see if I can improve it further by upping the voltage to either 24v or 60V using addition DC converter. But what change should I made to it?

I switched to ss rectification and no longer need these, they all worked the last time I plugged them in, shipping in the continental US is $5 for one tube, add $1 per tube for more than one, all 4 ship for free:

EDIT: For some reason the photos are compressed, if you click on them they appear correct.

Mullard/Hammond GZ34/5AR4, I purchased this as NOS (early 1960s) from Vintage Tube Services 4 or 5 years ago, I put about 1500 hours on it, no box. $40.



Ken-Rad 5Y3G VT-197-A, (coke bottle) I purchased this as NOS 3 years ago. I put about 100 hours on it, no box. $12.


Sylvania 5Y3WGTA, I purchased this about 2 years ago. I put less than 100 hours on it, box is rough. $12.


Sovtek 5Y3GT current issue, I purchased this new from AES 6 or 7 years ago, I put about 300-400 hours on it. $10.


I am cytovette on eBay with 594 positive feedbacks, never a neutral or negative.
Thanks

Hi,

I've built an integrated amp with phono stage using Peter Daniel's Audiosector LM3875 kit. It uses a rotary switch for signal switching and an Alps blue pot for volume control. I've got it working and sounding great (phono now works thanks to some help from this forum).

However - I have a Boozehound Labs JFET Buffer stage board sitting in my case, and I want to put it in the signal path to hear how it affects the sound. I've been playing with placing it after the pot, but can't get rid of the hum. I'm sure that I have a grounding scheme issue when I put the Buffer into the mix.

Here is a general description of my working integrated amplifier circuit:

Signal Scheme
- 4 RCA inputs
- 3 pairs of inputs send signal to rotary switch --> common pot --> LM3875 input
- Signal ground from the 3 pairs of inputs above to SG pad on LM3875 PCB
- 1 pair of inputs send signal to RIAA preamp PCB (http://cdn.shopify.com/s/files/1/0289/4153/files/Boozhound_Laboratories_JFET_Phono_Preamp.pdf?33) --> rotary switch --> common pot --> LM3875 input

Signal Grounding Scheme
- 4 RCA input grounds
- 3 pairs have each L and R channel signal grounds connected and sent directly to SG pad on LM3875 PCB
- 1 pair has grounds separated and sent to RIAA preamp PCB. Bottom of PCB is a grounding plan. 1 ground wire from PCB is sent to star ground (not SG on board) and this results in a fairly quiet configuration.

All power grounds go to common star. I'm using +24V regulated DC to power both RIAA and Buffer PCBs.

This is the buffer I'm trying to integrate: (JFET Buffer kit – Boozhound Laboratories)

My thinking is that I might not understand the signal ground return path properly and that maybe I'm creating a loop. I would have thought that I can run the signal ground from the RCAs directly to SG on the amp, and pass the signal itself through the various stages (switch, pot, buffer, amp). Maybe there is an interaction with the pot and buffer ground? Any thoughts are appreciated.

Martin

Im looking to buy the revised versions of these units.

Seller would need to pack up to send to New Zealand.

Any voltage is OK, but multi-voltage units preferred.

Hi forum

I have a Bryston 4bSST2 that makes a hiss sound. Not trough speakers, but the amp itself. Regardless if input signal are connected, speakers, grounded or ungrounded circuit.

The hiss is in approx. 12 kHz region, and seems to be on the input PCB. I have to turn off the amp when its not in use as its very disturbing.

Any ideas are very welcome.

Thanks.

Schematics at official Bryston homepage:
http://old.bryston.com/PDF/Schematics/4BSST2_SCHEMATICS.pdf

Hi,
?I would like lots of info on how is the 807/1625 sound?
Likely will be Russian G807 or the 12V G1625, input stage is 1 tube only 6SN7 or 6SL7, tube rectification, output is one single tube in Tetrode Single Ended A1 about 10W w/low NFB to drive one full range 94dB,8 ohms.
Thanks for help:00

Hi

Would someone know where I can buy these type of 1/8“ jack ?

Thanks
Eric

Pth1010hf-1 new pair unused.

$100 (free shipping)

48 us states only

Takes 1.5" jbl

Or you can get a pair from reconingspeakers.com shipped for $150.


Here is raw response of it with jbl 2435hpl post #1
https://www.diyaudio.com/forums/multi-way/342456-advice-pth1010-2435hpl-combo.html

Hi there,

I am actually looking for an schematic for an amplifier in the region of about 10W rms. Class A or class AB doesn't matter at all. Not more because the case is not the hugest on the planet and therefore the heatsink could not be big.

Because i already own a power supply (2 trannis 25v 60va which sit in an external case with lots of capacitance) i am looking for a schematic.

Hope you get what i mean.

Best regards
Xan

Hope some one can help me out here.
I Installed Audirvana and gave it permission to shut down some computer prosesses while playing, that otherwise potentially could have a bad influens on the audio quality.... It was great, sound was great BUT!!!
After instaling Adivarnia I am no longer able to send sound to my Airport express from other sources like fx. You tube, My speakers are visible and I am able to choose them, but no sound is coming out...
I have now uninstall Audivarnia, but it did not help, and it is not the airport because iPhone and iPad has no problem streaming audio to the airport.
Please help! 🙂

Good evening Campers,

I’m contemplating restoring some Celestion Ditton 66 Loudspeakers, but I’ve never done anything like this all 😀 They are in a poor condition cosmetically (and sonically) and thought changing their appearance for something more modern might be the way to go, as opposed to trying to repair it. I’d be be using the iron on method when it comes to the re-veneering.

However, what about removing the existing veneer finish, had anyone ever done this? I’ve watched some YouTube videos using scrapers, damp towel, iron and so on, but thought I’d ask for some firsthand experience.

It’s not a perfect box, what with the edging and little lip at the bottom, so it might all prove just a nightmare for a novice to handle. I’m not a DIY’er normally but thought this might be a fun project and save passing it over to a pricey cabinetmaker to restore

Any helpful advice appreciated.

Cheers
Phil

Hi everyone,

I'm David, very much a novice in all things audio. My daughter and I have been building a pair of open-baffle speakers with vintage drivers. We haven't gotten to the point where things are working yet - something is wrong with the amp - but it's fun.

Thanks!

I got a JFET Phono/RIAA Preamplifier kit from boozhoundlabs as a gift a couple of weeks ago. This was my first diy audio project and it was incredibly fun to learn and build! Planing to do many more.

So now the bad news. I plugged it in and all i hear is feedback and hum.

I'm just curious if people have any good trouble shooting advice? Like I said, this is my first time building. Attached are a few images of my build. It's getting the right power from my supply. The left inputs are coming from my turntable and right are going to my amp. While it looks like the color on inputs for positive and ground are reversed, that's not the case.

Thanks from BedStuy Brooklyn! And very sorry for such a beginner question.

Hi, I would like to build the Elliott Sound project 10 amplifier using transistors that I recovered from old amps. I did simulations with encouraging results, did anyone carry out this project? Suggestions and advice are welcome. I attach original and modified scheme with simulations.Thanks

Hi all, I'm new to the forum so please forgive if I posted in the wrong place or something like that.

I want to build some tube amps (5F1, 5F6A, Steel String Singer & Two Rock JM Sig) and I want to make my own transformer/s. I think I understand the power transformers well enough, but I'm having trouble with understanding the output transformers. I had an idea to make one power transformer that has multiple taps so that I can use it for anything from a 5W 5F1 to a 100W SSS, where the transformer is rated for the VA of the highest one. I have done all of the calculations regarding core area & turns per volt.

But then, regarding output transformers though, I understand that the impedance ratio is the square of the turns ratio (which is equal to the voltage ratio) and that the transformer has no impedance of it's own but simply "reflects" the impedance of the speakers to the tubes and vice versa. I have the impedance ratio (eg. 2K1:4 ohm for the SSS running 4 6L6's) and therefore the turns ratio but does anyone know how to calculate the actual amount of turns? I read that having too few or too many can both cause problems. I'm also not completely sure if the amps are push-pull or single ended.

I will probably be able to find a donor core from an old transformer to use but it'll probably be bigger than necessary, that won't cause any problems will it? It would only cause problems if it was too small right?

Thanks all, and sorry for the LONG post 🙂

Hi All

This build is based on Mr.Speakers Open Alpha which a thread is available over hereon Head-Fi.

I had the open Alpha cups created and have it partially made. I used the modded cup design from HERE which removed the HI-Rose connectors use by Mr.Speakers aka Dan Clarke. This works for me as I have a few other T50 mods and headphones that use 3.5mm.

I had some issues with the paint and had to strip the cups back to bare plastic and probably going to go with a color shift galaxy blue.


Here is one I made for my friend. It is very simple as it just has some cotton from the Mayflower kit. I tested with with Cascadia hybrid pads and it was the most layered T50 I have ever heard.


I kind of like T50's


Open Alpha, T60 Argon, Vibrolab in Zebrawood and a MK Argon with MK3 baffles.

I will be adding screws to the bass ports on mine and will have more time to test materials. Once I get everything in place I'll tidy up this OP.

Best sounding DIY modules currently available.

Anyone able to give advice ?

Hello ppl, longtime lurker here. I'm not really active in online communities, although I find a lot of useful info on diyaudio forums and keep coming back a lot.
Now I've a question where I would like to find out 1 or more answers to and hopefully have the collective hive mind to inspire me with practical and high quality analog and maybe original solutions.
I'm looking for a way to switch audio line level signals, after a linkwitz riley filter bank, with the least switching noise. My first choice incorporated the 4066 with some logic to debounce and flipflop momentary switches. Then I was thinking of putting in some more logic to incorporate zerocrossing switching with lm339 comparator. Does this work? I made some kicad sketches but I havent breadboarded it yet.
Or maybe do you recommend switching (fast fading?) With jfets?
Looking forward to hear your thoughts.

// Jeroen, Sweden

I recently purchased a Toriody transformer. I ordered a bottom mount plate but have since decided to mount it upright. I could order one straight from the manufacture, but I doubt they're shipping. Does any one have one to sell or know a source to purchase. The transformer measures 6" in diameter.

FS: This is a matched quad of vintage 1950's blackplate Sylvania 6BQ5 tubes. These are Baldwin branded and were specially selected by Baldwin for low michrophonics as they were originally used in their bass amps for organs. These have been tested in my B&K707 mDynajet tester and all test strong and matched. I have also tested them in my Tubelab SPP and they sound great.

These are basically 7189 tubes and can stand much higher voltages than modern equivilents. Ask Tubelab, I believe he found in testing that these Sylvania Blackplates could withstand the most abuse.

$75 for the quad plus shipping. Paypal friends and family please.

Please feel free to ask any questions you might have. I have other tubes listed for sale in the swapmeet if there is something else you might be looking for. Combining shipping save cash!

Shipping is from Vienna, Austria.

Thanks! -Mac

Hello everyone,

I'm wondering how we ended up with the materials for diaphragms that we have today.

To paraphrase Wikipedia Loudspeaker - Wikipedia:

The diaphragm is usually manufactured with a cone- or dome-shaped profile. A variety of different materials may be used, but the most common are paper, plastic, and metal. The ideal material would 1) be rigid, to prevent uncontrolled cone motions; 2) have low mass, to minimize starting force requirements and energy storage issues; 3) be well damped, to reduce vibrations continuing after the signal has stopped with little or no audible ringing due to its resonance frequency as determined by its usage. In practice, all three of these criteria cannot be met simultaneously using existing materials; thus, driver design involves trade-offs. For example, paper is light and typically well damped, but is not stiff; metal may be stiff and light, but it usually has poor damping; plastic can be light, but typically, the stiffer it is made, the poorer the damping. As a result, many cones are made of some sort of composite material. For example, a cone might be made of cellulose paper, into which some carbon fiber, Kevlar, glass, hemp or bamboo fibers have been added; or it might use a honeycomb sandwich construction; or a coating might be applied to it so as to provide additional stiffening or damping.

Click to expand...

The plastic cones or domes I see today are mostly made out of Polypropylene (PP), sometimes mixed with 10 to 30% mineral powder such as talc. The latter increases stiffness by some degree, but it also increases weight.

AFAIK, the plastic used for cones before PP came to light was Polystyrene (PS, also called "Bextrene").

PS and PP, also PET ("Mylar") are not materials with any remarkable mechanical properties. They are very cheap and some of them are certified "food safe", so they are commonly found in food packaging. (My personal recommendation: Stay away from this sh*t were possible. But that is a entirely different topic.)

One does not have to go far as for aerospace materials. In fields such as consumer goods and automotive, a variety of plastics I've never seen used for speakers (PA, PC, PEEK...) and mixtures of different grades are used for injection molding, often reinforced with (short) glass or carbon fibers to improve them. No rocket science involved.

I see no such more advanced plastic stuff in the loudspeaker industry, and I wonder why.

For Sale, in the UK (England). 2 of Hammond 735A power transformers. Never been wired up, 1 has been bolted down to a base, and the other is totally unused.

They are 3000V centre tapped and 250ma (CCS rating). 375ma (ICAS)

735A - Hammond Mfg.



They were for an 833A project, but I'm going in a different direction now. They came from Philip Ramsey at Bluebell audio in Dundee, Scotland.

I'm looking for £300 each (£600 pair),prefer collection from North Essex / Suffolk borders but they could be sent by courier, ask for a quote if necessary.

Thanks for looking.

regards Philip.



I also have 2 of Chinese 833C (carbon anodes) with bases and top caps, and U19 rectifiers with bases and caps. All unused. Ask if interested.



*

A brief history of the project:
-------------------------------

This small project has the objective of to create a delay in time of
about 5-6 seconds in order to provide sufficient time to the main amplifier to
stabilize its internal voltages and currents, as it is fully DC coupled. One
of the conditions of the designing, was to take advantage and give use to some
particular compactron tubes I have at home, like 12AL11, 6G11, 6BF11, 6T10, et
al. All them have a sharp cutoff pentode unit designed originally for the FM
quadrature detection/demodulation, with all its electrodes tied to an independent
pin on the base, including the suppressor grid. Some of them has an internal
shield tied also to a pin or to the cathode (earthed in this project) Also,
the suppressor has been designed to take good control over the anode current.
The tetrode section is a beam unit, similar in some aspects to the 6AQ5 while
the pentode unit resembles the 6DT6 or 6AS6 in the 7 pin tubes.

The first knowledge about a circuit of this kind I got, was written on
a patent from Burrows (3). There, he said that a single pentode may be conected
in such a way that behaves like a Schmitt trigger (2), originally done with a
dual triode unit, their cathodes joined and grounded through a common resistor,
and with a resistive attenuator from one plate to the other grid to provide DC
coupling between the two stages. There is a free grid to trigger it in the mono
or bistable modes or provide a variable DC voltage level to make it oscillate
in the astable mode. This will result in a regenerative amplifier with two only
stable states, depending on the signal that is feed to the circuit, or, eventually,
make it to oscillate. In his idea, a resistive coupling between the screen and
the suppressor grids would give the desired (necessary) regenerative coupling,
as one may think a pentode as two triodes composed by the cathode, control grid
and screen grid and the other by the virtual cathode between the suppressor and
the screen, being the former the second (Triode) grid and the anode, the plate
of the second stage. He gives in his patent, voltages and resistances values
for his set up. I tried a similar connection to my pentodes, and made a large
reckoning to obtain the values of the passive components needed to put the
circuit to job. But it was a complete disaster. There wasn't any bistability,
nor hysteretic behavior. After it, I downloaded the 6AS6 datasheet and reverse
engineered the circuit, but it was very unclear (and still is) for me how the
author could do it to perform. My attention was focused in the unbypassed cathode
resistor he used, but no description of its functionality in the scheme. I tryed
the setup as a Phantastron, feeding the grid from the plate by means of a small
capacitor and it performed reasonably well, so I was near the objective but I
didn't get it.

So, I searched in the literature I have, and in the Seely's (4) book
it is said that such a connection was discovered by an Engineer called H. J.
Reich, who also is mentioned in the above mentioned patent, and made public in
an article written in the "Review of Scientific Instruments" near 1939, but it
is restricted only for paid subscriptors. So I made and extensive and deep
internet searching for more info, supposing that some kind of info must be
elsewhere, because those big genius usually published or teach their knowledge
around the society. I found that he also wrote an article in one of the "Electronics"
magazine issue of 1939 (5), I got the article in the americanradiohistory.com
website and printed it. I took a deep read of it, and effectively, Reich
explains that almost all pentodes have some serious irregularities in their
IG2 = f(EG2) characteristic, mainly at lower plate voltages. It is far from a
straight line, and has loops or buckles in them, and in certain way are like
to the negative resistance seen in the neon bulb I = f(V) characteristic. I knew
about this straneous behavior but I didn't link them to any kind of bistability
or the like. The secret was to find them in my pentodes. Then, I prepared a jig
in which the cathode of my tube was grounded directly, the control grid biased
thanks to contact potential using a single 10MΩ resistor to cathode, grounding
the suppressor too, the screen tied to +275V with a fixed 10KΩ resistor plus a
100KΩ pote wired as variable resistor (Rheostat). The beam unit was momentarily
unused. The plate was loaded with a 1W resistor to the same power source, and
hooked the oscilloscope probes, one to the plate and the other to the screen
electrode. Surprisingly, the irregularity was there: near 80V of screen bias,
and slowly rotating the pote, a steep of about 15V appeared and it refects in
the anode voltage too. I checked four 12AL11 tubes in the same way, with another
four different values of plate load resistor. I initially I mistrusted from the
pote, and double checked it for deeps in the carbon trace with and ohmmeter and
with a resistor from screen to ground (Simulating screen current), removing the
12AL11 from its socket and the jump didn't appear. So, effectively, it was the
pentode itself who was creating the jump in the screen grid voltage to current
function. So, Reich was true in his affirmation. The only that remain was to
couple the screen to the suppressor removing the earthing piece of wire, and
computing a pair of proper resistor values that, with the jump voltage differences,
can carry the suppressor below the plate current cutoff voltage, (near -4V as the
datasheet sais), or free the anode current, near 0V or perhaps +1V. This table
resumes what I found for the four 12AL11's, numbered arbitrarily:


I also tried 100KΩ anode resistor loads, but at such low values, the
jump vanishes too much to be of interest or importance. The next steep was to
check some of the other type of tubes I have at hand, like the 6BF11 and 6T10.
Rearranging the socket wiring, I found that the 6BF11 has a more pronounced
irregularity and was the final choice, and also performs good with 100KΩ anode
load. Fortunately the value of screen dropping resistor always were about 45
to 50Kê, for the entire tube lot, and a 47KΩ @ 2W was the final chose value.

Then I coupled the screen to the suppressor via a resistor divider and
a small mica cap (220pF) and to the negative rail of -275V. A 6AL5 diode was
used from suppressor to ground to prevent the last to make too positive. A
300Kê small preset was included to adjust the optimum value. I was very happy
when accidentally the solder iron cable (A Taiyo SL20/200 with an external
TRIAC dimmer) fall near the circuit (The white and blue cotton covered saw in
some of the pics) and the pentode acted a true Schmitt trigger (Some pics below).
The oscillograms show the anode, screen and supressor waveforms perfectly
triggered and syncronized to the spikes generated by the TRIAC dimmer, their
amplitudes and times, taking into account that the oscilloscope's test leads
has a X10 attenuation not shown in the read out, and that line frequency here
is 50Hz (20mS). The coupling of the cable to the circuit appeared to be more
sensible when placed near the screen-suppressor network that with coupling close
to the grid bias resistor. Burrows used both grid and suppressor coupling trough
germanium diodes.

Once that the Schmitt part of the circuit was OK, the next steep was
to couple a time constant to the input in a manner like a CD40106/74HC14,
hooking a 240VAC 9Kê relay to the beam power unit, a resistive divider from
the pentode anode to the beam unit's control grid. Initially I though to make
the pentode to act in a similar manner than in a device called by F. C. Williams
"Multiar" (1), which uses a transformer to take the regenerative fast action
and a diode as a switch. Also, a small 470pF mica capacitor was paralleled to
provide fast transient coupling from plate to grid. The way in which the time
constant and the values of the elements changed with time and circuit, but
finally I added another 6BV8 tube composed by two independent diodes and a
triode with æ ÷ 33. However, a small problem soonly arise. The triode was used
as cathode follower to provide high input resistance to the time constant network
and low impedance to the switching diode, and the diodes were wired to clamp
the grid voltage between -50V and ground provided by a resistive atenuattor from
the negative bus. But as the plate was to +275V then, the tube cuts off near
-20 or -25V and the full excursion of the voltage swing at the grid wasn't
transferred to the cathode. So, once again, I changed the socket and migrate to
another compactron tube 8B10. As I have only 6.3V heaters, I used it with 6.3V
temporarily with no apparent bright differences or notable performance loose.
(Furtherly I got some 6B10's and replaced them). This compactron has two triodes
like a 12AU7 and two common cathode diodes like a 6AL5 (or the 6CN7). Then, one
of this diodes was used to clamp the grid of the cathode follower to ground,
the other to the same task in the 6BF11's suppressor. The slicer (6) so formed,
has the objective of to take advantage only the fastest part of the exponential
voltage/time rundown waveform where it evolutes quickier and a more defined
behavior is expected, and limit the grid (And the cathode) swing to safe values
as the heater is grounded and the final voltage of the capacitor in absence of
the slicer is near the negative rail. Having freed a diode from the 6AL5, it
was become available to use it as the switch. The second triode was used as a
voltage source, hooking a neon lamp at its grid, biased with 220Kê to the +275V,
and feeding the anode of the follower with this lower voltage value. So, by
this moment the screen of the beam power section was biased with 100KΩ to the
positive rail and soonly I took the idea of to replace again the 6AL5 with the
6BV8 and make a rudimentary voltage regulator moving the neon to the negative
side, providing fixed clamping levels for the grid, and at the same time with
the triode inside the 6BV8 and one of the 6B10 as a "See Saw" voltage stabilizer,
and also feed from this to the screen of the beam power section. The voltage
at the screen is about 60V and suffices to cutout the anode current with -12V
accordingly to the tube data sheet, although the excursion available at the
control grid is of near 90V. But the 60V was OK for the current I need to drive
the relay coil. The current drawn by the relay coil is (275V - 60V) / 9KΩ, or
24mA, well under the tetrode capacities. Then I finally arrived to the actual
configuration (The bottomed voltage of the beam power unit is near 60V).

How it works:
-------------

R80 and C80 are the time constants, tied from negative bus and ground.
R89 limits the current drawn by the diode slicer and cuts out any voltage
above ground and below the neon's voltage, near 60-65VDC, negative to ground.
This network also is found in many CMOS devices to prevent ESD and dangerous
levels at their inputs. V82 is a follower, and the diode at its cathode is the
main switch. When powering it on, the pentode 6BF11 starts conducting full anode
current, with the suppressor biased at ground level imposed by the resistive
network and the diode. Anode voltage is low, and the coupling divider to beam
power section maintains the grid well below the cutoff bias, near -90V. As C80
is going more negative (The heaters are lighted some seconds ago), the cathode
follows the grid potential, and diode D2 is cut off because the volage at the
join of P80 and R83 is several volts negative. As well as the grid and cathode
of the follower are going sufficiently negative to direct bias D2, it starts
conducting and pushes the hysteretic network to a still more negative potential.
The Schmitt-like circuit quickly trips (Regeneratively) and the suppressor goes
very negative (-20V) cutting plate current and deriving the electron stream
flow to the screen only, as the control grid here only decides how many current
is able to flow but doesn't participate in the timing mechanism. The fact that
the pentode's cathode current never is fully cutout difficult that its cathode
develop a oxide layer being too many time without cathode current (6). When the
current of the plate of the pentode is cut out, the anode voltage rises violently
and carries the beam power unit in the grid base making it conductive and
energizing the relay at its anode. V81 and V82 (as already said) only make a
rudimentary voltage regulator for the follower and the beam section's screen.
P80 adjust the trip level and the initial conditions of the circuit. C86 makes
with the relay coil a loosy tuned circuit and self damps the spikes at the
moment of removing anode and relay current, and a diode isn't necessary here
because of the natural ruggedness of tubes.

You may feel to find similarities in the circuit with the hysteretic
scheme that can be build using a comparator or an opamp, feeding the non
inverting input with part of the output voltage swing, maintaining the other
input to a reference potential (In my case the control grid), and effectively
is seems similar. However, the mechanism is very different, as here there isn't
any amplification deliberately did to the signal, using a peculiarity of the
pentode's screen characteristic, and there is more similar with the negative
resistance of a neon pilot lamp, or in a SS device, nearest to a DIAC than a
trigger made with opamps.

All this stuff may be made simply wiring a capacitor in series to the
relay coil and feeding it with DC, when the cap charges sufficiently, the relay
liberates its armature and a change in circuit is done. But my idea was also
to explore old knowledge now almost useless.

R96 and the pushbutton discharges the cap during the experimental period,
simplifying the development of the circuit and being a test method.

I didn't check it for repeatability, linearity or constancy of the
timed circuit, as for the purpose for it was designed, those parameters are
unnecessary and irrelevant. Only a small delay of time to switch a relay was
needed.

The circuit uses only two compactron tubes, a noval unit and no SS at
all. Current drawn from the Ebb line is less than 30mA with the relay energized,
a couple of mA's in the negative rail, and a pair of amperes at 6.3VAC.

Conclusions:
------------

I built a timed relay device that prevents noises and the saturation of
an output autotransformer while the DC coupled audio amplifier is accomodating
its internal parameters, using multiple compactron tubes which was originally
made for FM demodulation and power amplifier in TV sets, and taking advantage
of the pentode's screen grid characteristic to make a hysteretic switching
circuit.

References:
-----------

1) US 2,540,093 from F. C. Williams (06/02/51);
2) Otto Schmitt, "A Termionic Trigger", Proceedings of the IRE;
3) US 2,872,572 from J. L. Burrows, (03/02/59);
4) Samuel Seely "Electron Tube Circuits", McGraw-Hill Book Company, 1950;
5) H. J. Reich, "Trigger Circuits", Electronics (08/39);
6) J. Millman and H. Taub "Pulse and Digital Circuits", McGraw-Hill Book
Company, 1956.