Mooer Little Monster AC - SE EL84 - Partial schematics

Hi,

Just got a Mooer Little monster AC, 5W single ended EL84 guitar amp.
I've tried to trace the PCB and draw the schematics, it's incomplete, maybe some mistakes here and there, but since I didn't found anything about it's better than nothing
I've made some measures on resistive load, it's almost linear from 200Hz to 8k, output transformer looks like a good one. Clipping begins at 7V - 8 ohms (3W)

EDIT : added inside pics. Yellow caps from the first pic added by myself for testing purpose. First mod done by increasing "mellow" cap to 47n for more bass. Amp provides almost a flat response starting from 150Hz instead of the default 300Hz.

Attachments

  • Mooer_LittleMonsterAC.jpg
    Mooer_LittleMonsterAC.jpg
    40 KB · Views: 78
  • IMG_20250322_104930.jpg
    IMG_20250322_104930.jpg
    353.9 KB · Views: 38
  • IMG_20250322_110425.jpg
    IMG_20250322_110425.jpg
    284.5 KB · Views: 32
  • IMG_20250322_110502.jpg
    IMG_20250322_110502.jpg
    272.1 KB · Views: 24
  • IMG_20250322_110548.jpg
    IMG_20250322_110548.jpg
    260 KB · Views: 38
  • Like
Reactions: djgibson51

DC-AC 4-8V AC power supply for heating audio tubes.

Hello,

My latest creation.

LVPS DC-AC Regulated Inverter EGS

This DC-AC regulated power supply board is used to power the audio tube heater with AC voltage. The voltage is adjustable from 4V to 8V via a potentiometer on the PCB.

The LVPS DC-AC Regulated Inverter EGS board is based on the use of the EGS002 module. It is a control board for pure sine wave inverters, primarily used in DC-AC power conversion systems.

It integrates the EG8010 chipset for generating pure sine waves with digital control and dead-time management. The module supports fixed frequencies of 50Hz and 60Hz, as well as adjustable frequencies ranging from 0 to 400Hz (need mods on the EGS002 module).

It offers protections against overvoltage, undervoltage, overcurrent, and overheating. The board is also equipped with an independent fuse to protect the EGS002 module (5V and 12V). To avoid damaging the tubes, there is a 3 second soft start mode by default.

The LVPS DC-AC Regulated Inverter EGS board operates up to about 4A with an input voltage of 15Vdc or 24Vdc. It can be configured with different inductors in the output filter depending on the types of audio tubes to be heated and the needed power.

The board has an output on 3 pins with an optional middle point (2 resistors to be soldered under the PCB). This avoids soldering the cathode resistors onto the tube socket.

Gerber and documentation files are available on my GitHub repository.

This power supply has not yet been tested in real time on a tube amp. This will be the next step as soon as I have completed another related project.

Enjoy !

Stef.

Attachments

  • DC-AC-Inverter-1.0.3.jpeg
    DC-AC-Inverter-1.0.3.jpeg
    862.3 KB · Views: 58
  • LVPS-DC-AC-Inverter-EGS-3D-NOEGS.jpg
    LVPS-DC-AC-Inverter-EGS-3D-NOEGS.jpg
    360.6 KB · Views: 41
  • LVPS-DC-AC-Inverter-EGS-FRONT.jpg
    LVPS-DC-AC-Inverter-EGS-FRONT.jpg
    413.3 KB · Views: 38
  • LVPS-DC-AC-Inverter-EGS-BACK.jpg
    LVPS-DC-AC-Inverter-EGS-BACK.jpg
    302 KB · Views: 38
  • 300B&scope.jpeg
    300B&scope.jpeg
    932.6 KB · Views: 51
  • LVPS-DC-AC-Inverter-EGS-diagram.jpg
    LVPS-DC-AC-Inverter-EGS-diagram.jpg
    541.2 KB · Views: 36

Musical Fidelity P170 Power Amp

Hi.

I have a bit of older MF gear, and I've been looking for some extra power amps to complete the HT setup.

I picked up a P170 the other day, it must be '80's gear, but I can't find a production date on it, so I'm guessing. Anyway, I'm going to replace all the Electrolytic caps in it, and then see how it compares to my other P150 and decide if it will be front or rear...

There is some evidence of previous repair inside, It looks like one of the caps has let go at some time in the past - there's a burn on the circuitboard near one of the transistors, and one of the 100uf/63v caps doesn't match it's other brothers - it's a 100v job, and a different brand - photo attached.

Apart from the cap replacement, is there anything else I should be looking at? The amp runs ok, and sounds good on my test setup of old bits 'n pieces.

Should I be looking to up the spec to 105 degree caps? They're 85 degree in there at the moment.

Any history on the P170? Unlike my other P150, this has a rear heatsink, and it weighs a lot!

Thanks,

Michael

Attachments

  • mf-p170_burn3.jpg
    mf-p170_burn3.jpg
    56.7 KB · Views: 3,873

Musical Fidelity A1 Re-build

Its been a while since I have contributed to this forum so thought it could be useful and informative for others to post details about my recent Musical Fidelity A1 re-build. The re-built amp sounds pretty amazing and well worth the many hours of effort slaving over a hot soldering iron and emptying my pockets to buy replacement components😀

To summarise the re-build:

1) Replaced direct wire 2-core mains lead with proper 3-core mains cable attached to an in-line 3-pin IEC plug, PE connected to chassis. The steel case on this amplifier is really hard to cut so this mod gives a mains IEC connection with very little effort.

2) Fitted a decent Ground Loop Breaker (Rod Elliot ESP) to replace the original 3R3/100nF parallel circuit attached to the chassis green 4mm banana post.

3) Upgraded the original 06A10 6A rectifier diodes with 10A10 10A and fitted 47nF 250V capacitor filter across the AC on the board. Possibly, a high current bridge rectifier may have been even better.

4) Re-capped the PSU capacitors, with as much as I could squeeze in to try compensate for the rather puny mains transformer. This involved using my Dremel to drill new capacitor mounting holes - 12000u + 2200u + 2200u followed by 12000u + 2200u, the original 0.47R 3W dropper resistor was replaced by 0.56R 5W which I had available. The 2200u were low ESR, 105C rated however the main Nichicon capacitors were standard 85C (I know, I know!).

5) Replaced the signal input 1u electrolytics (C6 & C7) with 1uF MKT and other electrolytics with same or better value/rating 105C replacements.

6) Removed the quad TL084 op-amps and re-fitted in DIL sockets, just for fun.

7) Isolated the existing pre-amp output using a 2-pin shunt jumper (open) and fitted direct input phono sockets to the power amp section, connected across resistor R3, 100K. I fitted a 2K/1K resistor divider onto the phono sockets to match the signal levels from my source. NOTE: This must be connected to a pre-amp or an audio source with volume control/adjustment.

8) Replaced the original MJ2955/2N3055 with MJ15023/MJ15024 transistors and fitted new silica washers with heat-sink paste instead of the dry thermal Silicone pads. Its important to do this because the pads may have deteriorated due to the repeated hot-cold thermal cycling and will have loosened off the mounting screws anyway. I fitted a heavy-gauge silicon wire collector connection to the board.

9) Replaced the board to output speaker connector cables with similar heavy gauge silicon wire.

10) Resprayed the top heatsink cover with 2 - 3 coats of matt black high temperature "stove" paint, good for up to 600 deg so should be ok with the A1 toaster.

Probably the most important improvement was to by-pass the existing pre-amp circuit, which is well-documented to be a poor design and bad implementation. Secondly, the seriously beefed-up PSU supply does wonders for the low end bass performance and general mid-range dynamics. Its warm, lush sound may not suit everyone's musical taste but its getting plenty of playing time with my music collection, its a definite keeper. I am a Musical Fidelity convert and have just bought an el cheapo B200 and am in the process of yet another re-build 🙂

Attachments

  • RIMG0779.jpg
    RIMG0779.jpg
    534.3 KB · Views: 933
  • RIMG0781.jpg
    RIMG0781.jpg
    440.8 KB · Views: 820
  • RIMG0784.jpg
    RIMG0784.jpg
    404.2 KB · Views: 768
  • RIMG0785.jpg
    RIMG0785.jpg
    411.7 KB · Views: 776
  • RIMG0787.jpg
    RIMG0787.jpg
    500.7 KB · Views: 882

For Sale 2SK170GR/2SJ74GR

Matched 2sk170gr/2sj74gr quad
20usd +shipping 15usd +PayPal fee or friends family option

3.0ma-5.8ma

2sk170gr matched quad
12usd + shipping 15usd +PayPal fee or friends family option

2.8ma-6.0ma

0.1ma matching
There are more to measure
Shipping with tracking number (Japan post)

Attachments

  • PXL_20241110_183642041.jpg
    PXL_20241110_183642041.jpg
    511.3 KB · Views: 118
  • PXL_20241110_183529529.jpg
    PXL_20241110_183529529.jpg
    366.2 KB · Views: 112
  • PXL_20241110_183458997.jpg
    PXL_20241110_183458997.jpg
    386.6 KB · Views: 109
  • Like
Reactions: Brijac

ask to Grok: Guitar Tube Amp

yesterday I asked Grok:

DIY 50W guitar tube amp?

A 50W tube amp is a solid step up—great for gigging or just cranking it at home! Building one is more involved than a little solid-state amp, but totally doable with some planning. I’ll outline a classic single-ended or push-pull design (push-pull is more common for 50W). Think along the lines of a Fender Bassman or Marshall Plexi vibe, but simplified for DIY.
Parts You’ll Need:
Tubes:
  • Power tubes: 2x EL34, 6L6, or KT66 (push-pull pair for ~50W). EL34s are punchy and Marshall-esque; 6L6s are cleaner and Fender-y.
  • Preamp tubes: 2-3x 12AX7 (ECC83) for gain stages and tone shaping.
  • Rectifier tube (optional): 5U4 or GZ34 for tube rectification, or use solid-state diodes for simplicity.
Core Components:
  • Output Transformer (OT): Critical for 50W. Match it to your tubes and speaker impedance (e.g., 3.5k-4kΩ primary for EL34s, 8Ω secondary). Hammond 1750 series or similar works.
  • Power Transformer (PT): Needs to supply ~300-400V at 150-200mA for the plates, plus 6.3V for heaters. Look for something rated around 200VA.
  • Chassis: Steel or aluminum box to mount everything. Punch holes for tubes and controls.
  • Speaker: 8Ω or 4Ω, 50W+ rating (12” Celestion or Jensen are popular).
  • Capacitors:
    • 22µF-47µF @ 450V electrolytics for power supply filtering.
    • 0.022µF-0.1µF @ 600V for coupling.
  • Resistors: Various (1W-2W rated) for bias, cathode, and grid circuits (e.g., 1kΩ, 470Ω, 100kΩ).
  • Potentiometers: 1MΩ for volume, 250kΩ for tone (bass, mid, treble).
  • Input jack: 1/4" mono.
  • Switches and fuses: Power switch, standby switch, 1-2A fuse.
Tools:
  • Soldering iron, multimeter, drill (for chassis), and wire (18-22 AWG).
Basic Steps:
  1. Design the Circuit:
    • For push-pull: Two power tubes (e.g., EL34s) driven by a phase inverter (one 12AX7 triode splits the signal). Two gain stages (another 12AX7) and a tone stack (Bassman-style is straightforward).
    • Check schematics online—Fender 5F6-A or Marshall 1959 circuits are good 50W starting points. Simplify as needed.
  2. Power Supply:
    • Wire the PT secondary to the rectifier (tube or diodes) to get ~350-400V DC after filtering with caps.
    • Add a choke (5-10H) or resistor (100Ω) between filter stages for cleaner B+ voltage.
    • Ground everything properly—star grounding avoids hum.
  3. Preamp Stage:
    • First 12AX7 stage: 1.5kΩ cathode resistor, 100kΩ plate resistor, 0.022µF coupling cap to the next stage.
    • Second stage: Similar, feeding into the tone stack (search “Fender tone stack calculator” for values).
  4. Phase Inverter:
    • Use a long-tailed pair (common in 50W amps). One 12AX7 triode, 10kΩ tail resistor, drives the power tubes.
  5. Power Stage:
    • Bias the EL34s (fixed bias with a negative voltage supply, ~ -35V, or cathode bias with a shared 250Ω resistor).
    • Connect grids to the phase inverter, plates to the OT primary, and OT secondary to the speaker.
  6. Assembly:
    • Mount tubes, transformers, and controls in the chassis.
    • Solder point-to-point or use turret board (safer and cleaner).
    • Double-check wiring—high voltage means mistakes can fry parts (or you).
  7. Test Safely:
    • Use a variac or light bulb limiter to power up slowly. Check voltages: ~350V on plates, ~250V on screens, heaters at 6.3V.
    • Plug in a guitar, start quiet, and listen for hum or distortion. Adjust bias if needed.
Notes:
  • Safety: You’re dealing with 400V+. Discharge caps before touching anything—use a resistor or screwdriver with a clip lead to ground.
  • Cost: Expect $200-$400 for parts, depending on quality (transformers are the big expense).
  • Sound: EL34s give a mid-heavy crunch; 6L6s are smoother. Tweak the tone stack for your style.

Side Exit Subwoofer Ports? Calculate End Correction Factor

I'm looking for another port configuration for my tall 4 way cabinets w/ integrated subwoofers. I'm interested in making 2 ports that run vertically in a tall speaker cabinet, and exit out the sides of the cabinet with a flare somewhat resembling a NACA duct.

For reference, the speaker cabinet is 16"x 16"x 52". The subwoofer cavity is an "L" shape, allowing for a long and straight port. If I can figure out how to design it!

I already did a prototype with dual vertical ports positioned on either side of the sub driver. They exited out a notch in the bottom back corner of the cabinet. It worked well enough. But there is some audible distortion/ noise from the enclosed port exit.

I would like to improve on the same concept, with the ports running vertically in the tall cabinet section, but then venting them out to either side. (I have also thought about a vertical slot port against the back panel, that would exit right above the rear facing woofer...)

I haven't actually made these designs yet because I'm unsure about how to find the effective length and the end correction factor.

Photo of the concept:

2025-03-14 16_20_35-Rhino 7 Commercial - [Top].jpg

I'm curious if anyone has tried something like this? I'm looking for a theory to calculate the "effective" length of such a port. Of course I will have to do some prototyping, but it would be good to start with an educated guess?

The drawing below is the speaker as designed with a 10" subwoofer driver. I built this one and used a curved port that fit neatly around the 10" driver. But it will be difficult to re-use the same curved port with the 12" subwoofer. Before I build molds to make a new curved port, I'd like to investigate the possibility of a simpler straight port...with a complicated port exit/ flare!

2025-03-14 16_27_37-2023-05-07 completed speaker cabinet design (10 MB) - Rhino 7 Commercial -...jpg

Advice welcomed about interpreting Frequency Response plots

I have taken several plots of my Purifi /Satori beryllium speakers, detailed in the thread 'Possible MTM Fun Project'.
I used an Omnicron USB mic and REW on a laptop. These have been useful in voicing the speakers, but I might like some advice as I'm not familiar with interpreting the plots or using REW in other ways.

First, here is a plot where it is now. Here are details of the crossover.
Bass: 2nd order 0.8mH, 50uF, very small notch at 3.5kHz; the 0.8 set by trial and error to give a reasonable bass level.
Treble: 3rd order 12uF, 0.28mH, 12uf; dropper 2 ohm/12ohm, with added 6ohm between crossover and speaker.
Note that the green line is the current state with 0.28mH; red is with 0.23mH.

FR .8 50 all rs tweeter 0.28mh.jpg

I am not sure how good this FR is. Divisions are 1dB so it is fairly flat, but I don't know if this is 'acceptable'.
So I compared with other speakers, below.
Monitor Audio (old), stand mount bass reflex. I don't know the model number.

FR monitor audio.jpg

My SEAS ER18/27TDFC MTM:
FR SEAS MTM normal 12 march.jpg

The Monitor Audio sounds reasonable, the SEAS MTM sounds very good even though its FR does not look so good.

To me, the new Purifi/Be MTM plot looks the best, as with 1dB per division the variation is not that much. Am I kidding myself????

There is an issue, which Mark (who made a good TM with these drivers) points out; there is a dip at 1800Hz (xover frequency) and a peak just below.
I don't think I hear a problem, but then I don't know what to listen for.
I improved the dip somewhat by increasing the tweeter inductor from .23 to.28mH, but it also raised the peak somewhat.
So I am wondering if I should continue to try to improve the dip, or whether such a dip is minor, of a value to be expected and nothing to worry about.
If it is a case of accepting the dip (the sound is good), I will revisit the 0.23mH, as the peak below 1800Hz looks to be less.

Any advice welcomed.

Attachments

  • FR .8 50 all Rs 12 .28 12.jpg
    FR .8 50 all Rs 12 .28 12.jpg
    71.4 KB · Views: 53

MC1454 vs CD4541 timer IC

When repairing som PCB's at work I came across the MC14541B timer IC for the first time. It's fitted on a board from early 2000's, I guess.

From what I can see from the data sheet it's pretty much identical to the well know CD4145 IC. Pin campatible, both CMOS and about the same supply. I can't see the difference. Ironically, 4541 backwards is 1454.

Does anyone know something about these? Think I'll use CD4541 in repairs, since they still available, though they are listed as obsolete, at least the DIP version.

what do you think of this hypothetical mid-high frequency compliment of technologies ?

low frequencies = cones. not too controversial. yes there are belt driven subwoofers, modulated fan subwoofers, there is Powersoft M-Morce moving magnet subwoofer etc. but for our purposes here we're just going to cover everything below 1 khz using good old dynamic paper cone drivers and we won't discuss them because they're boring.

but mid to high frequencies there are options, for example:

cones
domes
compression drivers
planar
AMT
true ribbon
electrostatic

probably forgetting some like Plasma tweeters ...

the question is what would be the perfect compliment of technologies to cover the range above 1 khz ?

my thinking on this question has evolved over time but as of two days ago it is as follows:

1 - 3 khz = BMS 4596ND ( annular diaphragm compression driver )

https://www.bmsspeakers.com/index.php-71.html?id=4596nd

3 - 9 khz = Beyma TPL200 ( pleated diaphragm air motion transformer )

https://audioxpress.com/article/tes...a-tpl200-h-pro-sound-air-velocity-transformer

9+ khz = Aurum Cantus G3 ( pure aluminum true ribbon )

https://www.parts-express.com/Aurum...6kDwku8s0r1xlyZU5_2n0FYdFRLdz9xUaAmGJEALw_wcB

those technologies are the most efficient, flattest measuring, highest output and lowest distortion in their respective frequency ranges.

for example the BMS midrange begins to fall apart above 3 khz due to "mass break point" ( apparently function of horn geometry ) and flex of the relatively soft polyester diaphragm ...

the TPL begins to fall apart above 9 khz due to the depth of the pleat putting an upper limit on HF response much like in a compression dome the spacing between phase plug slots puts a limit on HF response

while the True Ribbon can go well past 20 khz but is limited on the low end by the fact that it doesn't seal the air chamber ...

as well the TPL can't go very low either because the forces are side to side - no force front to back which means air pressure at low frequencies can push the entire diaphragm in or out which means that the air chamber is only FLEXIBLY sealed and it can only go marginally lower than true ribbon - nowhere near as low as compression driver

a planar can go as low as a compression driver because motor force is front to back and the chamber is sealed. the problem is that at frequencies below those which AMT can comfortably handle the planar is soundly beaten by the compression driver, by about 10 decibels of efficiency, while also having the benefit of user replaceable diaphragms. i think one reason you don't see many planars in prosound is you can't replace a blown diaphragm in a planar like you can in a compression driver.

you CAN replace a diaphragm in the Beyma TPL as well you can replace a ribbon in most good True Ribbon tweeters like Aurum Cantus, which offers both replacement ribbons and replacement instructions. RAAL also sells replacement ribbons for most of their drivers. With some ribbon tweeters you can roll your own ribbons as well.

and at frequencies above mass break point of compression driver the planars exhibit rather erratic behavior that isn't close to flat. in fact the planars aren't flat anywhere in the frequency range. combine that with planars having the lowest efficiency ( between CDs, AMTs and True Ribbons ) and being the only one of these four technologies with non-replaceable diaphragms and you can see why i have eliminated them from the "optimum" setup.

and thus the winning trinity ( as mentioned above ) is Compression, AMT and True Ribbon with 1 khz, 3 khz and 9 khz crossover points where the compression is a dedicated midrange from BMS ( or if you prefer B&C, but i think BMS is better ) and the AMT is Beyma TPL200 ( which i think makes a better dedicated upper midrange, while the TPL150 tries and fails to cover a wider frequency range ).

sadly there is no True Ribbon on same level of engineering as BMS and Beyma because major prosound companies like that do not make True Ribbons and instead for TR you have to rely on less sophisticated operations, most of them from Asia ( except RAAL ) and there doesn't seem to be one real winner between the Ribbons.

RAAL seems to be the highest regarded and does seem to have the best ribbon element but i don't like how they put the transformer inside the acoustical chamber. it strikes me as a cost cutting measure so they don't have to build a housing for the transformer like every other ribbon maker does and the downside is the chamber now can't be fully acoustically optimized. it is now both too large ( for a supertweeter ) and asymmetrical.
  • Thank You
Reactions: GM

Did I murder my Philips CD303?

I got a Philips cd 303 with noisy outputs, so I replaced all the caps on the power supply and laser board, repaired broken soldier joints... And it worked fine.
Then I decided to change dac power supply caps, but when I was connecting dac board I misplaced two 3 cable connectors, the data and power supply I think.
Now only drawer works and mab8410p chip is getting really hot. I tried putting a chip from a cd 304, which is a little different, and with it it spins the disc but won't read, display doesn't work... but the chip is staying cool.
So, did I fry the mab8410p? Are there any replacements avalible?
Any help would be greatly appeciated.

Hiraga Super 30 Grounding

I’ve been struggling with this for a while.

Had to replace a couple of resistors on the boards, so figured since I was going to tear the amp down, I should try another grounding protocol to see if I could solve what’s been an ongoing issue.

As the amp sits on its own, with no source, it is dead quiet. After readjusting the voltages on the boards, I am getting 1mV of AC on the output. Very happy about that 🙂

However, as soon as I hook up a preamp, I get a buzzing sound.

The photo attached showed how it currently is wired up.

Made use of a star ground. The secondary centres are connected to the star. The star is connected to one end of each of the power supply boards. The ground terminals on the amp boards are also connected to the star. The earth ground is connected to the star, and the chassis. I just tried connecting the RCA ground to the star also. Both the grounds of the RCA terminals are connected together and a single wire connects them to the star

What on earth (pun intended) am I not getting here?

Thank you for your thoughts.

P

Attachments

  • IMG_8428.jpeg
    IMG_8428.jpeg
    1.3 MB · Views: 86

For Sale GlassWare AD PS-15

Asembled and ready to use with all original parts that come with glassware kit.
Good up to 400V of DC out voltage.The heater are now for 6,3V output voltage but you can change that to what voltage you want.The both heater regs can be made also als symetrical supply if you need this.The board comes with original manual.Price 100 euro plus shipping.This board works perfectly!!

Attachments

  • 20240521_204429.jpg
    20240521_204429.jpg
    632.1 KB · Views: 365
  • 20240521_204434.jpg
    20240521_204434.jpg
    620.8 KB · Views: 363
  • 20240521_204454.jpg
    20240521_204454.jpg
    377.7 KB · Views: 371

Building the Zenith a home

Thanks much to those of you who have answered some very basic questions I have so far. I have settled on a design for my particular driver and desired usage, and would like some feeback.

The principle driver in question is a Zenith 49CZ817.

I went down two separate trains of thought here, and ended up not going open baffle...mainly because I didn't want to have the speaker that far from the wall. And, I liked the thought of not needing a dedicated amplifier/dsp system.

I spent some time trying to copy and tweak the decca corner horn to match the size and t/s parameters of my driver, and it didn't really seem to work well. Backloaded horns tended to get too large (and definitely outside my skill set), so I compared MLTLs to the simple bass reflex. Back and forth, may times. And somehow I ended up staring lovingly at the onken style.

Well... that is not what I ended up with, since I really liked the concept of backward reflecting, along with a nice plant stand. But I definitely stole something from the onken and something from the decca...and probably a lot of other places too. I still have a few questions, before I fine tune and get all the numbers matching.

Attached is the design of the speaker, along with the T/S of the driver and a hornresp file which is close (not accurate to model), but shows the modeling strategy. A big box, with a big downfiring port...Not quite Onken big, based off what I read.

1). The ports fire down, and I can adjust height from the floor...but am unclear as to how to model this in hornresp. Is there a rule of thumb minimum here?

2). The angle of the speaker is 60 from horinzontal. It will be firing upward and backward into a corner that is 8.5' in height, and slopes up to about 17'. 60 degrees was pulled for aesthetic and math purposes. Is there a better angle to consider?

3). bracing. I have not yet added the bracing to the model, and know there will very likely need to be some. I have seen a lot of braces with the circular cutout approach. Is there some minimum hole size relative to being able to ignore, or else how do I model it. Or, do I do minimal bracing and accept that the Ql will be quite high? Looking at the rough modeling, this seems to be pretty insensitive to Ql. Though i have no idea what a Ql of 50 looks like.

4). Design - Am I missing something critical in a "never do that" sense?

5). Tweeter. As of now, I know I need something to get above 6-8K, and I have a kilpsch K-89-KV, which was ripped from a center speaker. I was going to throw that in the same baffle, and figure out the crossover once in. I know nothing about this, other than I have one. Any thought on how to present tis tweeter in its best light?

The back and forth between sketchup and hornresp is fairly intensive for me, so before going further down this hole....

Thanks for thoughts and opinions.

-jeff

Attachments

  • Screenshot 2025-03-08 at 9.09.52 AM.png
    Screenshot 2025-03-08 at 9.09.52 AM.png
    557.3 KB · Views: 51
  • Screenshot 2025-03-15 at 1.45.49 PM.png
    Screenshot 2025-03-15 at 1.45.49 PM.png
    141.1 KB · Views: 50
  • Screenshot 2025-03-15 at 2.52.47 PM.png
    Screenshot 2025-03-15 at 2.52.47 PM.png
    96 KB · Views: 52
  • corner_onk.txt
    corner_onk.txt
    2.7 KB · Views: 24

Bruel & Kjaer Sine Generator Type 1051 - need help on output stage

I have this equipment in need of repair. The frequency output is working fine except for its voltage level. For example when I set a voltage AC level of 1V, I only got 0.981 Vac, no matter which potentiometers I suspect is relevant.
Of course its difficult to find schematics and manuals for this OEM even from the WEB.
Any advise from members out there will be much appreciated. Thanks much.

Marantz 2230 Save with Wondom Class D Amplifier Implant

The Marantz 2230 became a victim of my careless shorting of the output resistor to ground while checking BIAS. Several attempts were made to repair this channel, but they were in vain. The only way to recover use of this 2230 was with a Wondom amplifier implant served by the Marantz dedicated single rail power of 66 VDC (You might have seem my other thread rescuing a Mitsubishi DA-R35 with a similar Wondom PCB amp, actually have done two of those). This unit has three secondaries, one to run dial lights, one for pre and line stages and the radio, and a dedicated high amperage for the driver/power amp individual channel units.

I was able to use this single 66 VDC with the Wondom AA-AB32516 amplifier module that could receive up to 72 volts and output 500 watts at 4 ohms--even at lower 60 VDC voltage. I removed both original amp assemblies and their heat sinks, along with the main power cap and two channel output caps. I installed a small black walnut board to hold the Wondom as you see and removed all excess wiring. I was able to use only existing Marantz wiring to power the Wondom and the only wires added, unrelated, were to provide a dimming switch for the dial lighting. Hopefully, someone will be able to make sense of this and the photos below in case another 2230 finds itself in the same predicament. May I add that the sound is fantastic and modern. Some may find the cooling fan noise a bit annoying since it cycles on/off, but eventually you forget about it.

A bit of the fat magic of the Marantz sound is gone, but not all. Treble is way improved and so is focus of sound, probably due to improved speed of class D. I will eventually add a wiring diagram. Some pictures for now.

BTW, after several days, no pops or smoke, and another 2230 saved from a dusty shelf.

Attachments

ideas for a modular preamp

Hi Folks,

here is the start of my idea for a modular preamp.
The pic shows the control (without audio) schematic for a modular preamp with 10 slots for different in/out modules.

happy to answer your questions to this and open for additional ideas.

regards

HP

Attachments

Bias current for a Parasound HCA-1200

I'm doing some work on a 30+ year old Parasound HCA-1200 amplifier. I bought this in the USA, but now live in Germany, and decided to get rid of the bulky/ugly external step-down transformer and convert the amplifier for a 230VAC mains. Fortunately this is easy since the power transformer has two parallel primary windings, which can be put into series for a 230V input. See annotated schematic below (bottom left corner is the relevant bit).

Anyway, I have a question regarding the correct bias current for the output transistors. I found a table online listing the correct bias current settings for all Parasound amps apart from the HCA-1200. Does anyone know the correct value? In other words, what's the correct DC voltage across the 0.33 ohm emitter resistors on the output stage?

I wrote to Parasound to ask this, and got a quick reply saying "we don't know". I then asked if John Curl could be reached to find out, and they replied that he had lost the information in a fire. Perhaps someone here knows the answer...

Cheers,
Bruce

HCA1200VoltageConvert.png

3 way cross over ideas

Hi, i just rebuilt a pair of JBL j900mv tower speakers on a budget and i want to set up a better crossover design than they came with.
The towers are built as follows:
8” chinese silver flute woofer
6.5” dayton audio fiberglass cone midrange speaker
~1.5” titanium tweeter that came with it
I was thinking of using the Dayton Audio XO3W-375/3K 3-Way Speaker Crossover 375/3,000 Hz as a prebuilt board which is way better than the 3.5khz crossover already installed from the factory. The speakers are also being paired with a polk PSW10 so the deep bass is already taken care of.
As a separate thought, i was thinking of using an inline high pass such as the Parts Express 100 Hz High Pass 8 Ohm Crossover
• this whole setup means the sub would play from 0-100hz, the 8” 100-375, the 6.5” 375-3,000, and the tweeter 3,000-20,000

How to: 2 Tone arms into one Amp

I have a Denon PMA-800. (Which has an analogue amplification path through the amp.)

I am in the process of reviving my old turntable setup, after a 30 year plus storage in the attic.

This comprises a Garrard 301 and an SME 3012 S2 with Shure V15 III.

I have got an SME 3009 which I want to add to the turntable. So that I'll have the option of using either tone arm to play my records.

The question I have is how do I connect both arms to the amp?

The amp has connect options to Phone, CD, Aux, tape/cassette recorder. These will have SME 3012 s2, Denon CD player, Web/NAS audio sources via Heos Link and Teac twin cassette deck.

I want to have the SME 3009 available for use without having to change the connections on the back of the amp, or changing the phono lead between the tone arms.

Thanks in advance for any advice.

"best" true ribbon supertweeter for line array

OK i know Beyma TPL is not supposed to need a supertweeter ( although Joseph Crowe measured 15 decibel drop by 20 khz on TPL 150 )

and in fact some here believe Radian LM8K doesn't need a supertweeter even though on Radian's own measurements it drops like a cliff above about 15 khz

and i agree for prosound use they can just be used full range and i probably wouldn't even hear the difference but it would not be a true audiophile system IMO

reality is that some young women can hear past 20 khz ... in fact i used to know one who would put her ear against the magnesium alloy tweeter on my studio monitors and just listen to the tweeter because she was fascinated by the sound of it ... as much as i tried to stop her ( explaining it was very bad for her hearing ) she refused to stop ...

back then i myself could hear to almost 20 khz and i'm sure it's a lot less now ( haven't tested it in recent years ) but even if i personally wouldn't hear the roll off in the Beyma TPL i would like to have a speaker where NOBODY can hear the roll off.

most true ribbons will easily hit 20 khz and go well past it ... but would they be able to keep up with an array of Beyma TPLs ?

i know what you're thinking - isophasic you can't afford an array of Beyma TPLs ! and that is true - i can't ! not a floor to ceiling array anyway. but i was thinking what if i do like a partial array that is only about 2 feet long ... it would still be hideously expensive but it might actually be worth it ( unlike a 8 foot array )

you know yesterday i compared the pictures of Beyma TPL to the non-prosound pleated tweeter from Aurum Cantus and there is no comparison. the Beyma is on a completely another level. check these pictures:

Aurum Cantus:

1742331890556.png


notice the uneven pleating. notice the thin metal bars ( the magnets are proportionally small to bar thin-ness )

1742331956772.png


notice the perfectly even pleating and the thick metal bars ( magnets are proportionally large to bar thickness )

you can clearly see the difference in price even without listening.

which made me wonder - if consumer AMT is so inferior to prosound AMT - how good can a consumer true ribbon really be ? Aurum Cantus is already on the high-end side of things when it comes to ribbons ( obviously not when compared to RAAL )

the problem with RAAL is even though it has very wide and flat frequency response ( to like 100 khz ) it is achieved by using thin flat foil and large volume rear chamber which ultimately exposes the ribbon to being SHREDDED by any kind of loud bass from the woofers hitting it. i mean there has to be a reason why you don't see any true ribbons in prosound.

so i am not sure that ultra-thin flat ribbon is really the answer. maybe a bit thicker and corrugated is better. also i think the back chamber should probably be stuffed with some styrofoam to reduce the volume of the cavity by like 90% so that there is less air movement at low frequencies and the ribbon isn't ripped. after all i don't need the ribbon going down to 2 khz, only to about 8 khz.

i actually think RAAL placing transformer inside the magnetic structure is a bad idea because although it makes for a more compact driver and larger volume chamber it is now more difficult to modify the acoustical properties of the chamber when the transformer is sitting in there. i prefer outside transformer like in most other designs.

but outside of RAAL the Aurum Cantus was already the best built true ribbon that i know yet the build quality of their AMT is a joke compared to the build quality of Beyma TPL ...

so what is a bro supposed to use as a true ribbon supertweeter to mate to Beyma TPL200 at around 8 khz ?

would be nice if Beyma made a true ribbon but we know it will never happen. and RAAL replacement ribbons are too expensive. Fountek sandwich ribbon not loud enough to keep up with TPL and Aurum Cantus loud enough but looks like it was made in some basement.

should i just accept the sketchy build quality of Aurum Cantus and simply look at it as charming rather than unprofessional ? i mean considering it would only cover the top octave a compromise there is definitely on the table.

the Beyma TPL by comparison would cover a far more critical frequency range - one in which i am not ready to compromise.

when comparing Beyma TPL to Radian LM8K there is no comparison. the Beyma is much flatter response, much more efficient and with higher power handling to boot. and this is true even when only using it to 8 khz. in fact i don't see any other driver coming near Beyma TPL between 3 khz and 8 khz. not even the Eighteen Sound AMT, which costs the same as the Beyma.

but the Beyma doesn't reach 20 khz ... i just need to find the right True Ribbon supertweeter to use above 8 khz which is about where Beyma TPL200 begins to falter based on tests by Vance Dickason:

https://audioxpress.com/article/tes...a-tpl200-h-pro-sound-air-velocity-transformer

1742333327481.png


3 khz to 8 khz this TPL200 is just PERFECTION

Can someone explain how this Linkwitz Transform works?

Hello, not sure if this is in the correct section, but will be used on a full range driver, so here goes.

Easy question, can i use normal audio capacitors and resistors? ie 250v or 10w.

Maybe the hard one, what the hell is happening at the triangle symbol where a positive line becomes negative at point 2, then looks to become positive going to ground? Not too sure what the triangle represents, i think i understand the rest of it with a few more grounds than i'm use to.

Attachments

  • circuit.gif
    circuit.gif
    8.9 KB · Views: 664

Need help with a low pass active filter

So as the title says i need a little bit of help to create this filter, i was able to calculate everything for my desired cutoff (80hz) and was able to build everything it works but there is a+
strange behaviour:
Screenshot (109).png

this is the circuit i followed (did twice having 2 op-amps in a single one) i am using a KIA 4558P but following the circuit for the gain feedback circuit (r1-r2) it doesnt filter anymore it doesnt even work if i dont want to use the gain and just wire directly the output to the inverting input it still doesnt work, but the moment i connect the output to the non-inverting input (+) it does work but now i dont have gain and loose usable power of the sub.


sorry for bad english not my main language, sorry for wrong words/terms eletrical engeenering is not my main subject is just a hobby. Thanks in advance

Hello my name is Brian Hernandez I live in Brea Orange County California need help and I will pay it forward I’m a certified hvac Tech but work on all

I have a T1500-1bdcp I got from my cousin for free I contacted it and it went to protected mode opened it and picture and video will show what I found can anyone help me with exact parts to purchase and where to purchase them please

Attachments

  • IMG_2872.jpeg
    IMG_2872.jpeg
    795.2 KB · Views: 67
  • IMG_2871.jpeg
    IMG_2871.jpeg
    740.9 KB · Views: 68

rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool

as rePhase has evolved a lot since day one of this long thread, it is certainly not a good starting point to understand what rePhase does and what its current state of development is. It is nonetheless a good place to ask questions, report bugs, or request features.

This first post is kept up-to-date with links and a change log for the current version of rePhase.

Here is the rePhase project page from where the last version can be downloaded.
It also list a few interesting tutorials.

enjoy 🙂

original post:

Here is a little piece of software I have been working on for some months now.
It is called rePhase, and is a tool for loudspeaker phase linearization, EQ and FIR filtering.
It does so by producing correction impulse responses that you can then use in your favorite convolution engine.

It is a free software, without restriction (but without warranty either).

Download: rePhase project page

At start the first goal of rePhase was to let you generate impulses specifically tailored to reverse the phase shifts introduced by the crossovers of your loudspeakers (passive or active) and boxes, resulting in a linear-phase system.
It is similar to phase arbitrator in this regard (but it is not a processing plugin, just an impulse generator).

rePhase has evolved and now includes other features: it can generate linear-phase EQ and crossover filters of arbitrary slopes, including Linkwitz-Riley (albeit linear-phase) and Horbach-Keele shapes.

With the paragraphic EQ sections you can alter phase and amplitude independently.

You will need a convolution engine to run the generated impulse, such as the minidsp openDRC, foobar2000 convolver or VST convolver, among many other software and hardware tools.

You can use rePhase to linearize passive loudspeakers, active ones, and turn IIR active crossovers into linear-phase crossovers (turn a DCX2496 into a DEQX or a Dolby Lake 😉 ).
You can also design the whole filter with rePhase.
In any case you will need one convolution per way (Jriver can be used to run the convolutions for example)

Features include:
  • generate impulse responses (FIR) for convolution engines
  • measurement import and real-time correction
  • loudspeakers phase linearization (passive or active crossovers)
  • linear-phase and minimum-phase gain paragraphic EQ and shelving
  • constant-gain phase praragraphic EQ
  • multiple gain EQ algorithms (constant Q, proportional Q, constant shape, raised cosine)
  • arbitrary slopes linear-phase filters (Linkwitz-Riley, Brickwall, Horbach-Keele, etc.)
  • arbitrary slopes minimum-phase filters (Linkwitz-Riley, Butterworth, etc.)
  • real-time graphical monitoring of target and results curves
  • automatic optimization for the best result with a given number of taps
  • multiple windowing choices

Sorry for the lack of documentation.
Questions, remarks and suggestions are welcome!

Changelog:

Code:
1.4.3 2019-01-16
  Bug corrections:
    - after loading a preset, correctly show "rotate" option when set in
      Filters Linearization tab
    - resolved the zoom out (right click) bug where too much zoom states
      were added
  Adjustments:
    - measurement interpolation is now logarithmic in both magnitude and
      frequency axes, so that an interpolation between two points will
      always show as a straight line

1.4.2 2018-12-27
  Bug corrections:
    - corrected graphical EQ manipulation behavior when gain offset is used
      on a measurement
    - corrected active EQ focus bug after loading a measurement

1.4.1 2018-12-23
  Adjustments:
    - improved compatibility of frd format:
        * enforce decimal form instead of scientific notation
        * use semicolon instead of tabulation as column separator
        * add commented info (software, url, columns description)
    - improved information in EQ bank drop-down menu, including EQ type and
      number of bypassed EQs
    - dynamically adjusted choices in FFT drop-down menu, removing unusable
      options
    - removed unused EQ type drop-down menus in paragraphic phase EQ tab
    - reworked links and contact info

1.4.0 2018-12-19
  New features:
    - graphical zoom functionality, similar to HOLMImpulse:
        * clicking and dragging the mouse over the response graph draws a
          zoom box that defines the new graph area when button is released
        * right clicking cancels last zoom operation
        * zoom only affects frequency and magnitude scales, not phase one
    - real-time graphical edition of gain EQ points:
      clicking an EQ fader or entries will turn them yellow to reflect the
      fact that this particular EQ point now has a special focus:
        * mouse wheel changes its Q value while on the response graph
        * middle click or ctrl-click on the response graph updates its
          freq and dB values in real-time while the button is held pressed
        * dB position is relative to the existing target magnitude curve,
          including optional measurement, and is limited to ±96dB per EQ
          (note: it is recommanded to use "constant shape" EQ type for
          high amplitude corrections)
        * modifications can be cancelled as long as focus is not lost
    - "frequency response (.frd)" format to export the generated correction
      (ie the red curves) as a three columns frequency/magnitude/phase file
    - Added "linearize"/"rotate" option in Filters Linearization tab.
      "linearize" is the default and compensates for phase rotation of a
      given filter (inverse all-pass), whereas "rotate" emulates the phase
      rotation of the chosen filter without affecting magnitude (all-pass).
    - New "throughout banks" EQ tools to bypass or order EQ points
      throughout all banks at once. Confirmation is requested as this can
      be an irreversible operation. Ordering EQ points between different
      banks requires EQ types to be identical in all banks.
    - try to let the user save current settings before exiting in case of a
      crash
  Adjustments:
    - frequency marker on the magnitude target curve is replaced with a
      vertical yellow line that reflects both magnitude and phase
      corrections
    - up/down key binding on drop-down menus to iterate values (same idea
      as existing incrementation/decrementation of entries with numerical
      values)
    - link to rephase.org in Help menu
    - stay in same tab after settings load/reset
    - more compact setting file
    - view preferences are saved on the fly instead of when quitting
    - force entry focus loss when switching tab
    - added Nyquist frequencies of a few common sampling rates in frequency
      upper limit choices in the Range tab
    - removed bypassed EQs from EQ points count in "Bank" drop-down menu
    - avoid saving measurement summary in the setting file, and generate it
      on the fly

1.3.0 2018-02-15
  New features:
    - "Load Recent Settings" entry in File menu, keeping track of the last
      15 opened or saved setting files
    - ctrl-click on a fader will reset its value to zero, similar to what a
      middle click does (useful for persons using a mouse pad)
    - improved measurement parsing heuristics to handle more formats;
      recommended measurement format is still the one described in the info
      box when failing to load a measurement
    - added CSV output format: 64 bit floating-point values in text format,
      separated with commas
    - allowed measurement importing from clipboard, both from File menu
      and Measurement tab
    - added "hide magnitude" button in Measurement tab
  Adjustments:
    - added a "Donate" entry in Help menu linking to a paypal donation page
    - made "Load Settings From Clipboard" functionality tolerant to leading
      and trailing spaces and newlines in clipboard
    - reworked output format names
    - renamed "invert" button to "invert response" for clarity in
      Measurement tab
    - rename "impulse offset" to "impulse delay" for clarity in impulse
      status report
    - add max impulse level in impulse status report, complementing max
      response level
    - reworked impulse and measurement status report areas to make them
      more visible
    - enabled DPI adaptation if forced to by the operating system (not
      recommended, looks nice but crashes might occur), and added scale
      ratio and DPI indication in View menu
    - added a no warranty disclaimer in "About" info box

1.2.0 2016-12-08
  New features:
    - REW automated EQ settings generated using the 'rePhase' equaliser
      type (as implemented in REW V5.17 beta 14 and up) can now be imported
      directly into a paragraphic EQ bank
    - EQ points in paragraphic EQ tabs can now be individually bypassed
    - added a "tools" menu in paragraphic EQ tabs, effective on current
      bank:
      * load/save current bank into a '.eq' file as a JSON object
        (gain paragraphic EQ tab only)
      * load/save current bank into the clipboard as a JSON object to
        easily copy it to other banks or rePhase instances, or share it
        through forum posts
        (gain paragraphic EQ tab only)
      * import REW EQ settings generated with 'rePhase' equaliser type
        (gain paragraphic EQ tab only)
      * convert back and forth between constant and proportional Q types
        (gain paragraphic EQ tab only)
      * invert corrections
      * bypass or activate all EQ points
      * order by frequency, active or reversed order
    - "Help" menu entry (albeit probably not very helpful :( )
  Bug corrections:
    - The long lasting encoding issues with paths when loading, saving, and
      generating files should now at last be solved. It was already
      supposed to be the case in version 0.9.7, then 1.1.0, and should now
      *at last* be effective. Please report any problem with files or paths
      containing special characters (accents, etc.).
    - Corrected a bug introduced in version 1.1.1: fader position could
      sometimes change based on the position of the mouse cursor after
      loading or saving a file
    - Stop confining mouse cursor within faders, as it could stay stuck
      under some rare circumstances
  Adjustments:
    - set default optimization setting to none, as optimization process can
      increase preringing and should only really be used when the number of
      available taps is too limited to obtain the desired magnitude curve
    - increased default number of taps to 16384 to reflect an increase in
      CPU and DSP power in the last few years (wishful thinking? :) )
    - boost FFT length calculation ratio to improve precision
    - changed default windowing algorithm from rectangular to hann for a
      more generic default behavior
    - EPS vector files screenshots including result curves are now
      significantly lighter and result in smoother curves compared to
      versions 1.1.0 and 1.1.1
    - suppressed flickering when switching between Views buttons
    - default to "Large" view mode

1.1.1 2016-10-29
  New features:
    - added a 64 bit IEEE-754 output format to accommodate BruteFIR:
      [url=http://tinyurl.com/htqvln8]rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 124 - diyAudio[/url]
  Bug corrections:
    - fader focus bug solved: [url=http://tinyurl.com/ztbp7cl]rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 123 - diyAudio[/url]
    - beefed-up clipboard handling to avoid bugs when loading measurements
      by dragging them over the interface: [url=http://tinyurl.com/jygt7jx]rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 125 - diyAudio[/url]
    - removed the final optimization step that was recently added in
      version 1.1.0 as it had some ill side effects in specific scenarios
  Adjustments:
    - renamed measurement "compensate" function to "invert" to avoid
      confusions: [url]http://tinyurl.com/j76wm5w[/url]
    - reworked focus and entries editing in paragraphic EQ tabs:
        * using the tab key now goes from one entry to the other as
          expected, making it faster to edit multiple EQ points in a row
        * clicking on an entry does select the entire content for ease of
          editing
        * middle click or double left click on the dB/° entry used to reset
          the value to 0. This is now gone, but the same result can still
          be achieved with a middle click on the fader
    - internal DPI scaling adjustments

1.1.0 2016-10-27
  New features:
    - screenshot functionalities added to the file menu:
        * "Save Graph Screenshot As..." saves current graph view with a
          logo added to the bottom right corner
        * "Save Window Screenshot As..." saves current window view with the
          directory entry hidden for anonymity concerns
        * supported bitmap file formats are PNG, GIF and JPEG
        * graph screenshot also supports EPS vector file format
    - 64 bits output formats handling (mono/stereo IEEE wav, txt)
    - All-Pass filters added to the Minimum-Phase Filters tab
    - measurement compensate mode to manually replicate a given measurement
      (eg replicating a mic calibration file and getting its missing
      minimum-phase response, replicating a target curve, etc.)
    - added a "Clear Result" entry in the File menu, clearing result curves
      and status
    - reworked "what's new" changelog window to make it easier to read
  Removed features:
    - removed "complex" windowing algorithm which was deprecated since
      version 0.9.0
      An alert box will popup when loading settings using it
  Bug corrections:
    - bug correction for 2nd order minimum-phase filters with Q<0.5
      An alert box will popup when loading settings containing that bug
        * bug report: [url]http://tinyurl.com/z59qw4t[/url]
    - filename encoding bugfix: accents and other special characters
      handling in setting, impulse, and measurement filenames and paths
      (was supposed to be resolved since 0.9.7, but was not...)
    - last updated graph setting sometimes remained unchanged after
      resetting or loading new settings
    - fixed a few instabilities:
        * Horbach-Keele filters with R=1
        * clipboard corner case errors
        * NaN detection bugs
    - fixed various graphical bugs:
        * generation status remaining after reset
        * misbehaved result phase curve when hide=-Inf
        * view slots flickering in large view mode
  Adjustments:
    - faster constant Q, proportional Q and constant shape EQs calculation:
      should be around three times as fast now, and should be notable when
      manipulating EQ faders as well as during the first generation step
      when a lot of EQ points are used
    - added "Pano Phase Shuffler" presets in the Paragraphic Phase EQ tab
        * source: [url]http://tinyurl.com/pano-shuffler[/url]
        * settings: [url]http://tinyurl.com/pano-shuffler-preset[/url]
    - status text under the generation button 
    - added a final optimization step in moderate/extensive/maximal modes
      with a correction factor set to 1
    - added ± 15dB and ± 18dB ranges for convenience in paragraphic EQ tab
    - increased EQ dB precision from 0.1dB to 0.01dB for manual editing
    - set default optimization floor to -100dB
    - faster startup time
    - made clear the fact the subsonic filters linarization options were
      based on optimized approximations (cf [url]http://tinyurl.com/hslsb8m[/url] ) by
      by naming them as such and adding "textbook" versions for the most
      reckless users :) 

1.0.0 2015-06-25
  New features:
    - Albrecht cosine windows implementation
      Ref: A Family of Cosine-Sum Windows for High-Resolution Measurements
    - multiple memory slots in range settings to be able to quickly go from
      one view to the other and focus on different aspects of the response
      curves
      These slots are preset with (hopefully) useful values but can be
      manually modified and copied.
    - "Load Settings From Clipboard" and "Save Settings To Clipboard" menu
      entries in order to be able to easily share corrections on web forums
    - frequency marker for the last correction point (5 sec persistence)
    - fader values can now be manually edited to arbitrary values
  Bug corrections:
    - bug correction in Minimum-Phase Filters tab: the polarity of low-pass
      Linkwitz-Riley filters of order 2(2n+1) was reversed
      (eg 12dB/oct, 36dB/oct, 60dB/oct, 84dB/oct, etc.)
      A warning will be emitted when loading correction files from prior
      versions using an odd number of such filters, as the polarity will
      now be correct and reversed compared to the prior bogus correction.
    - bug correction with higher than normal noise floor with even order
      taps (introduced in version 0.9.9 while solving a similar problem
      for odd taps numbers!)
    - bug correction with txt output file with 0.000(...)0 values
      (especially pregnant when using Hann window)
    - correction of the bogus flat top window implementation
    - corner case instabilities corrections (undue octal conversions on
      some value entries)
  Adjustments:
    - set "32 float txt" as the default output format instead of "32bit
      LPCM wav" in order to avoid  rising the result noise floor because of
      the fixed point format
    - added de-empahasis and pre-emphasis presets in the Paragraphic EQ tab
    - added Linkwitz-Riley linearization orders 11th to 16th (why not?) 
    - reduce default phase EQ range to ± 45° (was ± 90°) and removed
      unpractical ranges
    - increase default EQ range to ± 12dB (was ± 6dB)
    - added 384 and 352.8kHz sampling rates as drop menu options for ease
      of use (any other value can still be manually entered)
    - got rid of the "Curves" tab for the time being, waiting for the
      capture functionality to be implemented in some future version...

0.9.9 2014-12-10
    - shelving EQs with variable Q in Paragraphic EQ tab, with associated
      monotonic high and low shelv presets
    - centering can now be manually set to values in samples, percentage,
      time (us/ms/s) and distance (mm/cm/m).
      It is also possible to add or subtract several values, for example
      "middle+270us"
    - new centering adjustment layout:
        * 'float' is now 'use closest perfect impulse' and is explicitly
          recommended
        * 'int' is now 'round to closest sample'
        * 'use exact centering value' has been added for exact delays
    - 32 bits IEEE-754 float WAV output format added
    - output format noise floor is now shown in result curve
    - improved impulse and windowing symmetry, especially when an odd taps
      value is used
    - import/clear measurement file menu entries
    - fix partial installation catch

0.9.8c 2014-09-29
    - correction of a bug introduced in version 0.9.8 for closed-box phase
      linearization

0.9.8 2014-09-28
    - minimum-phase filters tab with common IIR filter types:
        * 1st order
        * 2nd order with arbitrary Q
        * Butterworth with slopes ranging from 6dB/oct to 996dB/oct in 6dB
          increments
        * Linkwitz-Riley with slopes ranging from 12dB/oct to 996dB/oct in
          12dB increments
    - 'compensate' mode for generalized arbitrary order Linkwitz
      Transform-like manipulations in minimum-phase filters tab
    - new centering options expressed as a percentage to easily obtain
      matched delays
    - default to "middle" centering instead of "energy" to avoid delay
      mismatch problems for the unaware user (principle of least surprise)
    - praxis measurement format handling, scientific notation in frequency
      column
    - smaller executable, new installation method
    - bug correction: crash on impulse generation with some specific filter
      settings
    - directory handling bug correction
    - measurements can now be loaded from the command line or drop on exe,
      similarly to settings
    - revamped file extension handling (settings)
    - revamped icon
    - smoothed out taps/fft size calculation

0.9.7 2013-09-03
    - Brickwall filters implementation.
      /!\ result slope relies solely on windowing /!\
      Iterative optimization and energy centering algorithms are
      automatically defeated when a brickwall filter is set, to make it
      possible to build complementary crossovers. It is up to the user to
      make sure he uses the exact same number of taps and same windowing
      algorithm on both sides of the crossover to ensure complementarity
    - sampling rate drop down menu can now also be directly edited to input
      arbitrary values, so menu options have been reduced to the most
      common values for clarity and ease of use
    - frequency, amplitude and phase ranges can now also be set to
      arbitrary values
    - optimization floor can now be set (was -40dB fixed)
    - B-weighting in optimization calculation was removed (for now)
    - new amplitude paragraphic presets with fixed frequencies (1kHz) for
      various Q values (0.5, 1, 2, 4, 8, 16)
    - got rid of scientific notation in txt output format to broaden
      compatibility
    - dark graph theme
    - various graphical bugs resolution
    - C float array output formats
    - filename encoding bugfix: accents and other special characters
      handling in setting and measurement filenames
    - "reject" filter slopes bugfix
    - "Save Settings" menu option 
    - "Save modifications" dialog box before loading/resetting/exiting
    - window title now shows the settings name instead of the impulse name
    - updated 'tips' in Linearization and filtering tabs

0.9.6 2013-04-16
    - show frequency, amplitude and phase from current cursor position in
      graph
    - improved measurement handling:
        * drag and drop loading
        * loading speed up (twofold increase)
        * gain and time offset settings
        * polarity inversion and phase hiding functionality
        * bypass option
        * description: name, number of points, frequency and dB ranges
        * ARTA format handling (trailing spaces in frequency column)
    - Save measurement inside *.rephase settings files together with
      corrections and other parameters
    - new 'constant shape' EQ, both for linear-phase and minimum-phase EQ.
      Equivalent to a constant Q EQ at 6dB, it keeps exactly the same shape
      at any dB setting. It should be preferred to constant Q and
      proportional Q at high dB settings as those two are bound to their
      2nd order definition and have to stay within a ±90° phase range,
      thus leading to odd gain shapes at high dB settings...
    - "What's new" menu entry, exposing this changelog, instead of having a
      separate REDAME file
    - bugfix when loading settings from version prior to 0.9.2: filter
      frequencies were lost
    - going back to forced 'middle' in energy centering when only
      linear-phase corrections are used
    - curve capture functionality teasing...

0.9.5 2013-04-06
    - measurement import implementation, following HOLMImpulse import rules
      and interpolation strategy
      (first draft with limited functionality)
    - Nyquist frequency is now explicitly represented in gain result curve
      as a brickwall low-pass
    - result curves are now cleared upon settings reset or loading
    - improved energy centering algorithm
    - stop forcing energy centering to middle when only linear-phase
      corrections are in use
    - middle click on a fader reset its value to 0
    - bugfix for result phase curve unwrapping
    - bugfix on curves when polarity is inverted and a phase range larger
      than ±180° is chosen
    - bugfix for 1st order high-pass filters

0.9.4 2013-03-16
    - crash at start problems (previsously requiring temp/ dir content to
      be deleted) should now be solved
    - up to 16 banks can now be used in paragraphic gain and phase EQ
    - removed bank EQ tabs (settings saved with banks EQ will be
      automatically reported to paragraphic EQ banks)
    - improved graph range options: frequency and phase range can now be
      set and saved in settings
    - new phase wrapping implementation, automatically adapted to current
      phase range
    - view mode (compact/normal/large) is now automatically saved and
      restored from one run to the next
    - double-click on a fader value entry reset the fader to 0
    - improved raised cosine EQ. Interactions between EQs should now behave
      exacly like the "Ideal Graphic Equalization" exposed here in this
      application note: [url]http://www.nordicsales.dk/imgdb/docs/lakewh_981.pdf[/url]
    - better precision for frequency entries (fractional up to 5 chars
      total to fit the entry) and appropriate up/down key binding (0.1hz
      steps under 10hz)
    - real 2/3 and 1/3 octave frequencies in paragraphic EQ sections
      (mandatory to make the raised cosine graphical EQ "magic" work...)
    - improved biquad precision (constant Q and proportional Q EQs) by
      adapting the sampling rate of each biquad to its fc
    - increased Q range (0.1 to 100)
    - improved phase deg precision in paragraphic EQ
    - added ESS sabre frequencies

0.9.3c (misnamed 0.9.31) 2013-01-29
    - mini bugfix for the taps entry...

0.9.3 2013-01-29
    - new Paragraphic EQ implementation, with multiple EQ types:
        * constant Q minimum-phase (new default)
        * constant Q linear-phase 
        * proportional Q minimum-phase
        * proportional Q linear-phase
        * constant slope linear-phase (former implementation)
        * raised cosine linear-phase (beta version...)
      ( bank EQ section remains constant slope linear-phase )
    - FFT size can now be set by user (minimum size is two times the
      smallest power of two equal or bigger than the requested number
      of taps). Setting a larger FFT size makes generation and optimization
      slower, but can increase the precision of the optimization and also
      makes result curves more precise (just a visual effect for that one
      though: no effect on the actual impulse)
    - bug correction: negative gains can now be entered directly from
      the keybord in the Gain EQ Bank tab.
    - exit on repeated errors to avoid "panic mode" effect

0.9.2 2012-11-04
    - added back '24bit LPCM mono' output format, missing since 0.9.0
    - improved up/down key bindings on frequency entries
    - added up/down key bindings for taps entry, with color warnings for
      extreme values
    - new '1st order' and '2nd order' linear-phase filters, meant to be
      combined with an existing (and already corrected in phase) rolloff
      to obtain a linear-phase acoustical Linkwitz-Riley filter
    - made sure 'middle'+'float' centering ends up within -0.5/+0.5 sample
      from middle (was -1/+0.5)

0.9.1 2012-10-29
    - centering 'int' option was not working, this is now fixed
    - more explicit error message when loading a wrong setting file

0.9.0 2012-10-28
    - new file format '.rephase', saving/loading all settings, including
      correction settings, impulse settings, and graph settings
      (old '.jason' files can still be loaded, but impulse and graph
      settings get reset)
    - impulse file is now a three-part thing: directory, filename, and
      format extension. The directory is the only thing that is not saved
      in the '.rephase' file
    - '.rephase' files can be loaded upon start (as a parameter or by drag
      and drop on rephase.exe) or by drag and drop on the user interface
    - new offset option "float" for fractional sample centering, avoiding
      HF ripples in the impulse when the phase target is not a multiple
      of 180° at the Nyquist frequency
    - make "rectangular" the default window function: this should be the
      best choice for phase-only corrections, and "complex" windowing is
      not needed anymore with the "float" offset
      Note: "rectangular" window is still likely to be the worst choice
      for filter generation, when gain target goes far below 0dB...
    - new "ovelapping" filters, to be used for example in the midbass
      region, under Schroeder's frequency...
    - stereo wav formats are now available
    - bug correction in offset calculation in time=inv mode

0.8.4 2012-10-14
    - new time inversion option in general tab, to reverse the generated
      impulse, thus opposing phase corrections. This can be used to better
      visually track a phase target (inverse during correction, and return
      to normal before generating the impulse), or to evaluate the
      audibility of a given correction with headphones for example (in this
      case the convolution of the impulse will simulate the speaker before
      correction)
    - confirmation box when exiting whithout saving modified correction
      settings, and avoid asking for confirmation on reset when correction
      where saved or loaded without modification
    - Improves advice section in the linear filter tab, and add one in the
      linearization tab

0.8.3 2012-10-12
    - range choices in paragraphic EQs (up to ± 48dB and 720°)
    - improve arrow keys binding after click in faders
    - stop constraining frequencies to 16Hz-25khz in paragraphic EQs
      (now 1hz-99khz like in EQ banks)
    - ask for confirmation before resetting correction settings
    - bug correction when resetting settings ("wrong format")
    - stop using '.rephase' as default extension when saving an impulse

0.8.2 2012-10-07
    - try to play nice with multiple screens
    - new "large" layout, and "View" menu for layout choice
    - change Q interpretation for phase EQ to be more in line with gain EQ
      (to maintain an ascending compatibility, phase corrections saved from
      versions 0.8.0 and 0.8.1 get their Q divided by 1.8 upon loading)

0.8.1 2012-10-03
    - fix small bug with slope/ratio display when loading a FIR filter

0.8.0 2012-10-02
    - real-time amplitude/phase curves for both target and result
    - save/load correction settings (/!\ beware /!\, still experimental)
    - Horbach-Keele 'last' ratio (special tweeter) is now a different
      filter type for ease of use and clarity reasons
    - added some more window functions

0.7.6 2012-09-20
    - resolved (hopefuly) some issues with windows XP with the program
      refusing to actually start

0.7.5 2012-09-18
    - bug fix (crash during otpimisation step)

0.7.4 2012-09-16
    - bug corrections
    - optimization iterations are now faster
    - new optimization options ("moderate" and "extensive")
    - Horbach-Keele filters
      (ratio above 4.5 is the special "tweeter" ratio)
    - "Reject low" and "Reject high" filters for higher low or high rolloff

0.6.0 2012-08-26
    - first version on SourceForge

Attachments

  • FIR.PNG
    FIR.PNG
    61.7 KB · Views: 18,183
  • PEQ.PNG
    PEQ.PNG
    55.6 KB · Views: 17,772
  • IIR linearization.PNG
    IIR linearization.PNG
    55.9 KB · Views: 17,500

GB for Salas Reflektor-D Power Supply for Digital

This is a GB for Salas Reflektor-D and Reflektor-Mini Power supply.
This is a Low Voltage supply designed specifically for digital.
3.3V to 7V at 600mA max output using board level sinks.
Schematic and build info in PDF attachment at the bottom of post.

parts_pads.png


In the words of Salas -

The Reflektor concept is a shunt voltage regulator made around a current mirror circuit. To keep equal current between its input and output legs, the mirror reacts to flow disturbances and drives a MOSFET in opposite direction. A current loop simple regulator is another way to describe it. Specifically Reflektor-D is a version focusing on the voltage region common for powering digital source projects. For convenient mounting and setting it utilizes board level sinks and simple ways to choose its current limit and voltage output levels


The board is offered in a Matte Black 2MM thick PCB with 2 OZ copper. The cost is $16.00 plus shipping.

image.jpeg



It was a very easy build, it took me 45 minutes to complete and then start testing, and got near target voltages rather easily via the LED swaps.
Three different kits will be offered. A transistor kit, a resistor kit, and a full kit, which includes all parts for board except standoff parts and MOSFET bolts and nuts for the heatsink.

image-2.jpeg



The Reflektor-Mini was created to be installed into some tight spots. It is a much smaller board that will require its MOSFETS to be chassis mounted and electrically isolated for sinking. It keeps a lower height profile too. It is DC input only, needs an incoming DC feed.
It uses a J2 2SK117GR to stabilize and tune output voltage. For keeping small size It has a conventional 2Pin output. It's a slightly more complicated build than the original in some aspects and simpler in some other, but will serve it purposes in being a potentially lower cost unit to construct, as well as a smaller footprint.





In conjunction with the Mini release, the DC Flexy board is introduced. It allows for use of industry standard Snap-in Elcos or 4 pin/pole Mundrorf M-Lytic AG+. These boards can be used as Raw DC units to front end the power supply for the Reflektor-D Mini. They come in a breakable pair so to be used in double mono or multi isolated configurations of more than one regulator to feed. if used in a system of one regulator only, the spare may be broken off and kept aside since it can be utilized in a variety of projects needing Raw DC. Even for the standard Reflektor-D when done in its DC input mode version.

DCFlexy.jpg


The signup for the boards is in my signature.

Attachments

GB for Salas SSHV2 regulator

This is a group buy for the purchase of the second generation Salas Shunt High Voltage Regulator.

It's a newer design that Salas did, with CRT brilliant layout work and a bunch of us prototyped over the holiday season. We had a zero percent failure rate of setups amongst us. (Which I consider great, as most of my work , well needs work)


Let's get to the point.
The board is $15 USD.

Mini-Kits will be available. The cost is higher than I wanted them to be - so I split them in two.

img_05841.jpg



Again a disclaimer - this is the most dangerous product I have shipped yet. Failure to respect the high voltage of the product can have deadly consequences. This is not a beginners project!

The full specifications are in the PDF file. (attached as zip)

SSHv2Works.jpg


The google spreadsheet needs to be completed with your username and includes paypal payment details. Please don't post that address here.


Transistor Minikit 13 USD
1-Toshiba 2SK117's unmatched
3-Vishay 12V .5W Zener
2-20ma Red LED's
2-Supertex DN2540
1-IRF840
2-Fairchild KSA1381ETSU
2-TO-220 Mica Insulator
2-Keystone plastic washers

Resistor Minikit 9 USD
2- Dale CPF2 100ppm 68.1k
1-KOA M/F 1/4W 100ppm 1.82R
1-3296Y Bourns 1/2W 100ppm 200R
1-T93YA Bourns trimmer - 1/2W 100ppm 1K
3-Vishay Bayershlag 50ppm 100R
1-271 Series Xicon 1/4W 50ppm 10R
1-271 Series Xicon 1/4W 50ppm 1.8k
1-271 Series Xicon 1/4W 50ppm 470R
1-271 Series Xicon 1/4W 50ppm 220R


SSHV2 FULL KIT - Does not include board (And no BIG heatsink for IRF840)
$45

1-Toshiba 2SK117's unmatched
2-20ma Red LED's
2-Supertex DN2540
1-IRF840PBF
2-Fairchild KSA1381ETSU
2-TO-220 Mica Insulator
3-12v 1/2W Zener
2-Keystone plastic washers (above is transistor kit)
2- Dale CPF2 100ppm 68.1k
1-KOA M/F 1/4W 100ppm 1.82R
1-3296Y Bourns 1/2W 100ppm 200R
1-T93YA Bourns trimmer - 1/2W 100ppm 1K
3-Vishay Bayershlag 50ppm 100R
1-271 Series Xicon 1/4W 50ppm 10R
1-271 Series Xicon 1/4W 50ppm 1.8k
1-271 Series Xicon 1/4W 50ppm 470R
1-271 Series Xicon 1/4W 50ppm 220R (above to 68K resistor kit)
1-Molex 4P Fixed Terminal Block Black
1-TE Connectivity 2P Terminal Block
2-274-1AB Wakefield TO-220 Heatsink
1-647-10ABP Wakefield TO-220 Heatsink
1-Wima 10uf MKP4 1100v
1-Wima .33uf MKP10 630v

Attachments

GB For Salas SSLV1.3 Ultra-BIB

SSLV1.3 - The Ultra-BIB
The group buy is for an updated Low Voltage shunt reg designed by Salas. This updated board uses no NOS JFETs making parts to supply and costs easier to predict. The circuit has been updated an initial clinical listening test have been positive, and noted as an improvement to the original BIB design. The board uses a universal set of components. So there is no need for calculations for voltage setting etc. As long as the appropriate transformers are used, the voltage can easily be set with the components listed in the schematics. The R1 current setting resistors will need to be calculated. This should make buying components for the project less worrisome.

Build Thread SSLV1.3 "Ultra-BIB"

Salas SSLV1.3 UltraBiB shunt regulator - diyAudio

Salas thoughts..
As the beloved SSLV1.1 BiB shunt reg was getting long in the tooth especially for NOS JFETS I had in mind for some time now to design its successor. The goals were: 1. In production parts 2. Much simpler to set up. 3. Better technical and subjective performance.

After many breadboard experiments and two prototype PCB iterations I feel that my goals were finally met. So here comes the UltraBiB

-Uses no NOS parts. (This was a short lived story, now the PF5102 and BC560c are NOS, but I have thousands)
-Can do 5V to 40V output without changing a thing in its configuration.
-Nothing to choose and match. No tolerances in predicting its CCS limit setting.
-Has 45dB more open loop gain and many times less output impedance than 1.1
-Sounds easily better.
-Its an electrically and mechanically drop in replacement for an upgrade.

Some initial quotes on sound

From Dimdim
"When we swapped it in place of the BiB 1.1 in my Soekris, the improvement was immediately obvious and not subtle. There was a general improvement in clarity and silence, but the biggest improvement (imho) is that the music appeared to have more energy in the lower mid area, where before it was kind of "dry". This was with Salas' very first prototype, built with standard (non-boutique) components. The board that I built with audio grade capacitors in the filter bank and MUSE BP caps in the output sounded even better."

From VGeorge
"The change in sound to better was apparent at first listen. As other have described, better clarity and definition throughout the audio range, but for me it was also apparent up high the frequency where I could hear more power but without any harshness."



The board will consist of three different reg's on one board. The board is V-Scored and can be broken into three boards, two boards etc.

On each full board there is a two positive regs and one negative rail reg.

Extra mini-kits (partial kits) are available to support building of the board - but leave out C1 Filtering cap and most resistors .

The boards will sell for $25 each, with $8 minimum shipping USA and Canada 10 min International, paypal only. Shipping has increased substaintially for international shipping, so I can only send a limited amount of boards and kits before it becomes a "package" as opposed to a "flat" and the cost typically doubles.


BOM List for board
UltraBIB Minikits - Google Sheets

UBIB Positive Minikit - Please note if needed 5V operation, J3 must be <7.5ma. You must also use RED leds here the 1.85 vF type. Please add this to private message so I can put a measured one to put in your kit. WP113IDT is a sub for WP1773ID LED stated below.


UBIB Negative Minikit
  • Like
Reactions: Alexlau 84 gogo

Marantz PM94D mosFETs

Hi! I'm new to electronics diy, basically retired and since no one near me wants to fix my receivers, yep, they will receive the service they need...somehow. my immediate issue is with the marantz pm-94 and finding replacement mosfets. The factory ones are 2sk405/2sj115, which are long gone, and according to an article that I read, somewhere, the replacement for those are 2sk1530/2sj201, which are no longer available. I really love this amp and would love to find a great pair of replacement mosfets. Thank you all!

Tony

Audio Research D115 repair

Hello does anyone have any detailed photos of the top and bottom circuit boards of an Audio Research D115mk11. Specifically I'm lookin for where the op amps, pass transistor and V17 tube socket are mounted. These were all removed from an Amp I just bought and I want to restore it back to original. There may be a small circuit board somewhere that had the op amps mounted on it? There are all kinds of parts and circuit modifications on this amp.. I need some clear photos to see how the original was. I do have the schematic but a board layout would save hours of time. Thanks anyone that can help. Guenth

Human frequency PERCEPTION range? (its not 20-20kHz)

Hello found these two quite interesting videos i wanted to share

Login to view embedded media Login to view embedded media
Cool guy! he makes a lot of these edge-case videos, quite interesting stuff

personally i definitely feel 0-20Hz
and i also can hear a lowpass set at 20-22khz even tho my hearing range rolls off after 15khz

while science claims 20-20khz is our hearing range, our actual "perception" of different frequencys might be complete different

Whats your expierence or thoughts on this?

I2S-Hat: A Raspberry Pi Hat for SPDIF <-> I2S Communication and DSP

For the past few months, I have been working on a project to utilize a Raspberry Pi with CamillaDSP in a standalone, fully-digital DSP system. To this end, I have created an add-on board, also known as a "hat", for the Raspberry Pi that allows fully bidirectional SPDIF <-> I2S communication with the Pi. As an advantage to other designs on the market, it also performs sample rate detection, providing the information to the Pi GPIO, and allows full software control of the SPDIF transceiver IC. No resampling of the incoming/outgoing digital audio is required, and it supports all stereo formats from 44.1kHz to 192kHz as tested.

The DSP capability of a modern Raspberry Pi 4 greatly exceeds that of the ADSP/SHARC implementations available. This should open up significant possibilities for much more complex and accurate DSP.

Please read more about the project on the GitHub page:

GitHub - raptorlightning/I2S-Hat: An SPDIF Hat for the Raspberry Pi 2-X for SPDIF Communication

KiCad files, Gerbers, and code are all available for anyone to use for their own build. This post is targeted for an open discussion about the board and implementation. Please let me know if you have any questions or comments about the information on the Git page, or if there is anything I can clarify further.

Enclosed.jpg

Electrocompaniet Ampliwire 100 With Issues

I've got one with issues, I'm WAY out of my comfort zone (just finished fixing a Marantz 2330B but I had the crutch of a SM and lots of internet archives).

Powering up on a 60W DBT bulb I got a malevolent glow but not being experienced with the amp I thought that moght not be enough current. Switching to a 100W DBT I got a puff of smoke (see pic for 220ΩX2), after disassembly I thought I found a shorted transistor but upon further review the trace is actually shorting them.

I'm pretty new but I don't think this is a thing?

I have the HFE schematic but I think I have the other one because they don't seem to marry up.

Anybody with AW100 experience is welcome to chime in.

PXL_20230802_022754450.jpg


2.jpg

PXL_20230803_022128953~2.jpg

For Sale Apex PA88 High Voltage (+/-250V) Op-amp modules

I have a pile of boards that each have two Apex PA88 high voltage op-amps on them with heatsinks. these are rated for something like 450V and 100ma each. the boards were used to drive Piezo sensors so the are set up with a single ended input, One opamp inverting and one non inverting with a bridged output. power requirements would be +/- 200v up to 225v max. so they could output 400-450V P-P at 200ma! gain is set around 18-20db something like that but can easily be changed.
The Apex PA88 are crazy expensive new. so they have been hanging on to these old boards and have finally decided to let go of them. so I am curious if they would have any use for audio stuff? maybe for driving electrostatic panels directly? or as Tube drivers or. driving output transformers directly for testing or something.

$20each plus shipping

Attachments

  • 20240827_185344.jpg
    20240827_185344.jpg
    348.7 KB · Views: 153
  • 20240827_185346.jpg
    20240827_185346.jpg
    370.5 KB · Views: 152
  • 20240827_185356.jpg
    20240827_185356.jpg
    368.1 KB · Views: 149
  • 20240827_185422.jpg
    20240827_185422.jpg
    400 KB · Views: 148

My first attempt at winding a transformer + iron core analysis and electron scanning

Good evening, I wanted to share the DIY story (2 years old now) on how I winded my first mains transformer starting from a broken one.
DISCLAIMER: I am a hobbyist, I do not claim my work to be professional or that this is the correct way to do it, this is just A way to do it.
All the information relative to this are well written and illustrated on my blog
https://www.mimifactory.com/, otherwise this post would've become too long.
So if you wish to understand more please visit the link.

1) I started by taking the old transformer apart, removing the burnt copper windings and the lamination (being careful to not remove the thin layer of insulation on them, you WANT your lamination insulated to avoid eddy currents in your iron core and prevent power losses)

2) I winded the copper, using nothing but my hands and two whole spools of double enameled copper wire. At the end of the page in the link you can find the calculations of the necessary amount of windings, you don't want to exceed the max magnetic flux density of your metal and turn all that energy into heat.
Transformer1.jpg
Transformer2.jpg
Transformer4.jpg


3) Put the lamination back in, then used a vacuum chamber to evacuate the transformer and replace the air with insulating varnish

Transformer5.jpg
Transformer6.jpg
Transformer7.jpg

4) Drying, assembling and testing are required before using the transformer in the device.

photo_2023-12-07_00-40-27.jpg
Transformer10.jpg
Transformer11.jpg

I didn't have much hope this would work, but the device is still operational after two years and the transformer doesn't hum or vibrates, indicating that the vacuum and varnishing really did its job. The amplifier is a classic 35+35w 2SC5200/2SA1943 from the '80s recovered at the scrapyard. The smoke detector is always on, just in case...

In the webpage I included more pictures and an analysis of the composition of the metal core that helped to understand the parameters for the right amount of turns.

Bench power supplies...

I'm starting to dabble in solid state a bit (mostly a tube guy) and want an inexpensive bench power supply to help. I have a Lambda high voltage supply that was given to me for working with tubes but that won't help with ss. My first ss project is a subwoofer amp so I need something that can handle 150W output or more, ideally under (or close to) $100. I'd like it adjustable to at least 30v, but 60v would be ideal. I've been looking around and new adjustable switching supplies are everywhere and can be had for under $100. I've also found a few linear supplies that might fit the bill, but they tend to be much lower current for a given voltage and price. I'm not looking for a high-end, highly accurate supply here (though maybe I should...).

So I have a few questions. First, I know linear supplies are in general quieter (in terms of output) than switching, but how important is that for building and simple testing? And if it is important, could I add a small cap across the outputs of a switching supply (small for higher frequency noise) to help reduce that noise? I've read that many switching supplies are also not very good at fast transients so I assume I could in theory also add a large reservoir cap. Basically, will I be happier with a linear supply even given the lower current/power? Any recommendations for something that fits the bill?

Or, should I DIY a linear supply to meet these general goals? I did some research on this many years ago but gave it up as too complex. I think I could do it now. If so, can anyone suggest some designs? I could probably design a very simple supply, but am looking for something a bit more carefully designed than I could do.

OK, I'm not a highly skilled builder (though I've built quite a few amps (tube and solid state), preamps, phono-stages, speakers, etc.) and I understand that this may be a lot more complex question than I realize, but some general replies would be helpful. At least for now. Thanks so much!

DIY bass sound absorbers

Since sound absorbers like I need are not sold, I came up with the idea of building them myself.
I need damping in a relatively narrow band on bass frequencies, at about 40, 60 and 80 Hz, it makes sense to build dampers with a membrane.
I found one page what seems to give easy absorber calculations: http://mh-audio.nl/Acoustics/PResonator.html
I also found an example where the damper made based on the previous calculation seems to work https://www.musiker-board.de/threads...echnet.741282/
You can see see results on example link on the post before last on pictures with header "Zur besseren Übersicht vorher - nachher", I used Google translate to read this page.
However, there are small doubts for me.
How the calculation sheet doesn't take into account membrane stiffness at all, only mass and panel depth?
How can such a small absorber have such a large effect on 30 Hz as seen the sample page images?
Did I miss something here?
Have someone other bass sound absorbers built ideas what can be DIY?

Kicker ZX700.5 low volume from Amp 1

Hello I have a Kicker zx700.5 that I got at a yard sale. Amp2 seems to work fine but amp1 outputs have very low volume in comparison like it's muted. Sub works fine. I can hear something buzzing on the amp but I haven't located it yet. Any suggestions? Thank you

Here are some measurements I took from the preamp board to speaker ground:

Fr in -18.3 mv
Fl in 9.6 mv
Rr in -19.3 mv
Rl in 10.0 mv
-15v -14.14 v
+15v 13.95 v
Rem 11.51 v
Gnd a -4.5 mv
+ Batt 11.63 v
Gnd p -4.5 mv
Rem- bas 12.3v
Sub in 15.3 mv
Sub pre -4.4 mv

Attachments

  • PXL_20250211_170556286.jpg
    PXL_20250211_170556286.jpg
    644.9 KB · Views: 65
  • PXL_20250211_170618219.jpg
    PXL_20250211_170618219.jpg
    664.1 KB · Views: 56
  • PXL_20250211_170626019.jpg
    PXL_20250211_170626019.jpg
    743 KB · Views: 56
  • PXL_20250211_170636476.jpg
    PXL_20250211_170636476.jpg
    618.7 KB · Views: 52
  • PXL_20250211_170802270.jpg
    PXL_20250211_170802270.jpg
    690.3 KB · Views: 52

Steinway Lyngdorf Model B

I’m trying to confirm how the 6 woofers (3 forward facing and 3 rear facing in adjacent vertical arrays) are configured in this speaker, including the baffle arrangement. It appears from some photos and videos that I’ve seen that the front and rear baffles only extend half way across the width of the speaker so as to allow the sound wave emerging from the rear of each set of 3 speakers to travel to the front or rear of the speaker, as the case may be. To put it another way, both sets of 3 woofers are operating as true dipoles and are projecting sound into the room in both directions.

Is this correct?

Also, presumably the rear facing set of 3 are electronically out for phase with the front facing 3 so that all 6 push in the same direction at the same time - ?

Troubleshooting an NAD T751 With Low Output on One Channel

I am more of a tube radio guy, but I am trying to figure out a low output channel on a friend's solid state amplifier. I have tracked down a number of symptoms and I am hoping that someone might be able to help me with a diagnosis. I would be very appreciative of any ideas you might provide. The schematic for the Front Amplifier board is shown, but I am not sure if it will be legible.

  • There is very low output on the front left channel, regardless of the input source. (Yes, speakers have been switched - problem remains on the left side.)
  • When the receiver is turned on, the left channel is too quiet to hear, but slowly increases in volume until it is about 25% of the right channel. The sound is intermittent until the volume is increased, when it becomes steady, though quiet.
  • A number of component in the left channel are warmer than those on the right. For example, R521 (right channel) is 44 C, while its corresponding resistor on the left (R522) is 51 C. R523 is 31 C and R524 is 44 C. More interesting is that C525 (470 uF / 63 V) is 29 C and C526 about 36 C.
  • Measuring the Voltage at the idle current measurement points, I found the right channel measurements at P501 to be 4.9 mV DC / 0.1 V AC, whereas the corresponding left channel measurements at P502 were 21.6 mV DC and 0.02 V AC. Cap leakage?
  • The problem is unlikely to be in 10,000 uF filter caps (C561 and C562) as they feed both right and left channels.
It seems to me that the problem is likely to be a leaking electrolytic cap on the left channel rather than a bad transistor and that I should remove the board and replace the electrolytic caps. Does that make sense to the more experienced techs?

Thanks as always for any help/suggestions/insights/next steps you can provide.

Andrew

1730310269121.png

Electrocompaniet Ampliwire 100

I have an opportunity to pick up an Ampliwire 100 with a shorting problem. Doing a quick take I was able to determine that the problem is from the right channel after the power supply. By any chance does anyone know if this particular model amp is plagued with problems? A few folks seem to think so - I just do not have much history on its reliability.

Thank you for the help.

what do you really think of Wilson Audio?

yes we all know that anyone in this forum can make much better speakers than wilson audio does, but still, what is it that wilson audio speakers do so well that so many audiophiles love? and audiophiles and critics have loved wilson audio speakers for a long time now!

it seems absurd that wilson audio would be a bad speaker manufacturer but still have so many fans

https://www.stereophile.com/content/icing-munich-cake-mcgrath-fon-nagra-wilson-impex

Restoration and Modification of Pioneer PL-71 turntable

This thread will be for the documentation of the restoration and (probable) modification to a Pioneer PL-71 turntable.

Why the PL-71? A couple of reasons:

1) It's a good example of pre-PLL direct drive. The motor is very quiet and it's bearing structure are good.

2) It has a wonderful tonearm made by Acos.

3) The turntable is generally considered to be a good example of "better than the sum of it's (quite nice) parts".

4) It's the turntable I grew up with, and 5) I didn't actually expect to win the auction.

Fantastic reference thread here - A new toy - PL-71.....

IMG_2428.jpg

Here it is in the condition I recieved it. The wood is a bit dry, there is a general light yellowing of all the metal parts indicative of it living in a smoker's household at some point, but other than that, it's in very nice shape.

IMG_2431.jpg

And here it is after a thorough cleaning. The yellowish tint is off the metal, the wood has a coat of Danish oil, and generally it look much, much better.

I didn't take any photos as I was cleaning it, just imagine a bunch of paper towels, cotton swabs, alcohol, wood oil and the like all strewn about. It took approximately 1.5-2hr of scrubbing, dabbing, cleaning, wiping, brushing and elbow grease.

I still need to treat the mat, it's clean, but the rubber needs something to restore a bit of moisture to it.

IMG_2433.jpg


IMG_2435.jpg

The dustcover is is great shape for it's age. I'm very pleased.

IMG_2436-1.jpg

The bottom cover is very 1973. But the metal chassis bottom is a neat piece, making a metal interface for the sprung feet into the wood chassis that makes up the rest of the table.

IMG_2437.jpg

Here's the up-skirt shot.

IMG_2448.jpg


IMG_2441.jpg

A few things worth mentioning, the tonearm is rigidly coupled into the chassis, and with the tagboard and jacks easily removed, would be a good candidate for a continuous rewire or conversion to DIN if that's you kind of thing.


IMG_2439.jpg

This board is mainly for AC distribution.

IMG_2438.jpg

DC rectifier and 'regulator' (Really just a zener-referenced cap multiplier.)

IMG_24401.jpg

The power/speed selector switches and speed trim pots.

IMG_2442.jpg

These little screws holding the motor cover were a royal pain to remove.

The power transformer is mounted on rubber feet. Remarkably quiet. Of course it would benefit from being in it's own external case, and I may try that. It also has a universal primary and voltage selection with one of those neat plug thingys.

The motor is rigidly coupled to the chassis and the control board is under the black cover.

IMG_2445.jpg

Cover removed showing the motor control PCB. The 38yr-old capacitors need to be replaced. (Yes, one is already replaced in this photo...)

IMG_2446.jpg

Not too bad of a job. There are a number of wires that got the the motor windings on the other side of the PCB that seemed to be in the way, but other than that, it's straightforward. No values changed

IMG_2443.jpg

The regulator board also got a set of fresh capacitors. The filter caps are a bit bigger than stock, but the can size is still the same. 😎

IMG_2447.jpg

Dead soldiers.

IMG_2450.jpg

I replaced some screws that hold the strobe to the chassis, over the years 3 of the 4 screws had fallen out. I had to scrounge for hardware that fit, but now it's solid.


IMG_2451.jpg

Lastly I added some secret lubricant to the bearing, I got it from a kindly old Dutchman who horse-swapped it from a Polar Bear named SY. I have little idea what's actually in it, other than it was made for low-heat, high-pressure interfaces, specifically TT bearings.


In my opinion, all I did was to get the 38-yr old table back in a condition similar to when it left the factory. So far it's just a tune-up, no hot-rodding, no mods.

Yet.

🙂




Edit -

There has been a number of reports of people trying different mats, and always returning to the stock one. I have a theory why -

(click link for video) https://www.youtube.com/watch?v=zxbj0a-AzuI
  • Like
Reactions: Bonsai

Electrocompaniet Ampliwire 100 (AW 100 AW100) different Versions

From this power amplifier I have service manual together with two different schematics (year of production approx, between 1979 and 1983). Additional I have newer version for repair service. The schematic, that I have create (Reverse Engineering), is different again compare to the already present orig. versions. Even the cabinet version is different in opposite to the older version.
How many different versions of the AW 100 are exist at whole ?
Thank you for your advices.
  • Like
Reactions: Nessman

DIY 4 Phase Sinewave Generator for Turntable Motor Drive

This is a shared DIY project for non-commercial use.

The SG4 generates 4 low distortion, high accuracy sinewaves suitable for driving conventional audio power amps to create a multi-phase drive for turntable motors. The generator outputs a reference sinewave at 60/81Hz or 50/67.5Hz on the 0° pin. The 90° pin outputs an exact replica of the reference sinewave, but shifted +90° (Cosine) for driving 2 phase AC synchronous motors. The 120° & 240° pins output an exact replica of the reference signal, but shifted +120° and +240° respectively for driving a 3 phase motor. The SG4 is a sinewave generator only. You will need to add the necessary audio Power Amps and step-up transformers (if needed) to create the final signal to drive the motor. Low cost linear and class D amps are readily available on e-Bay and other on-line sources. Working with High Voltage can be dangerous. Do not attempt this part of the project if you are not trained in handling power electronics: Seek competent technical help if needed.

The PCB uses all thru-hole components for easy assembly, but some soldering skills are still required. The µP is a PLCC package but there is a socket for it on the board.

The project consists of a bare PCB, a parts "kit" available as a shared cart from Mouser electronics, a µP with the operating system pre-programmed into it and the on-line documentation you see here.

The PCB is available from OshPark PCB fabricators at the following link: https://oshpark.com/shared_projects/E5zXjeJd
The PCB is created in multiples of 3 for a cost of ~$45 or $15/board.

24-Feb-2022 The parts kit at Mouser has been updated to include the MCP101 reset controller that replaces the DS1833 part. The pin out of the MCP101 is the mirror image of the DS1833 so the PCB has also been updated to Rev C to reflect this. If you are using a Rev A or Rev B PCB, you must insert the MCP101 part backwards; the Rev C PCB does not require any change.

The parts kit can be ordered from Mouser Electronics: http://www.mouser.com/tools/projectcartsharing.aspx
Enter the Access ID code: 7A6A645FFA. The parts kit to build 1 PCB costs $32.34.

The pre-programmed µP is available in the US from DIYAudio member Seth Hensel via email: sethhensel (at) icloud.com. Cost will be ~$12 plus $8 handling plus shipping.

The pre-programmed µP is available worldwide from DIYAudio member ralphfcooke via PM. Cost will be ~£10 plus shipping.

The following documentation is available below to aid in construction of the project:

SG4 Schematic.pdf
SG4 Parts Locator.pdf
SG4 Assembly Instructions.pdf
SG4 BOM.pdf (Generic bill of materials with part references)
SG4 CART.pdf (Mouser cart with mfr's part numbers and costs)
SG4.zip (Gerber X274 files if you want to use your own PCB fabricator)
SG4.png (X-Ray view of the PCB w/traces, pads and silk screen)

The SG4 uses a 20 bit DDS core implemented in software to generate the reference sinewave. Frequency resolution is 0.01Hz. Frequency range is 40.00-70.00 Hz for 33 RPM and 60.00-90.00 Hz for 45 RPM. There is an on-line video showing the frequency operation Here and a video for phase adjustment Here.

There are four 8 bit phase accumulators to generate the four phases. The reference signal is fixed, but the other 3 are adjustable in 1.5° steps ±15° maximum. D to A conversion for the 4 signals is done with 8 bit PWM at 18kHz. There are 4 LPFs on the board to convert the PWM signal to analog. The outputs are DC coupled, 5VPP and centered at 2.5VDC. Distortion is ~0.5% (-46dB) and frequency stability is 30 PPM.

Update: The firmware has been updated to version 1.02. A 7 bit linear taper attenuator has been added to ramp the voltage from 0 to 5VPP at start up when exiting standby mode. This should prevent the amplifiers from shutting down when first started as they will have time (~650mS) to overcome the core magnetization of the transformers. I also added a reduced output voltage mode, where the output voltage will automatically be reduced to a user programmed level after 5 seconds. The level is adjustable from 128 (maximum) to 64 (half voltage) in ~40mV steps which equates to ~1V steps at the transformer outputs.

19-Nov-16: I returned the phase adjustment to 1.5°/step. One of the peculiarities of using a 16 bit phase accumulator with an 8 bit DAC (PWM in this case), is the limited precision math can create different points where a carry occurs (and thus an additional phase step). For most frequencies, this isn't noticeable as the difference in phase shifts is usually in the mSec range. In certain cases, the math works out where it becomes quite noticeable and in the audio range where it could affect performance. Using a 16 bit phase accumulator in the SG4, a nasty phase spur will occur on either side of 81.92Hz, which is fairly close to the frequency needed for 45 RPM. The new firmware hasn't hit the field as of yet, so there will be no need to do another exchange. V1.02 will ship with the attenuator capability, but will retain the 1.5° phase adjustment of the original.

Users who upgrade to the new firmware should perform a Factory Default Reset after installing the new firmware.


28-Dec-2016: Just added the PCB files to OshPark for a Rotary Encoder to SG4 interface PCB. The circuit converts the 2 quadrature signals from the encoder to a single pulse train on the UP pin when turned CW and a single pulse train on the DN pin when turned CCW. The momentary push button switch of the encoder is connected to the STBY button of the SG4. This allows all of the normal operating functions to be performed by one rotary control.

The 1 inch square board uses all surface mount components in order to keep the size down. The IC is a CD4013 in SO14 package. C1 and C2 are both 0.47uFd Tantalum caps 10V rating in a 1206 size package. R1, R2 and R3 are not necessary if you use the Arduino Rotary Encoder which has the pull up resistors already on its PCB. If you use another encoder without pull ups, add the 3 resistors (all 10K 0805 size). Vcc is connected to the 5VDC output of the regulator on the SG4.

You can order the PCB here ($5 for three):

[url]https://www.oshpark.com/shared_projects/AbsVI39H[/URL]

The Encoder PCB can also be ordered with thru-hole component layout instead of SMT:

[url]https://www.oshpark.com/shared_projects/W6QWOCOF[/URL]

1-May-2017: Version 1.03 of the SG4 firmware reduces the lower frequency limit to 1.00 Hz for both 33 and 45 RPM. The reduced lower limit was needed for 3 phase BLDC motors, some of which require 20Hz for 600 RPM. If you are not using a 3 phase BLDC motor, there is no need to update the firmware.

16-May-2017: A DIY 3 phase amplifier project is now available to drive a BLWR172S-24-2000 or BLWS231S-24-2000 BLDC motor from Anaheim Automation.
It is not a universal controller and will only work with these two motors: 3 phase class D DIY BLDC motor drive amp


A suitable audio power amp and step up transformers are available here: 60-wpc-amplifier-diy-turntable-motor-drive


13-Dec-2021 The PCB has been updated to Rev B and the firmware has been updated to v1.04. These changes add the ability to use a 2 x 16 character LCD display that has an I²C interface PCB with a PCF-8574A interface chip. Version 1.04 firmware is backward compatible with the previous PCBs so an LCD display can be added to a previous build. The details of all the changes can be seen here: SG-4 Version 1.04 Update

It is important the LCD interface chip is a PCF-8574A and not a PCF-8574; the two chips work nearly identically, but have different address schemes. The new SG4 firmware will only work with the PCF-8574A addressing.

Attachments

Trying to Restore Mitsubishi DA-R35, Zener Diode Question

I have a couple of these nice 1980 receivers. Not top shelf material, but quite good. They use ICs for L/R voltage amplifiers (STK 3076/3106) and individual ICs for L/R power amps (STK-1080 II), three ICs in total 😕. Both have blown voltage amplifier ICs, but I am finding other surrounding problems such as 4558 op amp blown and zener diode MZ324.

Does anyone have an idea on this MZ324 zener diode and its specs? I need to find a current match.

I am attaching the part in the schematic with the MZ324 zeners highlighted in red. Maybe someone can have an educated guess as to one that will work in both positions.

Thanks in advance.

Attachments

  • ZENERS.png
    ZENERS.png
    211 KB · Views: 309

New project - PSU confusion

Hi,

Got a visit from my neighbor yesterday while enjoining my ACA + Korg, apparently it was to loud. Enter Honey Badger.

Still going through all the docs here and creating BOMs so I can pull the trigger on the parts, and of course have a ton of questions.
As the amp section is pretty much as is regarding the PCBs and parts (except some transistors and caps, still figuring that out...), I'm not sure about the PSU. I see there are two on the store: Nelson's bipolar PSU and Universal PSU.

The idea is to go with the 2x150W stereo config, and update to mono-blocks later if needed....

1. Which PSU to go with?
2. One or two PSU boards?
3. What spec toroidal to match with the PSU board(s)?

As always, much appreciated for all the help!

Cheers!

Cambridge 640A V2 with DAC module Musical Fidelity V-Dac V1

Hello

I modified my amplifier Cambridge 640a V2 & I install inside the amplifier a dac module from Musical Fidelity V-Dac V1 (Usb-Optical-Coaxial Inputs)
I need space for this Dac and I remove the A-Bus PCB Board, and I disconnect on Input PCB Board, connectors CN9 and CN10 (From A-Bus PCB Board)

The amplifier working fine, but like you, the remote control is not working !!!!!!

I am not a specialist of electronic systems
Do you have a solution to solve may problem, because I saw in the past you have same trouble
Let me know
Regards
Rabia

Attachments

  • 640A V2 - A-BUS PCB Board.jpg
    640A V2 - A-BUS PCB Board.jpg
    228.5 KB · Views: 22
  • 640A V2 - INPUT PCB Board.jpg
    640A V2 - INPUT PCB Board.jpg
    138.3 KB · Views: 23

Board 3 Way DSP Amp

I will post the availability and price of the 3 Way DSP Amp boards here.

TOP Plastina+ESP32.jpg


Read more about the boards here.
https://www.diyaudio.com/community/threads/3-way-dsp-amp.415065/#post-7734899


The price of one 3 Way DSP Amp board is $95.

The price of one I/O board is $15.

Delivery to Europe and America is at my expense. Delivery to islands or other distant places is negotiated separately.

Order more than one 3 Way DSP Amp board and get $5 off.

Payment via Payoneer.


We currently have four 3 Way DSP Amp boards and four I/O boards with ESP32 available.

For Sale Miro AD1862 DAC

I am selling my fully built AD1862 DAC based on Miro's design. Used all good components like DAC chips from Rochester, Opamps - OPA1655DR and OPA1611AID and I love both of them so will include both in the sale. JLSounds i2s dac board sitting on top of the dac which is USB powered and I have not modified it to power externally. I like this way as I use a galvanically isolated USB from my streamer. RCORE to handle the power supply using Miro mini psu works perfectly. IEC socket have used a EMI/RFI filter built in for filtration. So its a clean and distortion free DAC playback I can hear.

The dac is being used in my second system which replaced my Soekris R2R 🙂. I am moving homes where unfortunately I do not have enough space for 2nd setup, so most of my second system is being disposed off. Slight scratch marks on the bottom plate because of usage otherwise rest of the chassis is pristine without any marks.

Asking price $576 including CONUS shipping and I will include a IEC power and a USB cable.

More information of this DAC here :
https://electrodac.blogspot.com/p/dac-ad1862-almost-tht-i2s-input-nos-r.html
https://www.diyaudio.com/community/...s-input-nos-r-2r.354078/page-209#post-7049224

Attachments

  • IMG_0001.jpg
    IMG_0001.jpg
    556.3 KB · Views: 354
  • IMG_0002.jpg
    IMG_0002.jpg
    468.1 KB · Views: 366
  • IMG_0003.jpg
    IMG_0003.jpg
    420.7 KB · Views: 316
  • IMG_0004.jpg
    IMG_0004.jpg
    438.3 KB · Views: 312
  • IMG_0005.jpg
    IMG_0005.jpg
    649.1 KB · Views: 281
  • IMG_0006.jpg
    IMG_0006.jpg
    452.2 KB · Views: 331

KENWOOD KT-5020L INTERMITTENT FAULT.

Hi All. Sadly, my KT-5020 tuner has developed an intermittent fault. It works perfectly but every so often, it has started to produce no audio at all. This presumably is because something in the "front-end" is failing as the display shows zero signal strength when the sound disappears. If I leave it switched on, it will suddenly work normally again after a few seconds of making a rustling sound in the output, then normal clear audio. Sometimes switching off after the fault appears occasionally gets it working again normally when switching back on. Could someone help me here please?. I could fix it but have no hands on with tuner fault finding at all. I attach the circuit diagram. Many thanks for reading. The file is safe to open.

Attachments

Building a Crossover for 450w RMS Woofers

Hey Diy Audio Crew!
This first post here, don't be to hard on me:')

in the planning stage of building a pa loudspeaker, including six 6" woofers, rated at 75w rms each. (Dayton Audio PA165-8 6)
wondering what sort of cross-over components won't blow at 450 wats. going to cross it over probably at 2500 hz or earlier.
also thinking of using the Pyramid TW44 Heavy Duty Titanium Super Tweeter with the included crossover. (posting that here in case people see a fault in my parts choice.)
just having struggles with finding cross-over components that won't blow. 🙂

all parts on parts express if you want to look it up.

Revox A77 Re-Cap Job

Hi All

I assume someone here has done this already. I already sourced most of the trimmers for this job. I have already tackled the power supply too.

I am looking to re-cap these boards:

Switch Board
Input Amplifier
Record Amplifier (2x)
Oscillator
Record Relay
Playback and Drive Amplifier (2x)

Gee, this cap list is looking evil right now. Can anyone share their experience (and maybe their list) for this?


Some of the electrolytic values are hard to find too. Some of the 3,3uF values could probably be replaced by Film (MKP) once I look at the schematics. I am tempted to try Film caps in what looks like coupling positions. Any reason not to do this?

My machine is a high speed IEC one. Sounds really good. Fixing the meters was a challenge, but they are once again good.

FaitalPro XL3000 series

At ISE 2025, FaitalPro released its new beast 21XL3000.

21XL.jpg


The 6,5" CCAW coil 3000W/6000W behemoth.
Normalyzed motor force of ~313 puts it in the high end class with all these new B&Cs, 18Sounds, RCFs and such.
Looking forward to reviews and use cases.

Now, birds on the trees are chirping that we might:
A) See 4Ohm version, because this will be really difficult to feed with mortal-grade amps.
B) See 18" version. That would be way too awesome. I praised FaitalPro crew to them myself and asked for tad longer voice coil (to add my voice to that idea if it already exists, no way they are going to listen just to ME 🙂 ).

If that is the case, I am considering to not follow up on my oncoming 18" purchases in the name of this.
The price seems to be somewhat higher, but still realistically competitive with quite some competition being more expensive already.

Anyone also interested and waiting to put it into some sick design?

Preliminary datasheet here:
https://faitalpro.com/highlights/2025/21XL3000/files/21XL3000 - Preliminary Datasheet.pdf

Should be available towards the summer.
  • Thank You
Reactions: GM

is Beyma TPL 150 sonically identical to TPL 200 ?

is it just two different lengths of same pleat or is the pleat different ?

because the 200 is only 33% longer yet claims 50% higher power handling so i wonder if it may also be wider and / or deeper in the pleat.

they also seem to have slightly different frequency response in the specs, but that may be due to different measurement conditions etc.

and yet the higher power handling of TPL200 is consistent with it having slightly more rolled off top half an octave if that power handling is achieved by having deeper pleat. deeper pleat would mean more conductor surface area and perhaps more power handling as a result.

in other words you would expect a 4" compression driver to have slightly more rolled off top end than a 3" so maybe Beyma felt you would also expect the bigger AMT to be a bit more rolled off and made it not just longer but also ... different ?

i understand even with the same pleat they would sound different due to different radiation patterns but if you were to array them those differences would mostly cancel out ...

so is it the same pleat or not ?

Professional hobbyist

Hello!
I have been building and designing loudspeakers from 2019.
Master's degree in Acoustics and Audio Technology.

I'm an entrepreneur, alltough my business is not (currently) in audio. (It's in building acoustics and noise)
I have my own dedicated and acoustically treated "Studio" where I listen to music while working 🙂

I have been reading this forum for a while, but finally have decided to start contributing with my own projects.

-Jesse
  • Like
Reactions: tmuikku

Tandberg 3000x Transformer diagnosis

Hi all
Just acquired a tandberg 3000x reel to reel player that I knew needed work but didn't check it Powers on, which it doesn't. I've checked fuses and cable into transformer for continuity, still need to check the switch but want to check the transistor as well before getting to capacitors and pcb work. My knowledge of such things is non existent but can someone tell me if this looks right before I give it a good probing?
Thanks all
1000037367.jpg

Back to my System - new to this forum

Hi,
Let me introduce myself.

I'm Revoicer (really John). I'm from Buckinghamshire (Bucks) in England/UK.

I have just gotten round to unearthing my old HiFi set-up. [Stored for approx 30 years.] And am looking to integrate bits with my more modern HiFi components.

I have joined this group to get some advice.
I am looking forward to receiving your pearls of wisdom to my mind taxing (or bored you senseless) questions.
Projects by fanatics, for fanatics
Get answers and advice for everyone wanting to learn the art of audio.
Join the Community
507,718
Members
7,885,677
Messages

Filter

Forum Statistics

Threads
406,114
Messages
7,885,677
Members
507,718
Latest member
Brano KAntor