Excursion below tuning point? JL Audio 6W3V3-4 Vented

I'm trying to design my first ported enclosure of a little 6.5" sub. It's a JL Audio 6W3v3-4.

I've used WinISD to model a 10L enclosure tuned to 32.03Hz with 100w input power, which is basically the parameters for a vented enclosure listed on the JL Audio website.

I'm a bit worried about the spike in excursion below the tuning frequency. Is there any thing I can do about this? Do I need to run a high pass filter to avoid damage?

JL 6W3v3-4 Vented Excursion.jpg

The load impedance of the push-pull structure electronic tube amplifier screen remains constant

If the load impedance of the secondary 8-ohm and primary 3500 ohm electronic tubes; The secondary 4 ohm primary does not have a 3500 ohm electronic tube screen load impedance; Can an impedance switch be installed in the primary coil to maintain a constant load impedance of the output tube.

It was translated using translation software. I hope you can understand! thank you

A new conceptual on wall line array proposed

So another friend/client has asked me to build him a set of on wall stereo mains for his new 86” OLED tv which is currently wall mounted. There will be no center channel so wide horizontal dispersion is critical for a phantom center to work.

I have 46” of height to work with. As wide an enclosure as needed but I’d like to keep it slim. Enclosure depth is limited to 5.5” to keep the profile the same as the screen offset from the wall. The enclosures will butt up right on the outer edges of the screen so in effect, the screen will be a continuation of the baffle.

My first thought was to use a line of 8 ribbon tweeters but that’s the most costly solution. My second thought was a line of 24 3/4” dome tweeters…..specifically the Dayton ND20FB. Here’s Zaph’s measurement page

http://zaphaudio.com/tweetermishmash/compare.html

So we can see excellent frequency response and extension as well as great off axis response due to the small size. I can space these less than 2” center to center and work within my 46” limit. I can also get the center to center spacing close to whatever midwoofer I choose so crossing these at 3.5k won’t be an issue at all…..and with 24 of them, I don’t see a problem with HD either…..

……but I wonder what happens to the response of the driver vertically…..given the center to center spacing of around 1.75” inches or 45mm, I’m expecting a rolloff to begin at around 8000hz from the driver to driver wavefront interference….am I correct in this or is it at 1/2 wavelength?……16k? Also what would be the expected slope of the rolloff?……1st order?….2nd order?

Thanks in advance. Midwoofer selections and considerations will come after a tweeter solution only given the need for phantom center coverage is the limiting design parameter.

Speaker switcher

Hi all,

First of all, I apologize for potentially very long post, but I think this is an interesting topic and hopefully worth spending time reading it.

I am restoring vintage audio equipment, mostly receivers, and I already have small collection of different receivers and amplifiers that I keep for myself. The problem is that whenever I want to use one of the them I need to switch speakers cable which is very annoying.

I managed to find some kind of solution, the mechanical switch which supports using two different audio amplifier with single set of speakers, or other way around, using two set of speakers on single audio amplifier. Here is the image of that:

1730383985991.png


As this is mechanical connection only I was able to connect two different receivers to the same set of speakers and listen to differences in sound when tuned to the same radio station. Also, I use the same approach to test new set of speakers against the old one with the single receiver. It is really useful tool but it has few major limitations:
  • It is limited to two devices only
  • It is not cheap (I paid mine ~80EUR)
  • It still needs physical force to switch, etc.

I decided to start building a better solution which will work for more than 2 devices with hopefully still acceptable price. The main goal that I would like to achieve is be able to connect multiple receivers into single set of speakers, but it also can be used for similar things.

Speaker Switcher

So I started to design my own solution. It is in very early phase, but I would like to share it anyway to get some feedback and maybe some ideas for improvements. I don't have any business on my mind, I will make it open source and hopefully other people can benefit from it.
Here are some requirements that I have on my mind
1. Modular approach. The idea is to be able to upgrade it as you grow in order to lower the initial costs.
2. Based on existing components. The parts should be available on most popular shops (DigiKey, Mouser, etc.)
3. Have mechanical connections for speakers. So nothing that can affect the sound.

The initial solution that I come with is consist of three parts.

Speaker switcher​

This part is responsible for real switching using relays. The circuit is pretty simple. There are two relays controller by transistors:

1730385473622.png


On the image above is shown one switching "module". Left connector is where we connect output of the receiver, and the right connector goes to the third module that is connected with the speakers. The 3 pin connector above provides 12V for relay and signal for switch it on. The control signal is provided by control module.

Here is the schematic:

1730385770034.png


Control module​

Control module is used for switching between inputs and also has the controlling logic. Controller is Atmel 328p, which is very popular controller used in most of the Arduino boards. Here is the initial design:

1730385947427.png


Schematic:

1730385980832.png


The idea is that there are separate "push" type switch for each input. Atmel (Arduino) will monitor which switch is pressed and will try to activate appropriate module. It does it through de-multiplexer 74HC238 which makes sure that only one of the modules can be active at the same time.
It also have 12V power supply (needed for relays) and 7805 for converting to 5V required by logic chips. I am also thinking to find some 5V relays to reduce components and need for two voltages.

Speaker board​

Speaker board is just physical connections between all outputs from all modules:

1730386324809.png


As the initial post is already very big, I will not put all the details here, but in some other comments. For the end, here is the list of the things that could be helpful:

1. Recommendation for relays. Maybe you know some relays when can switch 4 circuits at the same time, work on 5V, etc...
2. In current design Atmel328p is used directly in the board. Maybe the whole Arduino mini or something can be used instead which will make programming easier.
3. Any suggestions about improving circuits, any obvious electronics mistakes., etc.?
4. Should it be a single board or multiple boards like it is now.
5. Any suggestion for case?
6. Really any feedback is welcome!!!

Looking forward to get some input from you!!!

ADCOM GFA-7000 - High DC offset

As the title reads, I bought a ADCOM GFA-7000 with two faulty channels (Right and Center). I have repaired them by replacing R25A (100k) and R27/R35 (49.9k) with uprated 250mW parts. I also replaced the 4 electrolytic caps when a couple tested poorly and one was leaking a little.
But when I power up the Right amp channel, there is about (500)-(800)mV on start-up, then immediately drops down to ~(100)mV and then after a couple minutes slowly comes down to ~(33)mV. It stays at (33)mV with no change, I've set the bias (15mV) and no change to DC stays at (33)mV.
Other channels power on at ~50mV, and immediately drop to ~20mV and settle at <5mV - so something's not quite right, on the Right (channel)

I have checked resistors around the input pair transistors - R14 (1.0k), R15 (33.2r), R16 (33.2r), R17 (392r) as well as R20 (1.0k), R21 (33.2r), R22 (33.2r), R23 (392r)

One anomaly, likely just me not understanding the annotation on schematics:
Both schematics I have located, show R17 and R23 as 392, which seems to be value code for a 3.9k resistor. However, checking resistors on the this channel and other channels, and the resistor color bands confirm 392r, with Orange - White - Red - Black - Brown (392r 1% color band code)

Would there be any other culprit of a high, unstable DC offset outside of the NPN, 2SC3381 (BL) and PNP, 2SA1349 (BL) dual BJT package transistors? Would the DC being negative point more towards the NPN transistor being faulty?

I know replacement transistors would be hard to find (originals anyways), and may have to resort to matching a quad of TO-92s

Would any of these be a satisfactory replacement, since I have them on hand?
  • BC547/BC557 (CBE pinout, 45Vce & Hfe 450-900)
  • BC550/BC560 (CBE pinout, 45Vce & Hfe 200-450)
  • KSC1845/KSA992 (ECB pinout, 120Vce & Hfe 300-600) --- I think this ECB pinout might be easier, but both can be made to work without too many pinout contortions

Thanks for the assist. Attached is the datasheet and two schematics I have been working with, also used a GFA-7500 schematic, which matches up closely too.

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Pi based streamer expectations on sound quality

I use an Arcam Movie 5.1 as a preamp/cd player into some Linn Room Amps, biamped, and biwired into some Monitor Audio Monitor 3 speakers that I've recapped and rewired, stuffed etc etc, and for what is pennies in HiFi world, I have what to me seems like a very good system, certainly the best I;ve ever owned, but TBH I haven't auditioned anything in a proper HiFi shop for a long long time. Anyway, liking the sound that the setup creates using a CD, I began to wonder what a streamer would sound like - CDs take up a lot of room (!) - so I bought a second hand RPi 5, and a Digi+ Pro (clone?) off Amazon. I didn't expect much, and was fully expecting to have to send it all back, but for a source that I've put no effort whatsoever into making sound good, it's to my ears almost exactly the same as using the CD - damn !!

So, the next stage up would appear to be £120 ish for an IanCanada Transport Digi HAT - but just how much better would anyone say the sound would be, if at all ?

Behringer NX6000 vs 2x EP4000 bridged

So i'm starting a dual subwoofer build - small enclosure, lots of power.
Since NX6000 can't be bridged dual doesn't make sense to go dual. Supposedly it outputs around 2,1kW per channel a 4ohms.

EP4000 is a QSC clone and can supposedly output 4000W bridged, or 2400W @1% THD (4ohm)

also, EP4000 is Class AB and NX6000 is ClassD.

I've seen various FP14000 or FP10000 clones - but including import and taxes it will cost me 40% more than dual EP4000, although they'd probably be great.

Also, it's going to be a studio build, so stable power is desired.

(there's also a tiny bit more expensive thomann t.amp TSA-4000 Class-D, which runs 2865W bridged but only at 8ohm - makes me wonder how it pairs against the cheaper EP4000)

plans for manifold 2x15 or D&B B2 style

hello everyone, I need a design for a nice powerful double 15 sub, I was thinking of a manifold or band pass hybrid type b2 design but with 15" but I can't find anything, can you help me?
I listened to the 18 nanifold eighteen sound project and a copy of b2 With 18nbx100 and they sounded like I like, I just need them to go down a few db less 🙂
currently I used the woofers on a bandpass design similar to fbt subline / maxx10sa and on tham 15. the bandpass design seemed stronger to me at outdoor parties using 15hp1060 3 15nbx100 cut at 100hz, but I would like something more!

there are these on youtube but will they work?
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thanks

EC AW-120 Dead after relay replacment

Hi on the Solid State forum!

After replacing the old relays (type GOOD SKY RE-404DP) with some new Schrack PT581048 on my old Electrocompaniet AW-120 power amp the protection circuit is triggered.
I only have the schematics on the power-amp modules and are unable to find the layout of the main circuit board with the Toshiba TA 7317P (IC2 i picture att.) based IC protection circuits.

I would be most greatful to receive a copy of the service manual for this "vintage" amp or other schematics including the AW-120 main board.

Any other hint's and advice on possible adjustemnt of the protection circuit are all welcome !

Best regards from
Arne , at Løvenstad - Norway


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Introduction

Hello everyone! I’m Marqus, and I’m excited to join this forum.

I love tinkering with gear, doing mods and just seeing how I can make things sound “better” (or at least different).

I think it’s really fun experimenting and listening how different tweaks and mods can shape the sound, but in the end it’s all about the music.

I’m always looking to learn from others, share some ideas, and swap tips on projects.

Looking forward to connecting and talking with everyone here!

Best TPA 3255 board yet?

Got theese babies because of the rave review from the YouTuber HiFi Zone who has reviewed many amps of all types.
His where the ready made ones that are fed by an external supply.
These boards here are identical except for the bridge rectifier on them.
Even in stock form they sound better than my Ayima 3255 that has the OPA1656 and other mods that where suggested in the Ayima thread. I will see where some discreete opamps and some other needed changes will get me.
They are configured as SE but one bridge will make them fully balanced, which I always found to be markedly better on all these TPA and Merus boards.
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Is inductor power dissipation important?

I am designing a crossover for a midtop speaker with a 12 inch woofer that can handle 250 watts rms. When I check the power dissipation graph in xsim one of the inductors shows that it will dissipate up to 8 watts. Is this relevant at all? I'm pretty sure this is something to look out for with resistors but I'm not sure if this is important for inductors but it surprised me how much more power that inductor dissipated than the other components.

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ESL Newbie questions...

Hi all,

me and a friend are going to build ESL after building "normal" speakers for centuries.
So far we've read a lot about ESL's. The more we read, the more questions we have... I gues you know what I mean 🙂
Anyway, we are going to build hybrid speakers, me i'm going to use a Ripol (since i am a big Open-Baffle fan) for the bass section, my buddy will use a closed box.
We decided to cross them around 200-250 Hz. If passive or active is not sure at the moment...
The panel size will be 285x900mm divided in 20-12-6-6-6-12-20 segments.
And here the questions start...
If we use aluminium profiles for the frame and also for the cross-braces and use duoble-sided tape 0,2mm for bonding the membrane to the frame, will there be enough insulation against the membrane and also the wires?
In the picture you can see that we tried two different sizes of ds-tape, one is 0,2 mm, the other one is 1.5mm thick. This was to check wether the braces will stick strong enough to the frame, unfortunately they do not. Next step will be to check wether glueing the braces with Sikaflex221 to the frame.
If we use rivets to fix the braces it is ok, but we lose the insulating effect of the frame material, this is a sandwich of aluminium and some plastic (here in Germany it is called Alucobond).
Don't mind the stator wires on the pictures, this was only to check how fast the glue is drying.

Another question concerns the distance from stator to membrane. Since the material we want to use for the frame was only available in 4mm and we want to keep things simple, the distance/gap between stator wires (outside of the insulation) is only 2mm instead of 3mm like described on the website of Steen Frank Jensen.
To be honest, to keep things simple we are going to copy his model since it is not that big.
We also like all the other stuff we have seen here in this forum but most these ESL's are too big.
Anyway we include some ideas from here in our building process...

So we are looking forward to your answers.
We are sorry for the chaos with the pictures... We couldn't find a way to delete the needless ones...

Best regards, Juergen and Dani
from Germany
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Tl494 Gate drive circuit design

Hello, i am trying to design a gate drive circuit with tl494, switching a gate drive transformer, which will switch two high voltage transistors on a half bridge power supply.
IMG_1096.jpeg

This is what i drew up then built, the switching transistors q1 and q2 are 2sc2655, with 33v 1n4752a diodes across them (hard to make out, i know)
At the moment i am trying to work out if its working properly. Its drawing about 0.2A of current at 24v from J1, which to me seems a little low for a gate drive transformer circuit…
This is the outputs of the gate drive transformer (secondaries) i measured.
IMG_1092.jpeg

IMG_1093.jpeg

Doesn't really look right, and im not sure what to think of the vpp.

Can anyone see anything wrong with my circuit , or that could be improved upon?

Using PSUD to optimise power supply in my KT120 amplifier

Hi folks! I acquired an old Luxkit A3600 push-pull amplifier a couple of years ago and I've enjoyed getting to know how it works, modifying it to take KT120 output tubes (from KT88s when I got it) and 6GU7s (in place of 6240G tubes that are long out of production). I've modelled the power supply in Duncan amp's PSUD2 application and wondered if anyone had advice about the goldilocks level of damping.

The original power supply consists of:
  1. a low DCR secondary winding labelled as providing 370V with max current of 400mA into
  2. a full wave bridge diode rectifier
  3. two multi-section caps (3x 47uf) split into:
    • 2x47uf then a 350mH (11ohm) choke coil
    • 2x47uf (provides B+ for KT120 output tubes) then a 2.2Kohm resistor feeds LTP phase inverter stage (twin triode 6GU7 tubes)
    • 1x 47uf then a 100kohm resistor feeds input signal preamp stage (twin triode 6AQ8 tube)
    • 1x 47uf then a 240Kohm bleeding resistor
I modelled this arrangement and tweaked the value of the last resistor, the capacitance values and the voltage at the secondary winding so they matched my voltage readings at each stage. I estimated the current at the output stage (4x 75mA fixed bias plus 4x 7.5mA for UL screen current) and at the phase inverter stage (2x 9.6mA constant current from shared resistor). I've estimated the inductance of the choke coil at 400mH because the current draw is 50mA below the 400mA current the choke measures at 350mH.

This is what I came up with:

1721735964870.png


I wasn't sure what to make of this so I did a bit of searching on the internet about optimising PSU components (shout out to DHTRob and many others) and created a new PSU model to make it faster and remove the overshoot and oscillation (see below).


1721736659380.png


Note that I increased the first resistor (R1) from 1Kohm to 2.2Kohm because I have recently moved house and I'm now experiencing a higher wall voltage. I can see from my PSUD experimentation that spreading out the capacitance across the amplification stages slows the power supply down, while front loading capacitance reduces ripple experienced by the rest of the amplification stages. The total capacitance in the stock configuration is 332uf and 451uf in the second model. The residual ripple for the output stage (I1) is less than half the stock config (1.3V vs 560mV diff min to max) and about the same for the phase inverter (I2) and input (R3) stages.

Do you think it is safe to try this configuration or should I reduce the capacitance of C2 and increase the capacitance of C3 and C4 to increase the damping and reduce ripple at those stages? Are there any stability concerns and what difference might I hear (if any)?

Endevco Signal Conditioner 133 frequency response

It seems, that I'm having an issue with my Endevco 133.

If I measure the frequency response w/o any sensor attached, it has a big bump between 40Hz and 60Hz. Other than that it seems to work OK.
As I bought it used from eBay some time ago, I can't say anything about the conditions it has been used in or about it's age.

Ideas/ advice would be appreciated.

Best
Stephan

Analog signal to HDMI input

Hi!
I am searching for possibility to organize analog input for a DSP with HDMI input only.
I am using miniDSP HDA (2x HDMI inputs, 1x HDMI output, 8x analog outputs) for my multi-way DSP based speaker system. Normally I am streaming simply from my computer. HDMI output is connected to a beamer, but is not always used, I can simply listen to music without a video.
Now I need an analog input for this system. Sound quality is not a real concern in my case, and I need only stereo. I have seen of course simple AV2HDMI or VGA2HDMI converters, but will they work without video input, maybe someone has expirience?
Or maybe you can help me with a way to do it?
Thanks!
Andriy.

HDMI/toslink multichannel 7.1/5.1 to usb audio out

I have various USB soundcards (7.1 channels) that I'm trying to feed with multichannel audio from my TV and console via HDMI or as a fallback Dolby Digital via toslink.

I was thinking of using my old raspberry pi 2B with some expansion board for HDMI/optical audio extraction or USB hdmi input device, if those exist. I saw some boards like hifiberry digi i/o and x6000 but they seem to only provide 2 channel input.

Does anyone know of a hat/shield/board for pi that could work or some USB device that could give me 8 channel input? I'm trying to build something that is not too expensive, shouldn't crack the 200usd bar 😉

Trying new LTSpice 24 - problems with .lib directive

Downloaded new LTPsice 24 today. I've been using LTSpice XVII).
In my sim file, have statement to load lib file from a URL:

.lib http://www.minek.com/lib/minek.lib

and it works perfectly fine with LTSpice VXII.

Of course it doesn't work in LTSpice 24. I'm getting error "Could not open library http://www.minek.com/lib/minek.lib".
Any ideas how to make this work?

Attachments

Introductions

Hi,
My name is Liam, I live in England. I work as music technician at a college. I do very basic repairs on instruments and audio equipment. I also have a music studio that I wired in but I’m trying to get into electronics so I can service my mixing desk and other audio gear in my studio. I’m looking for advice on equipment I’m trying to repair and here to gain general knowledge around troubleshooting equipment.

JLH 300 Series hiss

I recently come way from a charity shop with a pile of equipment which was only for sale as a job lot, after I spotted an interesting looking amplifier at the back.
It turned out to be a Hart Linsley Hood kit - Stereo 300 Series - and came with a pile of documentation too. It sounds fantastic, very powerful and controlled, but has what seems to me to be slightly excessive hiss. This is not affected by the volume control and is the same for all inputs, connected or not, shorted or open. It varies with the treble control, and doesn't affect the overall sound when playing music, but just seems a bit intrusive when just idling.

I was wondering if any experts could see anything in the design which would cause this; could it be ageing components or is it something I just have to live with?
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Recommended Books for starting out with Audio Electronics

Hi everyone,

I am starting out learning about electronics with a view to buy and restore non-working vintage audio amplifiers. I am looking for some good book recommendations to help me get started. I don't have a background in electronics or engineering, so some books that start at the beginning and walk through to more complex things would be great. Any thoughts? Be very interested to hear what people have found helpful.

Wiring an Alps BX2 balance pot

I'd like to replace the dual mono volume pots in my HV V2 with a stereo volume pot plus a balance pot. I know there are varying opinions on using balance pots; I'm not interested in that debate... Let's just say there are WAF considerations...

Anyway, I have the Alps BX2 balance pot pictured here. It seems to be uncommon, and I can't find any info on how to wire it in a preamp. The pot has 4 pins per channel, and I've included a schematic. I'm just having a hard time wrapping my head around how to wire it to attenuate one channel while leaving the other channel unattenuated. Any suggestions?

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Introduction

Hello all!

My name is Pierre and i live in a little village outside of Gothenburg, Sweden, called Öxabäck. I am new to DIY audio equipment as I have always only bought it before and not solder it myself. In the recent year I have developed a taste to understand the underlaying function of the different parts needed to make an audio board work. The first one I made was the ProtoDAC TDA1387 x 8 that Moode have on their site. A fun little board that sound really good.

A few itterations later on that board I am now trying to learn I2S to use SPDIF/Toslink as input matched with WM8804 on a Raspberry Pi with my ProtoDAC as audio out.
Still got a long road ahead with a lot of reading and learning!

I hope I can get some help here when I get stuck in my studies 🙂!

Karlson-Cabinet 10" build for E-Bass-Guitar?

Hello!

My name is Flo. I am new to the Forum - and new to the whole topic of Speakers/Cabinets... (total NEWBIE)
I am putting this thread here because i read most of the Karlson-related posts here in the fullrange forum... hope im right.

I play electric bass guitar and want to build a speaker cabinet in the style of the Karlson/Karlsonator.
I do have skills in carpentry but not so much in elektrotechnical or acoustic/physics stuff :/
I hope you guys can help me out realizing my idea!

I already have a 10" speaker for my project - OBERTON 10B200
These are the specs/TSP:

https://oberton.com/en/?option=com_content&view=article&id=180:10b200&catid=49:ferrite-loudspeakers

Nominal Diameter: 10"/262 inch/mm
Impedance: 8 Ohm
Minimum Impedance: 5.25 Ohm
Power Capacity AES ¹: 200 W
Power Capacity ²: 150 W
Program Power ³: 400 W
Sensitivity: (200-2000 Hz) 93 dB/W/m
Frequency Range: 45 - 3000 Hz
Voice Coil Diameter: 51 mm
Voice Coil Material: Copper
Voice Coil Former: Kapton™
Voice Coil Winding Depth: 17 mm
Magnet Gap Depth: 7 mm
Cone Material: Glass Fiber
Basket: Die cast aluminiumMagnet: Feritte
Flux Density : 1.35 T

Resonance Frequency: 47.88 Hz
Mechanical Efficiency Factor (Qms): 9.27
Electrical Efficiency Factor (Qes): 0.341
Total Q (Qts): 0.329
Equivalent Air Volume (Vas ): 36.21 Litres
Diaphragm mass ind. airload (Mms): 43.75 grams
Voice Coil Resistance Re: 5.10 Ohms
Effective Diagram Area (Sd): 317.3 cm2
Peak Linear Displacement of Diaphragm (Xmax)*: ± 5.75 mm
Mechanical Compliance of Suspension (Cms): 0.252 mm/N
BL Product (BL): 14.02 T.m
V.C. Inductance at 1 kHz (Le): 0.84 mH

*First question would be if it is possible to put that speaker into a Karlson-Style cabinet and can it be tuned around 60Hz? (at least, this is what i was told to be "best" for bass guitar (when vented??)).
*Secondly i want to stack another, existing box of mine onto the enclosure (or/and other way around) - meaning that depth should ideally be 325mm(12.8Inch) and width 328mm(12.9Inch). Is it possible to just adapt the height to get to an useful result?
*Third - i want to "integrate" my existing amp into a recessed area on the back of the housing/cabinet. Here is a picture of my Ibanez"Promethean"-Bass combo...
i want to do it in the same way.
Promethean.jpgPromethean2.jpg

I really love the "looks/design" of the Karlsonator with its curvy acoustic lense/aperture and i was already looking at some fascinating stuff you guys are building here.
Also i found this guy @freddi on youtube who plays an electric bass through his selfbuilt xki-Karlson:
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Could someone please help me out to calculate/design the desired cabinet/box or point me in the right direction?

Greetings from Vienna/Europe
Flo

Klippel data interpretation, B&C's Xvar info value

What is the possible bleed of Bl(x) and Kms(x) values between each other with Klippel testing?

Looking at B&C Speakers 18DS115 with long coil and 14mm Xmax, things apparently look different than they are. The voice coil doesn't leave the gap till past 13mm travel one way, yet Bl falls rapidly in the graphs.
Also, at Xvar of 14mm, this beast is run past 20mm in their 2hr test, making it more IPAL-ish than conventional? What could we take from that? Could I use it safely past Xmax for short/peak use?

Introduction

I am not much on introductions, but my name is Craig. I work on electronics as a hobby and make noise with my guitar (probably doesn't qualify as music). I have worked on tube radios since I was a kid (1970's), these days several decades later I have a decent collection. Some of those date back to the early 30's. I am also an amateur historian, amateur radio operator... and for fun I am the mayor of the small town I am in. I get bored easily.

My guitar playing is similar to my radio repair hobby (actually I am much better at repairing vintage electronics). I have beat on a guitar since I was a kid, never mastered it, but I do it as a hobby, and as a result, I have a few old amplifiers (and some newer ones). Some of those amplifiers are of course tube type. I am about to rebuild my grandfathers old Silvertone 1472 that hasn't been used since his death in the 70's. I am also working on my first amp that quit and was placed in a closet.

I figured I would come here to see if I could learn anything and maybe offer something in return!

Using Ian Canada's modules with Audiophonics EVO-SABRE DAC Kit to the next level

My search started with wanting to integrate an ESS ES9038Pro board (see link below) to an RPI4 via the GPIO (I2S) but power the DAC with a good linear power supply instead directly off the GPIO.

ESS ES9038PRO DAC Module I2S XLR 32bit 384khz DSD with Screen - Audiophonics

When that came up empty I thought about this kit (see link)
AUDIOPHONICS EVO-SABRE DAC Kit DIY DAC 2xES9038Q2M Fully Balanced & Streamer for Raspberry Pi 4 - Audiophonics

However, I felt that they have shortchanged the project by not using a really good power supply for the DAC and feeding it off the GPIO. I also wondered about the clock. So I dug further and I ended up on 3 very long threads from Ian Canada on DIYaudio. (see links)

ES9018K2M, ES9028Q2M, 9038Q2M DSD/I2S DAC HATs for Raspberry Pi

Asynchronous I2S FIFO project, an ultimate weapon to fight the jitter

Develop ultra capacitor power supply and LiFePO4 battery power supply

The Audiophonics 9038Q2M kit above looks nice and slim but is only 300mm wide and 185mm deep and is not a full-width component (440mm wide) or deep (400mm deep) so I figured if one has a case that is 440x400mm instead of 300x185mm could one upgrade the kit using Ian Canada's modules listed below.

  1. FIFOPI Q3 ULTIMATE FIFO Reclocker
  2. LINEARPI DUAL Ultra-Low-Noise Linear Power Supply
And maybe a SheildPi. (is there a link for this)

To keep the height to 55mm I would use a 40pin cable instead of stacking the boards.

A second option would be to use all parts from Ian Canada but I am not sure what this list of parts would be. He has a long list of modules and options.

Maybe
  1. FIFOPI Q3 ULTIMATE FIFO Reclocker
  2. LINEARPI DUAL Ultra-Low-Noise Linear Power Supply
  3. The elusive ShieldPi (I can't find a link for this board)
  4. Dual Mono ES9038Q2M DAC HAT
  5. ESS SABRE 90X8 controller with display

and maybe
TRANSPORTPI Digital Interface SPDIF Low Jitter
RECEIVERPI Digital Interface I2S SPDIF

Can anyone point me in the right direction? Thanks.

Powerful SMPS for tube ampifier

Hello, I designed a SMPS that creates 0 audiable noise in a tube ampifier. Originally this was a school project for a competition but things did not work out quite the way.

The powersupply steps up 10-30V input to whatever output voltage you desire up to the volate rating of the filter capacitor (1kV best estimate of the highest voltage). I have tested the powersupply on a resistive load and I managed to test at 450V 50W without any issues. The tube amplifier I tested the supply on is a well known 20W Mullard amplifier. The amp consists of a pair of EL34s a 12AX7 inverter and driver and a EF86 as the preamplifier. The onl change I did was to use a EF0 instead of the 86, and even it being a very microphonic tube does not create any issues whatsoever and it sound amazing. I designed the amplifier on a PCB and the powersupply too.

Right now the powersupply its been about the third prototipe, has a small issue so without a fan blowind lightly on the switching transformer an the heatsink of the IGBT, gets unbareably hot and the input filter cap too so I guess it acts as some sort of weird oscilator - I havent looked into it any deeper yet I was more interested to hear the amp.

What I would like to know, if there is anyone even lightly interested to see a switching powersupply for a tube amplifier. The current version creates the 410V for the amplifier from anywhere between 12 to 30V. It is also possible to use a different switching tranformer for rectified line votages- why would I use a swtiching powersupply istead of a regular simple transformer on the line voltage? Lets say the voltage if tightly regulated and stable-and this powersuppply is MUCH cheaper than the transformer itself.

Little about myself- Im a students I turned 17 last week and I was protiping this SMPS for about 6 months before i I got to a point of any real useable result. I made lots of amplifiers mostly transistor based or IC based AB amplifiers and Im also into analog technology. So basicaly a nerd 😀

I apologise for the grammar im not a native speaking.

If anyone is interested in any measurements I can do them I do have a digital scope and other testgear to do almost any mesurements I can think of except EMI and interference measurements

Illustrated guide to the 4U deluxe Chassis

Illustrated guide to the 4U deluxe Chassis



This is a visual guide to the assembly of the diyAudio 4U Deluxe Chassis. The “Jack of all Chassis”.

It has a pre-drilled back plate and pre-drilled and tapped heatsinks. The heatsink drill pattern is the diyAudio “Universal Mounting Specification” and the back plate has holes for input RCA jacks, Speaker jacks and a Power Entry Module that contains IEC mains inlet, power switch and fuse holder.

This chassis is ideal for the HoneyBadger, F4, F5, Aleph J, or Burning Amp. (The BA- with just one PCB of output transistors/channel)
It could also be used as the chassis for a Monoblock F5Turbo v2 or a Burning Amp with 2 PCB of output transistors.

Lastly this chassis has the “DIY-friendly baseplate” a sub-floor that has a grid of 10x10mm holes making no or little need for drilling holes to attach most of your components to the chassis. You will find that with just a little planning you will commonly need only to widen one hole to accept the mounting bolt for a toroidal transformer. Everything else can be mounted with the existing holes.

The chassis is also compatibletable with the diyAudio Chassis Riser system, allowing for more area to mount whatever your project may require.

Much more information may be found here -

Deluxe 4U "Jack of all Chassis" (All Aluminum) - Full width with 40mm Heatsinks - Chassis

Let’s look at the contents of the package -

IMG_2190.jpg

Feet and hardware.

IMG_2194.jpg

Heatsinks.


Top and bottom plate. These are custom aluminum, with an extra row of vents per side.

IMG_2192.jpg

Back panel.

IMG_2195.jpg

10mm custom front panel. Obverse.

IMG_2196.jpg

Front panel reverse. This is a custom part made for diyAudio with the array of holes shown - these are all blind, but most are tapped. For example you can attach the V3 PSU circuit board vertically to the front panel, or attach a riser plate. The bottom centered offset hole can be drilled out for a power indicator LED. Other holes are there to accommodate handles. I’m sure there are many ways to utilize these attach points that the community will discover.


These rails are the mechanical heart of the chassis. They attach to the heatsinks, and then everything attaches into them.


DIY-friendly perforated baseplate.


Everything.


The logical place to start is here, with the rails and heatsinks.
(sometimes there is streaking from the cleaning process. It makes no effect on the efficiency of the heatsink, and is quite smooth.)


Bolt the rails to the heatsinks, but leave the screws a little loose.


The next pieces will be the back and front panels.


Locate the back panel and this hardware.


Attach both sides as shown.




You should have this assembly when done.


Align the front panel and your assembly as shown.


Attach here with this hardware.


Looking good!


Next the perforated base needs to be installed. This is the hardware provided.

IMG_2224.jpg

It attaches into these countersunk holes in the bracket.


Using the supplied hardware it is intended to attach with the lip up


Which will give enough clearance in-between the black aluminum baseplate at the perforated base to clear small screwheads.
I like to mount it with the lip down, giving enough clearance to clear the through-bolt of a toroid transformer, but it requires some alternate fasteners. (See below for more information.)


Perforated base installed.
Now is a good time to snug all the screws and square the chassis as best you can.


The top and bottom panels are identical, but there is a front and back - but as the holes will align in only one direction, it is easy to determine.

IMG_2230.jpg

The top and bottom panels attach with the sheet metal screws provided. It is a much stronger interface that you would think, regardless the strength is all in the brackets.

IMG_2231.jpg


IMG_2232.jpg


IMG_2236.jpg

Start of multi-way project. Help with choosing

Good morning everyone.
I have just finished my no compromise subwoofer project composed of Faital pro 18XL 1800 thanks to your precious advice. So now I can do nothing but finish the job by also building the speakers.
I have a lot of work to do. At the moment, as a system, I have a Sabre Topping D90 III dac, a Topping Pre90 pre, an Audiophonics ET400A Purifi power amplifier and a Powersoft power amplifier with Loto DSP and 2 Litemod 2ch modules.
currently the towers (PL200 audio monitors) are moved by the ET400A purifier while the subwoofer by a single BTL powersoft module.
I could use 2 more channels to harmonize the subwoofer with the woofers in the project I will realize.
The project will be in pneumatic suspension. Speakers divided into medium-high and low sections. Woofers (2 per speaker) separated from the medium-high range. I am looking for high efficiency, precise, fast, dynamic components. In short, the best you can have while remaining within human figures. I am not very expert in transducers, so my research was very limited.
For the low end of the project I had considered the FaitalPRO 10FH520 16Ω x4 (16 ohm because they will be connected in parallel, but also because compared to the 8 ohm version these can be put in pneumatic suspension, while the 8 ohm ones unfortunately cannot).
Midrange FaitalPRO 6PR160 8Ω d'appolito configuration x4.
and for the high end the Beyma TPL 200.
The main doubts concern the integration of the Faital midrange with the Beyma AMT. From its MMS of 12.5gr (FaitalPRO) it seems that it can keep up with the Beyma.
But as I said before I am not sure because I am ignorant on the subject.
I chose the rest of the components left in Faital for a familiar sound problem, but I have no problem changing brands if there are better drivers that can work with high efficiency air suspensions.
I had thought of putting a wide band as the medium, limiting the band to the medium one. the fostex fe 168 ez sigma drivers. But I didn't know if it was a good idea.
I also have to choose whether to make a passive crossover filter or buy an active one. They say that the passive one is better.
The type of speaker must be a tower, trying not to exceed 1.30 meters in height. and I would like to stay flush with the transducers in width. While I can have margin in depth.

CEC TL3 - TOC recognized but sometimes lengthy table of contents, depending on the original CD

Hello, I have a relative who entrusted me with a CEC TL3 because the player was having problems recognizing the table of contents with two of these Cds, which are originals (not CD-Rs). After dismantling the unit, I found three cut chemical capacitors (330µf/6.3v) on the power supply board. I also took the opportunity to replace the three 47µf/50v close to the heatsinks. I didn't find any other chemical capacitors out of order on the power supply board. I should point out that the laser lens was cleaned in two passes (isopropyl alcohol). After reassembly and tests with several Cds, including the one that was not recognized, there is some improvement, but it's still not optimal. I'm thinking of checking the servo card settings and readjusting only if necessary. For this, I'll need the CEC TL3 manual service. The servo card has two markings on the PCB (TK-T71 94V-0) and (4B10012700C). I'll post a photo soon. Thanks in advance to anyone who can give me a hand.

power supply base on Pass Labs X2

I want to build the regulate power supply base on Passlabs X2's power supply.
The original on use two transformer to form a network of four 40volt rails attached in series and feed to the regulator as show in pic#1
For mine, I wanna use only one transformer. So, may I use the schematic as shown in pic#2. the 2 seconday form 40volt and 120volt rails to feed the regulator to get the same regulated voltage.

My quesion :
  • Is there any difference between two of this in term of performance? apart of using higher voltage cap.
  • Can I use IRF610 mosfet instead of IRFP240 for the plus voltage rail? since the the current draw from the psu is about 60mA. Is there a special reason why IRFP240 is picked for this job?

Thank you.


org.png


sim.png

Mission Cyrus Integrated Amplifier Series - various DC Sockets/Plugs for PSX - Overview wanted

According the attached images there are in use different DC connector versions for the outdoor power supply - not only dependent of the model and the age.
4-pin DIN- and XLR versions I note, so as a blue version from Hirose.

Who can upload the whole overview of all used brands and types ?
Thank you very much.

P.S.: The blue socket on PCB from third and fourth image seems to be a HRS (Hirose) RPC2
https://in.element14.com/hirose-hrs/rpc2-12rb-3p-71/plug-chassis-3pole/dp/1077726
https://in.element14.com/c/connecto...IkPXrOiQPIqa71CYuHfX_wVg3NAgSYUHf8Fm-b6ifV3VY
https://uk.farnell.com/hirose-hrs/r...lMXJkXeNffff-UbqayHKFkQLoXvmtmmmDoFJndGNxKsUf
Unfortunately no longer available

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Introduction

Hi, I’m Phil. I have been into HiFi since about 13yo (1970’s). My uncle was a TV and Audio engineer. He was semi retired and working from a home workshop that seemed like an Aladdin’s cave! He gave me an old valve radio that I set up at home with a long ariel. I think the warm sound, the heat, and the glow of the valves got me hooked. I’m retired now myself and hope to learn enough about electronics to fix up some equipment, and maybe build some kits.

MB112-II purple light

Good evening good people,

I just bought a secondhand GK mb 112 ii. It had the pink/purple led (instead of the normal blue) on the power button, which indicates a fault, but it still worked fine and sounded good, so I took a gamble on it (I got it for around 150$). I'm trying to troubleshoot the amp and hopefully make it as good as new. So far what I know:

- The amp's manual states : "Smart Protection Circuitry: The MB Series protection circuitry constantly monitors for unsafe operating conditions such as short circuits, miswired speaker cables, blown speakers, improper ventilation, and incorrect speaker configurations. In addition, protection is provided during power up/down. If the MB Series senses any unsafe operating conditions, the output signal is immediately muted and the power light changes from Blue to Red. The amplifier will remain muted until the fault is removed. Be sure that you are not exceeding the maximum recommend load described in this manual. If the problem persists, contact GK directly for technical advice."

No info about purple light, or if it can still work if the protection circuit detects a fault.

- When I plug a bass, turn the amp on, I get sound. When no bass is connected, there's a very low hum, that is (very) slightly affected by the gain button and the -10db pad and moderately affected by the treble control.

- I tested the speaker on my guitar amp too and it works perfectly fine.

- The phones output is not working. I wonder if the purple light can be caused by a faulty headphone circuit ? If it’s the case, I can totally live with that.

- The DI output does work, but I can hear some static on the recording.

This is my first attempt at repairing an amp. I think it would be a good learning experience. If all fails, I'm gonna turn it into a cab. Any help would be appreciated.

Thank you

New member elrjay

Hi All. I am a new member from Australia and found many links to articles on this site while I have been repairing musical amplifiers in recent times.
My interest in electronics started around 50 years ago when I was building small DC power supplies and music effects devices - all discrete components.My "old" 6w valve amp from 60 years back is now in vogue! But then I got into solid state transistor powered amps - made a couple of 100w kits all good.
I have been a guitar and bass player for even more years. Since retiring from professional work, I have recommenced my interest in the same equipment but with many "new" components. I have had a few repair successes and have a few projects for further work.

All-DHT amplifiers: no indirectly heated signal tubes!

This thread has the goal of collecting schematics, pictures and comments of any kind about amplifiers where ALL the signal tubes are directly heated - both input and output tubes. The only indirectly heated tubes allowed will be in the power supply or support positions e.g. rectifiers or active loads. Can be push-pull or single ended, and directly heated pentodes are also allowed, in triode or pentode.

Since the majority of directly heated tubes are fairly low mu (up to 15) it is likely that designs will include 3 stage amplifiers or 2 stage amplifiers with step-up transformers in the input or interstage. Maybe also some other creative solutions.

So please post your ideas, schematics and comments of any kind and let's make this thread an interesting one!

To kick off, here's a design by Ale Moglia titled "300B SE Amp: 47 Driver: Going DHT end to end"

https://www.bartola.co.uk/valves/2020/07/25/300b-se-amp-47-driver/
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On sparse speaker arrays

Even the best loudspeakers sound quite poorly in nowadays squeezing living rooms. The problem of room reverberation can be addressed by extensive soundproofing and/or by using very directional loudspeakers - but covering the walls and ceiling with class A acoustic foam is not cheap & utterly destructive to the peace in the family; commercial controlled directivity loudspeakers cost a fortune while developing your own without an anechoic chamber nearby is unrealistic. However, in some situations it is possible to use the room geometry to reduce room reverberation and improve the listening conditions.

Introduction​

Let’s consider an alternative as sparse (3…6 drivers) speaker arrays positioned in the room corners so that the floor & ceiling of a typical room create a tall virtual column. The driver’s height is defined by

gap=(C2F-H)/NS; aH=(0:NS-1)'; aH=aH*gap+gap/2;

where C2F is ceiling-to-floor (typically 9’, ignoring 1’ of ceiling space for pipes, etc) height, NS is the number of drivers, and H is the ear height for a sitting listener (typically 2’8” … 3’). Drivers shall be focused (arched and angled, as per near-field beamforming) onto the DIYer and his ever-grateful family at 9’…11’. In (a simplified) theory, such array shall have wide horizontal dispersion and its impulse response (on the axis) shall look like
rir-6.png


with floor and ceiling reflections interleaving, each having a small (~1/NS) relative amplitude, with loudness decaying as (1/sqrt(distance)), contrary to (1/distance) for point source loudspeakers. However, I could not find any published (by others) real-world measurements, to either confirm or disprove this theory. Thus, I measured what I could myself. Of course, the results are specific to the room & drivers – but something, even little, is better than nothing.

Methods[1]

This design and measurements are concentrated on midrange applications, 300Hz – 3kHz, of the maximal sensitivity of human ear to the Direction of Arrival (DoA).

The drivers are Dayton Audio Sig150-4 mounted in approx. 3.6L boxes angled 60⁰ inwards, which boxes are mounted on a long plank which is mounted into the corner of a somewhat spartan room 25’ x 14’ x 9’. Multiple drivers were connected in series.

P7130004.jpg

P7130006.jpg


The amplifier output was set to get 80 dB SPL @1m from a single driver. A (pre & post) calibrated 12 dBA cardio microphone on 2i2 Gen 3 USB audio interface was used. Room Impulse Responses (RIR) were obtained by a regularized Exponential Sine Wave (ESS) method Loudspeakers for AEC: Measurement and Linearization - File Exchange - MATLAB Central (mathworks.com) (to be updated).

The RIRs were postprocessed to extract the Direct to Reverberant Ratio (DRR(f)) as the Y-value that the polynomial approximation of the corresponding normalized reverberant slope(f) crosses at X=0:
rir-drr.png

Obviously, speakers’ arraying does not affect RT60(f).

Measurement distances are 100:50:400 cm, with 300cm as target. The default measurement height is 85cm, to be varied 70:5:100cm. The first 20ms of RIR, DRR, arete (top of the ridge) of wavelet-transformed RIR, and total FR are considered to be more important than other metrics.

Results​

number of drivers [h->1, n->3, m->4, n->5, j->6] summary​

rir-13456-300-85.png


fd-13456-300-85.png

At target distance & nominal ear height. See the details below.

Single Driver​

rir-1-distance.png


fd-1-distance.png


Red lines are for quasi-anechoic conditions (alas, wrong polarity).

6 drivers​

Variations of Listener’s Distance​

Here and below, red lines are of a single driver at 300cm, as the reference.
rir-6-distance.png


fd-6-distance.png



Variations of Listener’s Ear Height​

rir-6-height.png


fd-6-height.png


5 drivers​

Variations of Listener’s Distance​

rir-5-distance.png


fd-5-distance.png


Variations of Listener’s Ear Height​


rir-5-height.png


fd-5-height.png

4 drivers​

Variations of Listener’s Distance​

...oops, only 20 images
Variations of Listener’s Ear Height
3 Drivers
Variations of Listener’s Distance
Variations of Listener’s Ear Height
6 drivers, straight line [e] vs angled only [f] vs arched only vs arched and angled [j]
rir-angled-300-85.png

fd-angled-300-85.png

As usual, [h] stands for a single driver as baseline.

Other Observations​

  • The measurements show only 10 dB of improvements in DRR but subjective perceptual improvement is much stronger, far beyond expected.
  • Due to 15.5 dB lower SPL at each driver, non-linear distortions and Barkenhouse noise are proportionately lower.
  • The dynamic range widens, high SPL are not distorted.
  • The (1/sqrt(distance)) effect is realized only partially per measurements but nearly fully by perception.
  • Judging by ear at 10’, 6 drivers with 90dB SPL @ 2.83V sensitivity are louder than a 110 dB SPL single driver, ~same Z.

Discussions​

Many more drivers & configurations were built and measured in the last year. None of them contradicts the data above, and these have relatively low additional meaningful information and were omitted to avoid overloading readers.

TBD

Conclusions​

  • Sparse speaker arrays are a compromise. Particularly, they are not a good fit if you require that listening experience shall be invariant to ear’s height. Left/right/closer/farther are not a problem.
  • You do not need 20 drivers / column to realize line source advantages.
  • You do not need near-perfect expensive midrange drivers to get decent sound.
  • You do not need 400W amplifiers, for sure. Even 100W is too much.
  • An average DIYer can build a sparse speaker array and adjust it to his preferences without getting into much expense or serious technical difficulties.
GNU General Public License v.3+ https://www.gnu.org/licenses/ applies to all information provided.



[1] Disclaimer: The objective measurements below are somewhat related to but do not characterize human auditory perception to any highly disputable degree of fullness or precision.

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Another interesting Aluminum clad horn tweeter active loudspeaker from Oz

cast-speaker-tom-fereday-piit-giblin_dezeen_2364_col_3-scaled 2.jpg


Not the usual source, an Architectural “magazine”. Looks like a waveguide, which has one asking, “Midrange?” but there is a short c0omment at the very end that suggests it is an MEH, but the second picture (and the makers bumf) suggests something else. Pretty cheap given the amplifiers come with (if pricng is /pr)

https://www.dezeen.com/2024/11/05/aluminium-cast-speaker-tom-fereday/

tbij4vofyQsG6RQncButgU-1600-80.jpg.png


Pitt & Giblin: https://www.pittandgiblin.com.au/cast

More coverage:
https://www.wallpaper.com/tech/cast-aluminium-speaker-tom-fereday-pitt-giblin
https://leibal.com/products/cast/
https://www.gessato.com/cast-aluminum-speaker/

dave
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S. Harsch XO

This work started in the Filler Driver thread. I was studying transient XO's and built a test bed speaker to quickly test out new XO ideas with miniDSP. Here is a photo of my test bed. It is a basic open baffle for ease of construction and has a pair of Eminence Beta 12cx (no CD, with coax horn sealed up), a pair of PRV 5MR450NDY mids, and a PRV D280Ti with a PRV 45x45 deg waveguide. These are all fairly efficient drivers and are what I had on hand. The amp is provided by TDA7498 100w class D amps powered by 19v (mid and tweet) and 24v (woofers) SMPS bricks.

494160d1437200618-filler-driver-ala-b-o-gen2-beta12cx-5mr450ndy-d280ti-front-photo.png


I first started with the B&O XO which requires a wide bandwidth mid that is band pass filtered with 1st order Butterworth and straddled on either side with a LR2 filter for the woofers and tweeter. I was then turned onto the XO topology proposed by Samuel Harsch by member Phase_Accurate. The paper is in French but I figured out that the basics are:
1. Set the low pass filter for the woofer as a 4th order Butterworth at central frequency, fc for the XO centerpoint.
2. Set the high pass filter for the tweeter as a 2nd order Bessel at fc.
3. Set the delay of the tweeter equal to 1/2 of the period of one cycle at fc.
4. Use all positive phase on woofer and tweeter.

Here is an example of the XO plot from the Harsch paper:
494790d1437489518-filler-driver-ala-b-o-harsch-xo-plot.png


Here is the step function from the above XO as predicted by Harsch:
494791d1437489518-filler-driver-ala-b-o-harsch-xo-sr-plot.png


So I tried it out on my 2-way 10F/8424 & RS225-8 FAST Ref Monitors and got this for the XO plot:
495043d1437630819-10f-8424-rs225-8-fast-ref-monitor-harsch-10f-rs225-xo.png


For the 10F/RS225 FAST, this was the corresponding measured Step Response:
495046d1437630819-10f-8424-rs225-8-fast-ref-monitor-harsch-10f-rs225-sr.png


For the 10F/RS225 FAST, here was the measured phase (notice the little bump at the XO frequency as predicted - but no wrap):
495044d1437630819-10f-8424-rs225-8-fast-ref-monitor-harsch-10f-rs225-phase.png


For the 10F/RS2225 FAST, here was the measured group delay, not over 5ms above 40Hz:
495047d1437630819-10f-8424-rs225-8-fast-ref-monitor-harsch-10f-rs225-gd.png


What the Harsch XO does nicely is that it allows one to use a steep XO to block out the nasty cone breakup on a woofer with a high order filter. Yet still not have a large phase wrap in order to maintain a time aligned transient perfect response.

So I got to thinking if this can be applied to a 3-way by mirroring the XO topology. That is Butterworth 4th order for the low pass on the woofer, Bessel 2nd order for the mid high pass and Bessel 2nd order for the mid low pass (or Bessel 2nd order bandpass for mid), and a Butterworth 4th order for the tweeter high pass. So what this means is that pretty generic components can be used in that a special wide band mid is not required anymore as the filter is already a fairly steep 2nd order. And for the woofer and tweeter, we can use 4th order filters which are great for controlling distortion, and preventing cone breakup leak-thru from the woofer.

As I had a 3-way XO test bed already handy, it was just a matter of putting the XO settings into miniDSP. So here is the electrical XO I implemented in miniDSP:

495707d1437975878-s-harsch-xo-harsch-3-way-minidsp-xo.png


And here is the resulting acoustic XO that was measured (yes the SPL levels are fairly low so that I did not wake up the family during sweeps). As you can see, the curves are fairly common and innocuous. The 1st order textbook curves of the B&O were especially hard to achieve in practice. With the Harsch topology 2nd order and 4th order are much easier to get. The tricky thing is what to do for the time delays? Well, it is exactly the same delay assuming driver acoustic centers are already time aligned. The mid needs to be delayed from the woofer by 0.5/fc_low and the tweeter needs to be delayed from the mid by 0.5/fc_high. For my case, I used a 500Hz and 3500Hz XO points:

495708d1437975878-s-harsch-xo-harsch-3-way-xo.png'


Here is the resulting Impulse Response:

495709d1437975878-s-harsch-xo-harsch-3-way-ir.png


And here are the resulting Step Responses - I know there is a little pre-rise but overall it is much better than a jumbled up-down-up-down mess you would see with an all LR2 or LR4 XO:

495710d1437975878-s-harsch-xo-harsch-3-way-sr.png


This is still a work in progress. Does anyone see any problems with it? I think this works. I am still having a problem getting the phase to be wrap free - it seems to wrap at the centeral frequency and not at the low and high XO frequencies. Anyhow, still debugging it. This XO may be useful for folks looking for an easy to implement quasi-transient perfect XO using IIR filters in DSP. How does it sound? Fantastic! My speakers are essentially 98dB sensitive and super loud and clean with this XO. I am really enjoying the sound and how the percussive sounds like guitar and drums really have a "live" feel to them.

Attached are some sound clips (stereo recording using Zoom H4 of the mono- Right channel playing). Change file extension from .asc to .mp3 in order to listen.

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Office Build Thread Version 2 - things changed...

Alright so we are here again. Originally I was going to run only run two monitors in this new office but I need to go to three. I could move the monitors aparts by maybe 5 inches but that will make working harder. More head movement.

So yeah, once I actually got in here I realized the 3 way I was making is just going to be too large for this work area.

What build would you make for this setup? I really want a three way but I don't think that is possible unless I mount the speakers on the wall above the monitors. I could put them on adjustable arms have them pointed down at me. The window kind of gets in the way of that.

I really don't want to be boring and put a couple 3" Mark Audios in some spherical enclosures underneath the monitors pointed up at me.

Anyways, let me know what you'd do if this were your new space

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XSim free crossover designer

I'm announcing the public release of XSim, a "free-form" Windows-based passive crossover design and simulation program intended to be as intuitive and non-restrictive as possible. . The program is ***free*** for personal or commercial use. Any speaker builder with at least a basic understanding of what resistors, capacitors, and inductors do, and with decent response measurements (FRD and ZMA) of their drivers in their boxes, should be able to get useful results right away.

download from:
http://libinst.com/Xsim/XSimSetup.exe

Experimenting with "ideal" driver data (the default for each driver) can also be fun and helpful in understanding the effects of various circuit types. Some built-in multi-part "Circuit Blocks" also provide for basic pre-configured circuits to be tuned per their overall parameters (Q, corner frequency, attenuation, etc). There are "Example" files inlcuded in the XSim download (currently less than 1.5MB) as well.

XSim allows you to see simultaneous multiple graphs of your choice, including impedances, voltages, power consumptions, group delay, and frequency, impulse, step, and square wave responses. You can watch the performance and/or stress on any part vary as you tune structure or component values in real-time. The circuit network is re-solved automatically with every change you make.

The "free-form" aspect of XSim is that crossover designs are unrestricted, no particular circuit structure need be used. Essentially any R-L-C passive network you can think of can be easily entered using graphical schematic entry, with results calculated as you design. A common "ground" node is provided, but its use isn't required.

screenshot2.png


Circuit designs (incuding all in included driver data sets) can be shared by email or forum post in XSim's ".dxo" format.* Graphic files can also be saved in standard picture file formats.

Stuff yet to come: [edit: the following aren't actually going to happen now, as it is too much work to herd vendors into getting this info together and hosting it on their sites!]
The release version of XSim featiures automatic downloading of part and driver information based on vendor and part number, making it easy to find and order what you need to turn your design into working hardware. However, no vendors are as yet signed up to provide XSim access to part or driver information. XSim will be able to download these lists itself, when available. XSim can also update its own program (with owner permission) so people who use it online can always be up-to-date. ...And so I can add or fix features without asking you to deal with downloading and installing each time.

XSim does not yet include geometric features (such as varying mic position, baffle or boundary-bounce effects) as in Jeff Bagby's very successful "PCD" and other programs derived from it. But that is in the works, with some assistance from Jeff. There are also as yet no box-design features, but this is also planned so that full designs can be worked with using the infinite baffle data provided by vendors. [these might still happen, when/if I get back into programming and study mode 🙂 ]

edit: * as it turns out, this forum doesn't let you attach the .dxo files (or other files it doesn't know about). So you'll have to first zip the file and then post the ".zip" file containing the ".dxo" file. On downloads, do the opposite: unzip and then open the .dxo file into XSim. Maybe someday we can get diyaudio.com to let these files be attached? They are only text, no way for a virus to sneak onboard one of them!

Added Nov 2017: There is a procedure summary for collecting and adjusting driver FRD data for correct relative delays (which you need to accurately simulate crossovers) here -- XSim free crossover designer
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Introduction from UK

Hi all,

Audio well where do I start?
Grown up in a music cafe with Saturdays live music created interest at a very young age
Then when 16 became a FOH Sound Engineer in the late 70s so all ANALOG gear 🙂 but of course moved over to Digital

Other hobbies are electronics, audiophile, sea sailing and mostly computing as work next to FOH

My Audio start time was mostly McIntosh, Crown, Altec & JBL and later T+A

Current having Urei speakers (love point source) and Tube Amps with one SE EL84 from the late 60s or may be 50s and a Croft Phone Integrated (8 EL84)

For Sale Tubelab SSE, 2x Simple SE, TSE-II, populated

Various Tubelab populated PCB Board for sale.

I populated them with quality parts, then never completed the builds.

They should all be working, but you may want to double check them in case I made any mistake on them.

  • 1x Tubelab Simple SE (dated 2006, no cathode resistor installed), $100
  • SOLD: 1x Tubelab SSE (dated 2012, 680r cathode resistor installed), $100
  • 1x Tubelab Simple SE (setup for KT88), $100
  • SOLD: 1x Tubelab TSE-II (set up for 300b)
  • SOLD: 1x Tubelab TSE-II (set up for 45, photo attached), $100
  • SOLD: 1x Tubelab SSE (dated 2012)
  • SOLD: 1x Tubelab TSE-II (dated 2020)
  • 2x Tubelab Universal Driver Board PCB 2015, new unused $30 each

Need to add shipping from Brooklyn 11211.

Accepted payments:
  • Zelle
  • Paypal (add 3% fees if "as purchase")
  • Venmo (add 3% fees if "as purchase")
  • Revolut
  • Wise

Thanks

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I built this with your help now I need advice

With lots of help from posters on the site, I put together this modular three-way top for small outdoor concerts. It's still a work in progress, but so far I'm thrilled.

It's a 3 inch compression driver, a 10 inch cone and a 15 inch cone. The 10 inch is from an EAWKF 650, and the 15 inch is from a bag and TA 15. I have attached PDFs of what multiple sources have stated are the closest equivalents to those drivers in aftermarket gear.

It occurred to me that the TA 15 I got the 15 from was crossed over at the same point the 3 inch compression driver crossed over, so for smaller shows I could bring a two-way speaker, the main drawback being that a 15 inch will be beaming for the upper frequencies. But it worked pretty well for shows where there was small audiences and limited budget.

The other day I noticed that the specs on the 10" driver, which actually weighs more than the 15 inch driver (it's a beast!), is rated to go down low enough that I could create a two-way system with a top two components crossed over at say 120 Hz.

With some planning and effort, I can pull the equipment out of storage and find somewhere to test it outside of a gig, but I still felt like I wanted to ask advice, in case either it was categorically doomed to fail, or if I'm taking a chance of some sort of long-term problem if I take this route.

it feels very counterintuitive to me, replacing a 15 inch with a 10 inch, but again, this 10 inch weigh more than the 15 inch and has amazing specs for excursion and says it can cover the frequency range.

So please share opinions about whether this is something worth exploring (or avoiding it all costs 🤣).

IMG_8196.jpeg

Attachments

Velodyne FSR Series -

My Velodyne FSR 10B6 sub was working for about six hours and suddenly made a cracking sound then the sound stop but the subwoofer power LED still shows Green. I turn it off from the wall socket and after about 3 mins, turned it back on. Now the Subwoofer power LED is green but when I connect an audio jack using line in, there is no sound from the Subwoofer. Please, I need help. Thanks

WAD Phono II CCS Modification

Ages ago I modified my WAD Phono II scratch build. Inspired by a Nick Gorham modification (attached), I added CCS on top of the first stage. I also put a CCS under the third stage CF (not documented here).

After a couple house moves, I lost all documentation of my current build. And I like it so much, I thought I’d document it here.

So I pulled the bottom off my phono stage to take a look at what I had done.

The CCS on top of the 1st stage is classic Gary Pimm, and I documented it on Nick’s modification attachment. I believe I set it for 0.9mA, but it looks to be 0.94mA as measured.

The first stage is the hard part as I understood the mod did change the RIAA equalization (3180+318).

I did the math back then (I think), but I document the ("unchanged") first stage output resistance here.

The WAD ECC83 operating point (schematic attached) is B+ = 250v, Anode load resistor is 100k, 150v on the plate, -1.1v with 1.2k (un-bypassed) on the cathode. See the load lines attached.

I calculated anode resistance as (332v-78v)/(3.5mA-0mA)=72.57k

Since it is not bypassed and u=100 and cathode resister = 1.2k, I get 72.57+[(100+1)*1.2]=194k

That in parallel with the anode load of 100k = 66k

With the CCS load, I used 1M resistance (Nick Gorham's suggestion) in parallel with 72.57k, giving 67.7k

So 66k vs 67.7k per my calculations. Close enough?

Referencing Morgan Jones I can’t prove out the rest of the equalization circuit (oh well). But I sure do like this mod.

I’m thinking one day to figure out how to put a CCS on top of the 2nd stage valve. (Allen Wright's approach - if it is not supposed to move, lock it down tight.)

I attached a copy of the voltages of my modified build too.

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Yamaha R-N 500

Good morning,

I am looking to repair a Yamaha R N500 amplifier.

The amplifiersafe in safe mode. Nothing is displayed on the screen (which doesn't make things easy!). Only the power indicator seem to work.

I forced the amplifier to start using the method described in the service manual by protecting the amplifier with a safety bulb.

The relays have all activated but not the screen!

In terms of voltages it doesn't seem bad, I have a 3.3V, a 5.5 and a 12V.

The temperature sensor is not the cause of the problem either.

Do you have an idea or a path to follow to try to troubleshoot it ?

Thanks in advance

For Sale Some parts

I have for sale some parts ,some used and some new.Plese look on the picture.If there is something you are interested just contact me.If you have question also plese contact me.Payment by paypal.

List!
Clarity ESA 250V caps:
2 x 9.1uf - SOLD
2 x 4.5uf - SOLD
2 x 6.8uf - SOLD
2 x 3.0uf - SOLD
1 x 12.0uf
2 x 14.0uf
2 x 68.0uf
Clarity Csa caps 250V:
4 x 1.0uf - SOLD
Mundorf caps:
2 x 1.5uf Supreme Evo oil - SOLD
2 x 1.0uf Supreme - SOLD
2 x 1.8uf Supreme - SOLD
2 x 0.22uf Supreme - SOLD
1 x 0.47uf Supreme
2 x 2.2uf Evo oil - SOLD
2 x 2,2uf Evo Silver Gold Oil - SOLD
2 x 0.22uf Supreme(red writing)-SOLD
Jantzen caps:
2 x 5.6uf Standard Z
4 x 0.82uf Superior Z
2 x 0.47uf Alumen Z - SOLD
2 x 0.22 Silver Z (totaly new)-SOLD
Duelund caps:
2 x 0.1uf JDM tinned copper- SOLD
Mundorf back lack coils:
2 x 0.68mH - SOLD
2 x 5.6mH - SOLD
2 x 0.33mH
Intertechnik coil back lack:
2 x 0.15mH
Russian pio caps K-75 series:
2 x 0.1uf 500V
Ampohm PF-XTI Alu pio caps:
2 x 0.1uf
Miflex Kp-Al caps:SOLD
2 x 0.022uf
Mundorf M-Resist ultra resistor:
2 x 33.0 ohm
Tube socket:
2 x UX-4 backelit top quality-SOLD

2 X 1,20 mH Jantzen air coil 16awg

Jantzen Superior Z caps:
2 x 4,7uf
2 x 1.0uf
Jantzen Standard Z caps:
2 x 3.3uf
Mundorf CfC16 Copper foil coil:
2 x 1,8mH
Mundorf air back lack coil:
2 x 0.18mH

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