I have this model of my midbass enclosure in Sketchup but it won't give me volume. As a matter of fact it won't even close up the sides, maybe something about the odd shape but none of these surfaces are curved just different angles. There is one side it closed by as you can see I made an X in the other side and it didn't close it up.

Solid Inspector has errors but they dont' make any sense to me. I even erased half of the walls and redrew them, then deleted the others ones and redrew them with no difference made. I don't want to go down the math rabbit hole on this one!


I dow

Hi,
I've just finished a protection schematic for my audio system. Some circuits are my idea, others are found on the internet. The schematics are tested only in LTSpice so far. I want to know your opinion before I start building it.

It has the following features:
- 2.2s start delay
- disconnect speaker in 120ms after power down
- disconnect speaker in 50ms after 50V DC on amplifier output
- disconnect speaker in 100ms after 1MHz @ 3Vp on amplifier output
- disconnect speaker in 1...10ms after exceeding maximum current
- thermal shutdown with programmable temperature
- variable fan speed controlled by temperature
- clipping indicator

If you know other usefull fetures, please let me know.

SOLD



For sale is a pair of unused Genelex Gold Lion Reissue PX300B tubes manufactured in Russia and not presently being imported.

The tubes were taken out of their boxes for inspection and photos, I added an R and L to the bottom for identification.

They have not been in an amplifier.

Date codes match at 20 02

Factory testing matches at 57mA @ a gm of 5,600 umhos

Come in original boxes.

I will ship only via USPS Priority insured mail.

For sale in Germany a complete kit for a Salas DCG-3 preamplifier as it came from Tea-Bags group buy: https://www.diyaudio.com/archive/blogs/tea-bag/865-group-buy-salas-boards.html

Price: €70,00 plus Shipping

Hi people,

Long time lurker here, finally decided to make an account to ask a burning question.

I recently came into possession of a NAD 705 that sounds very nice but runs way too hot. Within minutes of being turned on the central heat sink would literally get hot enough to burn fingers. Top case off made no difference. 😵

At first I thought this might be a DC offset issue, but I measured the speaker terminals and got 15mv, which was within spec. The I did a visual inspection, looking for bulging caps and burnt looking resistors, couldn't find any either. I'm at a bit of a loss what to check next.

What does the collective wisdom here think? Any tips for me?

Service manual attached, for anyone interested.

Cheers

I've just finished a full recap and service on my recently acquired Philips CD100 vintage CD player.

After overcoming a few unexpected issues the player now reads and plays well part from the fact that towards the end of a disc, say track 10 or 11 on a typical album, the player starts to make a strange high pitched ringing noise.

It sounds just like running a wet finger around the rim of a wine glass! Up to around track 10 it is silent and then beyond that point the ringing noise starts.

Has anyone here ever experienced a problem like this and what did you do about it?

Hello everyone!

I was wondering if it is possible to run TPA3255 in a composite amp configuration? LM4562 is something that everyone seems to be using as error correction opamp, so I figured will be nice combo.

Where can I find resources on that kind of information?

Thanks in advance,
Pavlo

OK-

So I need a sanity check- I have all the parts- needed. FRD/ZMA for SDS woofers for simulation provided and measured on 37.5H 8.5W 10.5D box with DATS by someone in another DIY forum.
FRD/ZMA for D220TI were also provided and measured I was told- not sure baffle size...anyway I am just trying to get close and use some these drivers I have. WG will most likely be HM17-25 seeing as I already have 2. I have 3 dayton audio H6512 but not sure how id like such a large wave guide going vertical. If you think im being an idiot please let me know lol- if you see a issue please explain it to me like im an idiot-

This will be a good way to use up a lot of stock sitting around and if it turns out they way I hope, I plan to do the rest of my speakers like this for our theater.

SDS 160F25PR01-08 is the woofer. again I have everything I need minus the 30uf cap if thats better- I have 1.5Mh but no labels haha so will be pain figuring out which ones they are..


With this XO i have 5DB NULL at XO when inverting tweeter..




WITH THIS VERSION i have 10DB NULL AT XO- I just changed CAP on Woofer to 30DB

Hey everyone,

New to this forum.

I'd like to get better (and nice looking!) sound in my small office room, which is about 10 square-meters big, even less when I come to think of it. I'd like to get a pair of regular sized bookshelf speakers together with a vintage Marantz receiver.

I love the look of the 22XX-series, but I can't really decide (or know) how much power I actually need. I've heard that "marantz watts" can't be compared to a modern receiver denon/onkyo watts, and that's my only reference as I've never owned vintage.

I know the 2270 is the one to go for as you can never get too much power, and things tend to sound better even on low volume with more power. However the price for the 2270 is a bit steep so I'm thinking if the 2245 will be enough. Again, I want to emphasize that I won't connect floor standing big speakers, but a set of smaller bookshelf speakers. To give you any reference, I am currently running a Marshall Woburn in this room, and I found that loud/good enough for the size of the room.

Do you think I'm set with the 2245 or would you go 2270? Or is even the 2245 too much for what I'm pairing it with? Will even a 2215 do?

Thanks for any input, completely new to vintage receiver.

Vol

I have a unique situation and as I'm more mechanically inclined that electrically, I could use some help from the resident experts. I have the Corsair HS60 Haptic headset. For bass frequencies, they use a Taction Technology tactile transducer instead of an audio driver. Thus, it seems they have an analog or digital crossover to divide the audio frequencies between the tactile transducer and the traditional audio driver.

For my application, these are the only headset/headphones that are desired and currently commericially available. Taction Techonology used to sell the Kannon headsets which have a 3.5mm/1/8" TRS plug. While I also own a pair of the Kannons, as they have been discontinued for a while, they are expensive, somewhat fragile, and hard to find used.

What I would like to do is convert several Corsair HS60 headsets, which only accept digital audio via USB, to analog headsets (stereo audio + microphone). I recall in car audio enthusiasts would tap into audio components PCBs before or after the DAC (I don't recall which). Thus, I am wondering if it is possible to bypass the DAC on the HS60 headset so it can be converted to analog?

One complicating factor is the dual drivers (tactile + audio transducers) and whether the crossover is digital or analog.

It seems the current HS60 audio path is one of the following options (Left/right audio only, excluding the headset microphone):

1) USB>Digital Crossover>DAC>Analog amplifiers
2) USB>DAC>Analog Crossovers>Analog amplifiers

I suppose there is remote possibility that the amplifiers could also be digital, but I doubt it as digital audio amplifiers are very rare, especially in headphones.

Is it relatively easy to identify the DAC/analog crossover on a PCB? If it is, then it seems that it should be possible with good soldering skills to convert this headset from USB audio to analog. Another complicating factor is the amplifiers likely drawer power from the USB port, so a 5VDC power source would likely be needed.

Attached are some photos of the PCBs (there is one in each ear cup) for the experts who understand the components. These are old photos, so if I need take the HS60 apart for better photos, let me know.

I'm curious if just from looking at the PCBs whether it is obvious if the crossovers are digital or analog and if the analog signal after the DAC can be tapped into to convert the headset microphone audio and Left/Right channels to analog.

I am looking for plans for a simple, easy to build, light weight top that is useable with the Tham15, which can't play very high.

Can anyone link or suggest some recommendations? Everything I am finding doesn't go low enough or is way too heavy/complicated with too many drivers.

I am very green and my building skills are new.

Eminence, B&C and Goldwood drivers all available here, but I struggle to find most other brands.


Thanks!

Hello Jason! 🙂
I'm not sure wether I updated my membership (and to what level), so there are 2 questions:
Can I see my membership-status?
Can I up my membership (from aficionado to fanatic or even conoisseur)?
thanks, for everything!
david

Hi, i building this kit from Aliexpress.
im repurposing the heatsinks from an Onkio amp I salvaged for parts.
made some modification with:
1. added foil capacitors parallel to the original electrolytic once.
2. bypassed the power supply diode with a short.

i wanted to cramp it all in a reasonable small enclosure. but common sense indicates EMI RFI potential interference.
what do you think? i guess i can try it before i cut the rear panel with CNC. if the SMPS will stay out i will make the enclosure smaller.

link to photos

A few months ago, @gpapag posted up some linearity plots of a couple of experimental 'Abbado' DACs I'd sent him. Below around -66dB (equivalent to the 11th bit), the results begin to look not too great, certainly not in comparison with DACs based on much more modern S-D chips which tend to have near-perfect performance on this particular metric. I played around with making some low-level measurements myself and confirmed the basic premise that low-level linearity is an issue and from varying the input levels t looked to be due to inaccurate bit-weightings in the DAC's internal architecture. What I didn't know though was how big of a bit-weight error is needed to give rise to such deviations from linearity - I wanted to get a handle on the order of magnitude involved. I figured maybe I could coax LTSpice into doing some simulations of this.

LTSpice allows input of .wav files to the sim, the data is converted to analog automatically before being applied to the circuit. What I really needed in order to test out DAC linearity was the ability to input digital data, not analog. As far as I'm aware that's not an available feature so I wondered if I could develop an ADC which would convert the .wav file back to digital in the simulation and then I could apply that digital stream to my simulated DAC. By tweaking the bit weights in the simulated DAC I might be able to see how linearity is affected and get a feel for the degree of effect. But how to design an ADC?

There are two kinds of ADC which might be suited to this task - a 'flash' and an 'SAR' type. The flash type has the disadvantage of complexity - it needs of the order of 2^N comparators for an N-bit ADC. I figured I'd want to begin with a 5 or 6 bit ADC and to save on a lot of copy pasting to create a flash ADC I wondered if the SAR type might be better suited. An SAR ADC only needs a single comparator but has a significant disadvantage of needing a clock and a register to remember the results of earlier comparisons. I've never built any clocked logic circuit before in LTS and, being very lazy I decided to at least try to combine the clock-free nature of the flash converter with the low component count of the SAR. Perhaps I'll call my approach the 'FA' for 'flash approximation' - it falls between the two in complexity, only using one comparator for each bit. It needed a little bit of tweaking to create what appeared to be a working 6 bit ADC out of 12 voltage controlled switches and 6 reference voltages - being as the switch type in LTSpice is a single pole and I wanted a changeover function I wired two single pole switches in anti-phase. Of course such a design would never work in the real world but that doesn't matter in this instance, its only going to live inside the sim. Being as it produced a nice stepped waveform for an input sinewave with very little effort I began to be suspicious if there wasn't some hidden gotcha going to bite me further on down the road. So I needed to characterize my 6 bit FA ADC to ensure such a simple design really delivered the goods.

To characterize it it needs to drive a DAC back to back and then I can do tests on the DAC's output waveform. The DAC is almost the same as the ADC and uses 12 voltage controlled switches and again 6 voltage references (but different instances) for the bit weights. I hooked up a sinewave voltage source at full-scale into the input and ran an FFT on the DAC's output. Looked respectable indeed and the SFDR was in the region of -60dB, too good to be true! I took the input amplitude down to 2LSBs (about 30mV) and wondered what would show up. Quantization distortion, that's what. Ah, dither would come to the rescue I thought. I had to delve into the workings of LTSpice for this as I'd never created an explicit noise source before - the 'White' function serves a good purpose. TPDF dither (needing two uncorrelated sources summed) would have to wait as just creating two instances of 'White' can't uncorrelate them. How to set the dither amplitude? I figured do it by eye when the FFT completely lost the spurs of the harmonics of the test tone. The 'White' function also has a parameter which seems to relate to the bandwidth of the noise generated, I got decent enough results setting this to 44,100. But I couldn't resist high-passing the output before applying it in series with the test tone.

So now I had what appeared to be a kosher ADC feeding a perfect DAC and at least based on FFTs, the results were as close to textbook as I could have hoped. Time came to try making some linearity measurements - I figured to get the best measurements of amplitude I'd want a low-pass filter do get as clean signal as possible, so I plugged a 1kHz 5th order Chebyshev lowpass on the output of the DAC, the stimulus being 1kHz. Using the 'step' feature I ran a sequence of different levels into the ADC and made '.meas' of the output. Still being a bit suspicious of my ADC though I also put the amplitude quantized analog version of its own digital output through another instance of the same filter and compared. The ADC line wasn't straight at lower levels, it had a bulge. Hmm, what could cause this?

<attached George's linearity plot of the prototype Abbado DACs which use TDA1387>

I'm still fairly new to DIY audio and had a quick question (that may fall into "there are no stupid questions, but this one is stupid").

I have a TOA P-300D I am using as a sub amp, I do realize it is not the best amp but it was part of audio lot being sold by my local school. I do not think it has been serviced in quite some time but it is in operable condition. My question is will servicing the amp have a chance to improve the headroom of the amplifier. Within its limits it sounds fine but once it crosses over into clipping (even for brief transients, like a kick drum) the distortion really picks up.

Or is it simply just the wrong tool for the job? My guess is this is an amplifier more for a paging/intercom (very compressed signals or speech) in its design where it can provide its maximum RMS power forever, just don't ask for more. The user manual does show the power curves vs distortion and past its rated power there is a huge almost vertical spike in distortion. So maybe it is only operable for 1/4 of its RMS rating for musical applications.

I know some of the other more dramatic symptoms of an amp needing service but I'm not sure if this is one or not.

Any input will be greatly appreciated, thanks in advance!

Also incase it would help with the question, the amp is running a jbl 2242 in a 9 CF box tuned to 25hz, with B6 alignment (almost like a 4645C), but it is 1 CF larger so it would have the same footprint of the Altec 518 cabs (speakers that came with the amp). I know the 2242 are not the ultimate in VLF reproduction, but once again the price was right ($200 for an unopened box NOS pair)

I believe PLLXO is the quintessence of crossovers. Super clean, no ill effect, just pure loveliness. But they have a huge problem. Matching with the actual amplifer and its input impedance. And especially your next amplier. Until now! Our hero enters - Adason!
Here is his contribution to humanity - a buffered PLLXO.
His own thread: https://www.diyaudio.com/community/threads/simple-active-crossover.368702/


I have a strong desire to try this out. This thing is sleak!

However, I lack any skill, or even talent, in the electrical field. I have to do this in the short brakes I have from cutting human tissue samples looking for cancer, and other unimportant tasks.

I have come up with this completing schematic which might help anyone who tries to mimic this circuit.




Some things I am trying to figure out:
¤ Bi-polar caps on input and output. Should probably be used.
¤ Dual Rail - important to understand how this works. See V+ and V- and ground in schematics.
Adasons preference is something similar to this PSU that can reduce the need for the 100uF/V35.
https://www.ebay.com/itm/164845881741?hash=item266194898d:g:ObUAAOSwxmdgj2z3&amdata=enc:AQAHAAAAwNGeP5fYh2oIx+yK4Io3k2fHDvEu5iQEkIL08ckzHZTCZERincMe2HroKGMBjI9NoePqlCIF04GR+QU43jNO/3SE6NKRPArKb8eDvh8Lhgtr3PSM1NmE3c7C2cJ99GSr5GFyHrxVm6oxJTKYz/Yuj6Xlceo0NN9gfvm1zdjGnQFIirgfSnixy3ZW1k/WGBwITaSy4U3TtIW7bX6Sjd8JnXKltnMAp0I7bkxrjvW2TcZIuZsMh6rbXCG9i4b9EM1rkA==|tkp:Bk9SR_jc5unWYA

¤ How to modify the circuit for other frequencies:
The schematic says first, choose C. Well, choose C based on what?
In Dave's pages, which Adason refers to, is says choose R.
Somewhat contradictory,
take
Let a known example of the 150Hz that Adason was working with:

If C= 0.1uF
Then,
R = 1/ (2 x Pi x "crossover frequency" x C
= 1 / (2 x Pi x "crossover frequency" x 0,1 uF
= 1 / (2 x Pi x 150Hz x 0.1uF
= 1 / (2 x 3.14 x 150 x 0.1uF) = 10k Ohm

Adason is using these values a lot in his schematic. I am not sure if every value is based on this formula.

And reversed:
C = 1/(2 x Pi x "crossover frequency" x 10 000 ohm
C = 1/(2 x Pi x 150Hz x 10 000 ohm)
= 1/(2 x 3.14 x 150 x 10 000 ohm)
= 1.06157113e-7 F
= 0.106 uF
= 0.1uF

0.1uF - same can be found everwhere in Adaons schematic.
According to Adason, if using a really clean PSU, you might not need to use the unmarked "rails ripple shaving caps", as ZM so kindly pointed out that they were called.

Anyone are welcome to help me and others to understands this circuit and solder ourselves a working example. Please correct it!

Hi,

here's something that may sound like a beginner's question, but believe me, I really don't know how to: I've got some old transformers from Hammond organs. Their leads are isolated by some white plastic material covered with coloured cotton (?) fabric. How do I remove the isolation successfully? I tried my insulation stripping pliers that works well with modern PVC and silicone isolations, but fails here. Using a knife always leads into cutting some of the braid wires. How do you guys proceed with that?

Best regards!

Same old story, the freaking neighbour's dog keeps barking, yapping and howling almost every day all day long when left in the garden. Not sure what breed this dog is but it barks even without any reason, even if neighbours are at home (but not in the garden).

We have a wooden fence. I've tried 4 different ultrasonic deterrents but none of them were effective. I've done some Googling but nothing seems to be working.
It is not possible to do direct line of sight. Cutting a hole in the fence and putting the device there is also out of the question.

I'm in the UK so 👎th amendment or whatever you call it in the US doesn't exist here. Complaint, court, local council or any other ******** doesn't work. I've already tried. Can't shoot that fu**ing dog because we don't own guns. My neighbour is an *******. Want to try something which works without resorting to the "Ultimate Solution".

Does anyone know any device, custom build or how to build it or whatever works in shutting up this neighbour's dog behind the wooden fence?

When making an enclosure, ideally one would make the walls non-parallel. But how is non-parallel is non-parallel? What about 5 degree on the front, is that non-parallel enough with a vertical back? I would assume at only 5 degrees enough wave would get back to the front. So what is the lowest limit? My thought would be one where the other side would not hit the other wall with a perpindicular line so maybe this would vary with the distance between the 2 walls? What has your experience been?

hello everyone, I am writing because I would like to have some opinions on a midwoofer that is rarely talked about on the forums and that is the seas w12cy003.
in particular, it could be used with a 9500/9700 or the illuminato 660000 scanspeak tweeter?
my idea is to build two mini monitor of 4 or 5 liters.
thank you

Hi all,
I'm building a pair of wmtw towers. Drivers are Beyma 15lex1600nd, 12p80nd and JBL 2451 in a JBL 90x50 waveguide. Om my testboxes i have the slots for the 15 inchers located horizontally below and above the drivers in each end of the baffle. On my final boxes however, I'm thinking of placing them vertical on each side of the 12" and waveguide like on the Genelec 1032.
My concern is how it will affect the sound? The bass wont get additional support from floor but what else?
Thanks for any advice!

I just completed a hot air repair to the output stage of a VX1000/1i, replaced all the outputs and the HIP2101 drivers. But I noticed the silpad making the thermal link between the output devices and the sink is some sort of putty type stuff, and normal silpad isn't thick enough to fill the gap. Anyone happen to know what this stuff is or where I can obtain some?

For sale some 10 years old transformers taken out of Sound Carrier 807PP amplifier. I contact the company in Croatia and they confirm that this transformers were hand made in this company. The company makes some verry good transformers and hiend tube amps. This pair was in amp that costs 3k euro. So they give me some data. Primary 5Kohm,secondary 4&8 ohm,40% UL taps and good up to about 60watts.Each transformer is 12x12x10cm big and 5 kg havy. Price is 50eu for bouth plus caa. 30eu shipping inside EU.

Hi...

One of my friends will give me some parts of old SA3 amplifier. Parts will include amp PCB's and PSU, unfortunately missing part is amp enclosure and heatsink with output transistors. Some modifications are existent on PCB's (he told me that) . Please help me with some guidance. I found on internet not so clear schematics of S300 (basis of SA3) and to put PCB to original state my idea is to put all new transistors and acquire all transistors for output stage.
My question is: is this transistor lineup in SA3
MPSA42 14 pcs,
MPS6571 8 pcs,
2N4250 2 pcs,
MPSA92 6pcs,
MJ15022 14 pcs,
Mj15023 14 pcs
2n3440 2 pcs
and 2N5413 2 pcs.

Thank you all for help on this technical topic...

Hi Guys,

Hope you can help with a quickie:

I'm building some big active three-way monitors and have a question regarding passive radiators: is it possible to mount the PR on the front baffle together with the bass driver? So they're basically pointing the same way?

Hope you can help a complete noob🙂

Just picked it up, no output. Found F601 Fuse blown so replaced that but same symptoms i.e. VU lights power up, but no output. No movement on meters with a known good line input and no headphone output either.

I can't hear the RY401 protection relay click and have measured voltages from the main transformer and 2 of the outputs are low (see pic). I suspect they're being dragged down but have no idea how to proceed?

The Tancinco topology was patented in 1964 (US3136766), supposedly as a way to iron out crossover notch distortion in Class B amplifiers, and I've always wanted to give the concept a try. The idea is to add a pair of diodes so that when the anode swings above B+ the screen grid is driven directly from the anode (triode mode), but when the anode swings below B+ the screen is fed directly from the B+ (pentode mode). In theory this would give you the output power of pentode mode but a somewhat lower output impedance and the softer cut-off characteristic of triodes, which should suppress crossover defects. I've always wanted to try it out, so here are some quick and dirty results.

Test circuit below using a pair of 6P14P (basically EL84s) with an 8k : 8R transformer and regulated power supplies. Tubes biased to class-AB datasheet conditions. I have presented results for ordinary pentode and triode mode too by fitting D1/D2 alone (triode), or D3/D4 alone (pentode), or all four diodes for Tancinco mode. Tested on an Audio Precision System 1, 80kHz bandwith.

Results for Tancinco are dire. The diodes introduce a painful discontinuity into the signal which is worst at small signal levels; the oscillogram shows a 1V output signal together with the distortion residual. Tancinco tested the idea with vacuum diodes which of course would soften the switching, but I still kinda doubt he actually saw his claimed benefits even in 1964. Anyway it was fun to finally try the idea out.

Output impedance:
Pentode = 43 ohms
Tancinco = 9.5 ohms
Triode = 5.1 ohms

Input level (grid to grid) for 1W output:
Pentode = 9.4Vpp
Tancinco = 15Vpp
Triode = 24Vpp

Is there a simple way to match pairs of 2SK170 low noise JFETs for use as an input differential pair

Hi, I was wondering what sorts of distortion numbers people have measured in their Class D (and related) amplifiers, measured at full power? Both DIY and commercial, in the ~300-500W range. I'm looking over the docs for my Bryston electronics and I'm thinking of an upgrade at least for the LF and LFX drivers, and Class D was my first thought. The Brystons achieve around 0.0015% across the frequency range. That's roughly -40dB (on a power scale). The Brystons (3B, 4B, 7B, from the SST 2 series) are a bit of an older design (Class AB) and I guess modern Class D/T/etc designs can be much more efficient and more linear, and I wonder what performance numbers I should be looking for when replacing them. I want something good but also something that's reasonable.

The numbers measured during final checkout as supplied on the certificate by Bryston for my units are as follows:


So I'm primarily looking to get a lower distortion unit while keeping or improving the power handling (especially on transients) and keeping or improving the noise levels.

How do those numbers compare to more modern designs? Alternatively - what would you say are the lowest distortion and noise amps you've tried that I can look up these numbers for?

One thing - I'm not really interested in talking about how you should be listening to the amplifiers rather than looking at numbers etc - I appreciate this train of thought and find it's valuable but this is not what this thread is for. I'm looking for measured numbers to have an objective comparison, long before I start doing all sorts of listening tests. Always feel free to start your own thread and I'll be glad to contribute 🙂 Thanks

I’m thinking of making Oma Mini and I’m looking for sizes , components specially the aluminum horn, I hope someone can help me here 🙂

Hi
I want to make a breuer type tonearm ,has anyone experience of using oscillating weight ball bearings from an automatic self winding watch?. I am thinking of using them for the vertical pivots and computer hard drive bearings for the horizontal at least the bottom one. Absolute novice on a mini lathe but intending 10.5" either aluminium or magnesium rod .Would be nice if someone had done this already and ironed out the issues.

Hi everyone....😉

Sorry to introduce myself so late but better late than never.

My name is Ricardo, I am totally hooked on DIY... (I know that now)....

After two years modding, I believe I have one of the best CD53 in the world.

I do know how to study and that gave me the oportunity to understand the principal basics related with analog and digital sound reproduction.

I am amazed with the quality of the information and knowledge available in this forum for free.

I am even more amazed with the unbiased unpretentious and very helpfull people I met in this Forum.

After modding the CDP, I built a Maggeddon for my TT and am waiting for my friends to develop a PSU / RIAA / PCB to upgrade my TT.

Thank you all for the precious support.

Regards

Ricardo

Early this year, I built an Aleph J. Some of you may remember helping me with that. Thanks!

A month ago, I built a Korg/B1 and use it as my pre.

This week, I completed an F6.

My speakers, for the moment are Klipsch RP600M's. They're rated at 8 ohms, but their impedance drops at lower frequencies.

The months I listened to the Aleph J, my appreciation for how it presented texture in the mids grew to a large degree. It's difficult to tell if it was me adjusting to it, but I think it benefitted from break-in. The low end was, as expected with these speakers, weak.

This is my first week with the F6. The bottom presents dynamics I'd not heard with the Aleph J, making some selections sound much more alive. I feel as if the mids don't have quite the same magic as the Aleph J, but can't tell if it's potentially my imagination or a lack of a break-in.

I don't have the room for large speakers (bummer), nor do I have the budget for anything exotic.

Is my experience with the F6 typical when compared to an Aleph J?

If the F6 is going to break in, I might be in great shape. I really think it's quite good. If not, would an F4 provide the best of both?

Does the Korg/B1 have enough gain to drive an F4? On paper, it looks like it does. If not, I can drive it with a 300B SET I built a few years ago.

Thanks in advance!

Hi,

I use a fully upgraded Wadia 9 DAC (really a fabulous DAC) that has only 3 ST optical inputs: it has no BNC nor anything else.

Actually I use a Belcanto Reflink, having a USB input, and a ST+BNC outputs.

The Reflink is the only device (that I know) having a ST output, and I am looking for something having a [BNC or TOSLINK] inputs, and a ST output.

Do you know any device that could offer a ST output ?
Thank you

Hi, im trying to finish my open baffle design. it uses FANE sovereign 12 full range driver and one AMT driver for upper frequencies reinforcement.
there is a sub woofer filling the bottom end. (it wasn't connected for the measurements)
im new to REW software and audio measurements. iv got these graphs. i think that the frq response is ok +- 5 db and the beaming frequencies from the AMT tweeter are expected.
am i correct? do i get good measurements? (EMM-6 microphone).
i reversed polarity for the AMT. it sounds better. what do you think?

I have a 10V audio source that I want to bring it to line level 1V. That’s pretty easy: I just need to use a voltage attenuator 10:1 like image below. I used resistor values of 1k ohm (R1) and 100 ohm (R2) and it worked great! BUT I have a question: what if I had used 100k and 10k ohm resistors? The ratio is still 10:1 but I think it would not produce a good result (I have no idea why I think so).

Would you please tell if I am right? If so... should I use 100 ohm (R1) and 10 ohm (R2) resistors instead of 1k ohm and 100 ohm that I am currently using? Would it be a good idea? I just want to bring this 10V speaker level to line level and so far using 1k/100 ohm worked fine (no noise, no humming...)... but I am wondering if I am doing it right... should I try different resistor values (keeping the same ratio, of course)?

Hi diyers,

I have a dared vp-845 6SN7 model which is the later version than the older 6L6 model
The amp recently stopped working. Upon inspection the tubes are lighting up but the plate voltage and bias for the driver and preamp tubes is missing. Also the PSU readings are showing a lower than expected reading normally one expects to drive an 845

I’ve been looking all over for the schematics for this amp including the power supply section.
If any of you have this to share I would be most grateful

Hi Guys,

I have a few large EI core power transformers salvaged from faulty home theatre receivers. They all seem to be a centre tapped transformer rather than dual secondaries as is my preference.

So the question is:

Can I 'cut' the 2 sides of a CT to create dual secondaries?



Its not quite clear on the photo, however the 0v clearly has 2 wires going to the 'spike'. Could I separate these 2 wires and create 2: 0v-21v-42v secondaries or would I be doing something silly that I am not forseeing?

I am looking at purchasing 2 of these drivers but am confused.
On the Peerless Tymphany website the FS is given as 78hz and vas 2.38 l.
But when I go to a website to look at the drivers the figures are all different and yet the model no is the same. I have seen the FS given as 156hz and vas as 0.7 l.
Any thoughts

For sale 4 pieces of this top pio tin foil caps 0,1uf value. They are nos in top condition. They not more in production.
Price 100eu plu shipping!! Test from enjoythemusic:
So it came to me as a pleasant surprise when the AmpOhm Tin PIO directly replaced Mundorf SIO and seemed to be sounding at least as good! From experience, I knew oil caps need to settle in the specific circuit through regular use, so I let AmpOhm play/turn-off for many days.

AmpOhm really seems to hit the right balance among PIO's. In some circuits, the wonderful Jensen copper PIO can sometimes sound a bit too refined and buttery for my tastes, but AmpOhm avoids that while still remaining smooth. Speaking of smooth, AmpOhm is smoother than the Russian K40y, which in comparison can seem to have slightly more prominent grain-structure. Mundorf SIO is not paper-in-oil, and as such, it has more top-end "air" and speed compared to most true PIO's. AmpOhm does not necessarily make you notice that extra air and sparkle up top, but it's still very extended when one listens for acoustic instruments like bells and triangles. What it has over Mundorf is higher density of tone and richness while not sounding syrupy or slow.

I still remember the first time I played Bach Cello Suites via AmpOhm. The cello sounded so magnificent I almost fell off my chair! Every detail, texture came through with power and vigor. They are much larger than similar-value Mundorf SIO, so make sure you have enough space, but they are very well-built, look beautiful, and sound even better. One caution. If you are looking to PIO's as sort of a filter to hide system faults by rounding, glossing-over, or rolling-off, these AmpOhms are not the way to go because they are still incredibly detailed and revealing of the signal chain.

Hi All...

I need a Hermes-AmaneroUSB Isolator Module and a Toslink module. If anyone has one of these please message me.

I am planning a compact three-way loudspeaker. It will have the Beyma 12LX60V2 as woofer and the FaitalPro M5N8-80 as midrange driver.

I still have a pair of RCF N350 compression drivers here. Which horn would fit in conjunction with the M5N8-80?

Many greetings,
Azrael

I have a pair of JBL VP7118's that were converted to passive (proprietary Crown amps were garbage). One has a blown JBL 2268 driver because we assumed since the sub claims 1600w/3200w power handling, that's what the driver inside was rated for....bad assumption. It's actually half that @ 800w/1600w.

Instead of just reconing the blown driver and living with the half-sized real world capability, I got to thinking if it'd be possible to "upgrade" the box with a beefier driver that would actually handle closer to the 1600w/3200w rating. The amp we're using is a Yamaha PC9501n (1400wpc @ 4ohms).

Here is the VP7118 spec sheet: https://jblpro.com/en-US/product_documents/vpsb7118dp_spec_sheet-pdf

Not a ton of info about the box itself except:

Box Construction: 5/8 in. multi-ply exterior grade Baltic birch. Internally braced. Black DuraFlex™ finish
Suspension System: 8 standard air-cargo 3 in. track and 24 M10 fittings.
Grille: 14 Gauge Black powder-coated perforated steel with foam backing.
Dimensions (H x W x D): 514.4 x 701.8 x 812.8 mm 20.25 x 27.63 x 32 in.
Net Weight: 54 kg (129 lb)

The box is roughly 200 liters ported to 40.

I was looking at B&C 18DS115 or 18DS100, but a very knowledgeable speaker builder friend recommended going with a B&C 18TBW100 because it'd be better for this size ported box. He said the DS115 has higher BL, tighter suspension, and higher moving mass than the TBW100, which isn't ideal for my purpose. But I love the high Xmax spec of the DS115 (or SW115 with less Xmax but higher Xvar).....also like the lighter weight, as we might be gigging with these (I'm an audio engineer with a commercial recording studio). Also like the higher RMS power handling, as our amp is 1400W, I'd like to be able to tell my engineers they're fine to drive the subs til they see the red clipping light start to appear, then back off a bit. If we do this on a 1400W driver, it'll be pushed to it's max....having a little headroom might be wise? It appears the DS line performs better/more efficiently at lower volumes, where the SW catches up at the top end.

But I'm a novice, and I'm being told the TBW100 is a better option overall than the DS or SW 100 or 115.

High SPL is important to me, but I'd happily trade a some SPL for better LFE and/or sound quality if I have to. Of course cost is a factor, but not a very big one.

What say you hive mind? Thanks!

B&C 18TBW100 specs from https://usspeaker.com/B&C-18TBW100-1.htm



B&C 18SW115 specs from https://usspeaker.com/B&C-18sw115-1.htm


B&C 18SW100 specs from http://www.loudspeakerdatabase.com/BC/18SW100


B&C 18DS115 specs from https://usspeaker.com/B&C-18ds115-1.htm


B&C 18DS100 specs from http://www.loudspeakerdatabase.com/BC/18SW100

I was brainstorming an idea last night about a 3 way system with 15" LF, cone mid w/ WG and HF driver mounted in a concentric configuration - see pic below.

The specific recipe is eminence kappalite 3015 LF driver, eminence beta 6A in JBL WG and selenium ST200. Thr HF driver will have the mounting flange trimmed to unshroud the mid WG and possibly serve as a quasi phase plug.

I'm I wasting my time with this idea? Then main reason for choosing this configuration is lack of baffle space.

Downsizing house so time to downsize toys. Quite a rare cart, I had while developing my range of Temaad arms, this is one I had that I developed the Temaad cartridge body cap for. It is estimated the cart has done under 150 hours.

Considered by all that this is the best sounding of the 103 range, the cap is a lot bigger than most people realise. After fitting the body cap the cart is easily comparible with my $4500 Accuohase AC-2 cart, The other big advantage is it is a medium compliance cart. Therefore will play on most of the popular medium Eff Mass arms out there, However with the body cap the cart can be played with low Eff Mass arm like a Hadcock, Technics 1200 arm or 3009 etc. Where is older design sibling the 103/103R require a high Eff Mass arm to sound at it's best. The other advantage it has over nearly all is other siblings (except the 103S) is that it has an Elliptical stylus, whereas the other 103's are conical.

The ave selling price of these on the Bay is 550, (without CAP)!! MY PRICE US$450, plus postage, TBA depending on location.

Type: MC
Output Voltage: 0,25 mV (1 kHz 50 mm/sec) Frequency Response: 20 - 65,000 Hz Output Impedance: 35 ? (20 - 20,000 Hz) Load Impedance: more than 100 ? Channels Separation: over 28 dB (1 kHz) Channel Sensitivity Difference: less than 1 dB (1 kHz) Stylus Tip: 0,1 mm special elliptical solid diamond Tracking Force: 1,5 g + 0,2 g Compliance: 12 x 10¯6 cm/dyne (100 Hz) (read 16 to 20 Euro specs) Weight: 12.5 grams with body cap

what is (0.3490658)? is it pi/9?

how does it come up with the location for driver entry point that perfectly aligns to fill in that void and get an extension to the 5×1/4?

example :

300cm or 3 m x 0.3490658
or 1/x (2.864788975)

is 104.719cm

this is (just like ) 20 degrees and 60 as radians (x100) or pi /9 and pi /3 ? weird to me, probably makes sense to people better at remembering math learned long ago🙂

and 300
-104.72 is
195.28

195.28/104.72 is

(1/x) 1.864788975

and 104.72
209.44
314.16

?? the actual pressure node locations are just outside the 300 cm pipe?

I realize this is just a bunch of numbers to some folks and it may sound stupid but there’s more to it once I get the ball rolling to explain further questions in the same area using the other side of the driver as well in a compound TL arrangement. (hopefully??)

I don’t understand where the odd harmonic interval numbers are derived from?

0.217
0.349
0.424
0.6
0.712
0.848??

Some of the stuff is probably boring or inconsequential to many but no matter what you’re helping educate people who find this fascinating(because all these numbers can be derived from the interior and dimensions of a rectangle , a very specific area inside the eGyptian pyramid but also in square roots of numbers that pivot off of the number three or 1.5 or .75… which is another can of worms maybe I can open up at some point without confusing myself or others, LOL

Looking for heatsinks in the UK

is this the best cable geometry for preamp to power amplifier?

Good day folks.
I have Citation 11 speaker switch (controller) question.

I recently purchased a Citation 12 & 13 for my son. While I was playing around prior to sending it off to a tech for cleaning and testing I found that the speaker selector on the pre-amp will not isolate the signal to the amp.
I set my patch cords from PREAMP OUTPUT 1.
As an example, switch the selector to headphones, speakers still produce the sound, place the selector on system 1 speakers work, switch select to speaker 2 and the speakers still work. There was zero difference, not even a cut out while I was switching them over.
Then I set the Citation pre-amp to selection PREAMP OUTPUT 2 and repeated the test, everything worked again without any disruption.

My tech did not see anything wrong with it and noted that that's they way they did things back then (around 1972-ish). I beg to differ and that's why I am here. I attached the op's manual to this discussion. On page 12/ 17 it speaks about the speaker switch and isolating the speakers and phones.
In short, am I looking for a new tech?
thanks in advance
Byron

Hi,
I would need advice how to lead I2S bus for 2 x DAC converter in monoaural connection. I2S input bus is isolated from xmos or fi-fo. I want the best possible quality and I'm not sure if it's better to use a parallel connection for both DACs without a driver or use driver for each DAC separately. I'm afraid about my driver, that it would degrade the I2S bus. If you recommend me a driver as a better solution, can you write me what type I should use to avoid degradation? The distance of the input connectors from the DAC is very small. I don't have enough space, so ideal SSOP package. DVDD voltage is 5V, the driver must be able to perfectly process the incoming 3.3V logic state. Or 3,3V and I will have to use another stabilizer for this driver because I don't have a 3.3V voltage on this board.

Can I use one driver for both DACs and output resistors for DAC-A and DAC-B or better for each DAC driver IC separately?

FREON TF Solvent New Old Stock
DATA CHEM 5oz net pressurized cans
Was considered the best magnetic head cleaning solvent ever produced
Will not affect plastic, high dielectric and dries residue free
Unobtainable now, have had these for a number of years
Do not have any extension tubes for these
I have 7 full cans of this solvent.
Please do not ask for a discount as I am disinclined to offer any.
I will only ship USPS services with delivery confirmation.
The least expensive being parcel post
If you want faster service ask me but will only use USPS services
1 can will ship as 1 lb and each can will add 8oz ship from 55406 zip
Payment thru PayPal is preffered (cost + shipping) +4% will ship next day
Will accept cashiers check without 4% but must clear before shipping
Will not accept any other type of payment i.e. personal check etc.
If you would like info and total cost email with subject line FREON TF,
number of cans, payment type and shipping address and type
to fwaldoch@aol.com
please be patient as I check this email only once a day

I have 244 of these that I got on Digikey. I will sell them at 50% ($3 each) off to anybody who wants them.
https://www.digikey.com/en/products/detail/ATS-55300D-C1-R0/ATS1271-ND/1284985

They are all brand new, I just cannot use them.

Hello, I have an Amp powered by a 5s 18650 battery pack (so voltage is between 21 and 14ish) connected to two nd 90 8ohm speakers(20w rms). The Amp is supposedly able to output 50w at 4ohms(I doubt it since it's propably a tpa3116 chip so around 30watt at 24v max). Should I bother to interpose this DC boost converter between the Amp and the batteries? I am not sure if it will make a difference with this speakers. I can hear the speakers clipping at full volume when the batteries are almost discharged but apart from this I doubt it will increase the output of the speakers and it will also lower the battery life because of the not perfect efficiency.

Anybody have a spare cable set of Hypex SMPS 1200 wiring? I need 2 VLP-06V sockets and 6 SVF-61T-P2.0 pins. Could be with or without cable.

Any offers?

Dear all,

I'm new to the field, and appreciate any help...thanks in advance!

I inherited old Scott S-15 speakers (29 L). Some of the drivers were dead. I was recommended to change them (I changed all 6 of them) for the following ones:

• Monacor ht-22-8
• Monacor MSH-116/4
• Audax HT240M0

I've also recaped the filters (with equivalent capacitors).
Now they sound pretty bad, and after running a quick test (using this), I am assuming the filters are not doing their job properly at all.

I've started to look into filter designing, but I must admit that I'm having a hard time figuring things out. I would very much appreciate any advice on the circuit I should build.

All the best,
Eli

I have a question about rated power in a zerer-diode coupled in the protection unit in a esl57. I have seen several units with surface mount zeners but also seen its suitable with 5W rated zeners but only 1,25W available in surface mount like BZG03C220. Will it be ok? Or problem in a clipping situation with 10*1,25W= 12,5W terminated effect in treble unit?

Hi guys,
has anyone heard this PA amp sound? it can be used in HIFI, it seems well done !!

Hi,

I have a MONO line level (signal and GND) and I want to create L and R channel from it (to output sound to 2 separate speaker) BUT I want to be able to control the volume of the R channel.

Creating L and R is pretty easy, I just connected the signal line (from the MONO line level) directly to L and R of my amp and it works fine (I connected GND too, of course). Whenever there is sound in the MONO line level, both L and R speaker play the sound perfectly.

To be able to control the R channel in order to be able to make it quieter I took a 1k potentiometer and soldered it like below:




This works fine, I mean, whenever I rotate the potentiometer, the R channel gets quieter and quieter according to the potentiometer position. HOWEVER I realized that when I rotate the potentiometer, the L channel changes a bit the sound too, not as much as R but it still gets its volume changed... How can I prevent this? How can I control R channel volume without affecting the L channel?

I've a couple of threads about Circlomanen's SLAPS in the solid state section but I hope that I can get more responses in Pass Labs section as the SLAPS concept is both simple, unconventional and also involves single hot MOSFETs. That is, it's kinda Pass Labish.

I'm very pleased with the two incarnations's performance, especially when it comes to reproduce complex music.

But there is an unexplored cross-feed configuration that I'd like some input on - the X-SLAPS suggested by Circlomanen.

This is the single 4 W SLAPS:


This is the triple 4 W SLAPS:


This is the triple 4 W X-SLAPS:


There's even a single 4 W X-SLAPS, you'll just have to imagine that M2,3,4 and 6 and their associated resistors are not there in the above schematic.

This is how the experimental hook up looks like for the triple 4 W X-SLAPS (don't bother about the steel profile across the picture, it belongs to another project of mine):


Now to my concerns about the 2nd and 3rd HD.
(Red - Single SLAPS, Teal - Triple SLAPS, Orange - Single X-SLAPS, Blue - Triple X-SLAPS)

THD:


2nd HD:


3rd HD:


The balance cancellation of the X-SLAPS improves the 2nd HD up 10 dB, but not the 3rd HD where it's only the Feed Forward Error Correction at work.
The Feed Forward Error Correction doesn't have the needed infinite inductance, so it falls off at lower frequencies.

For the Triple X-SLAPS and above 200 Hz or so, we practically are in the noise floor (Brown) due to the SMPSs and measuring environment:


Any thoughts?

This amp is similar to MTX 81000D style of amps. Its using A3120 devices near outputs. Are these optocouplers? Or IGBT output mosfet drivers?

Referencing ground;
While this particular amp has +-75vDC rail, Im unsure about the regulated side of things.

The amp has -7vDC on RCA input shields, and -9v on RCA output shields
+2.5vDC on positive speaker terminal.

Audio is not passing to terminals.

Where are the positive and negative regulators in this amp?

Some recent threads on a Swedish forum has triggered me to evolve SLAM! into a SALS?A!

SALS?A! stands for Solhaga's AMT Line Source? Absolutely! and is a 220 cm long line source made of stacked broadband AMTs.

To just stack three SLAM! in a MTMMTMMTM configuration would be very costly as there's two magnets per centimeter and each magnet is €5.

So a weaker motor with one magnets standing instead of SLAM!s double laying magnets, would yield half a magnet per centimeter.
That is, only €500 worth of magnets for a full line source sized speaker.

Here's a first draft of the motor:


Simulated:


Magnetic flux density in the gap:

The magnetic flux density is about half of what SLAM! has, but as SALS?A! is three times as long I reckon that the sensitivety will be in the same range.
That is around 100 dB/1m/1W.

I'm a little bit concerned how a MTMMTMMTM, as three stacked SLAM! would be, would be as a true line source.
Even if the M and T are playing in the range in the lower frequencies; T takes over above 10 kHz, it might be ok in the far field.

Then I was thinking of the possibility to make a broadband membrane using various pleat depths across the gap. As HEDD AMT headphone.
That gives also the possibility to use the higher magnetic flux densities in the ends of the gap, that is between 0 and 10 and betwen 50 and 60 mm.

I have 6aq5 lying around. I wonder if there is any good headphone amp design for this tube? I seen elekit have one but cant find any schematic.

Or i better go with other tube? Suggestion?

I would use transformer out as i want to drive 32ohm

Thx in advance
Erwin

Hi,
I plan a cut off circa 3000 hz, center to center of 1 to 1.2 WL. No particular high spl needed so choice is big but the monney. Passive XO so resistor attenuation. I favor passive 12 dB slope but 18 dB still possible.

Thougths :
Sb26CAC
Monacor DN25
Seas 27 no Ferro
Sb26STAC
Peereless BC25TG35
Peereless NE25-VTS
Dayton ND25FW

Have you a preferred one among these you already tested or close in price ?

Thanks if you have experience with those.

I have tried to design an i2s-swtich using SN74HC157N. I can switch between inputs but it is terrible backgound noise. So bad I can hardly hear that it is music there... As source I have Raspberry Pi and it sounds good when I connect it directly to DAC.
Originally I had the GND going thru the SN74HC157N, but I saw a switch on Internet where GND had a separate line outside the SN74HC157N so I tried that. No improvement. 🙁
I also tried to feed the SN74HC157N with battery. The Raspberry still have its original power supply but will be changed in the future. As the Raspberry sounds good when I connect it directly to the DAC I am thinking the power supply is not the issue.

On the same PCB I have the analog output from the DAC. I am thinking about trying to have separate PCBs for analog and digital. One reason is to get bigger space between the digital and analog signals, but it would also be easier to place inside my preamp. Before I order those PCBs I am asking for some advice:

EDIT:
//
I forgot to say I have seen schematics with and without 10k resistors between inputs and GND. What I have drawn is without this resistor.
//

1. Have anyone used SN74HC157N for i2s-switching audio and does it work as intended? If so, any ideas why mine sound that bad?

2. Schematics seems pretty easy, but do anyone have schematics to share? (I have made my own but it is not easy to follow from a screenshot as it is now included in bigger schematics.)

3. I am planning to have 3 inputs so I have drawn the outputs from one of the SN74HC157N to the input of the next SN74HC157N. Will it be ok to put the signal thru two SN74HC157N?

4. Any other suggested solution for switching thre i2s- input signals?

Cheers from Sweden.

I'm finishing up one of Miro's AD1862 DAC's, which runs in NOS with no filter. I am interested in tinkering with oversampling and filters from the PC side. Is anyone else doing this? What software should I be looking at?

I see HQPlayer talked about but it's a little bit cost prohibitive

I need a new Infinity EMIM for my Infinity Renaissance speaker column - not sure if its 80 or 90 as it is so long since I bought them.
Does any one know where I could get one from?

Hi all,

I have two vintage Monarch units with transformer coupled power amps and Germanium output devices. Both sound great after a recap, but different. Subjectively less bass with the receiver.

I've seen that feedback is handled differently – see the pic posted (apologies for not so great quality). Is it right to assume that the SA-500 feeds back the output signal only via an 8.2K resistor, but the SAT-360 receiver does so at two points, via a 6.8K Resistor and a 47K one? Why might this be so?

There are also other slight differences which I don't really get. But I am curious. Many thanks in advance for clarification!

Eschenborn

Since George passed on the news to loose the bypass switch on th SSE I've been wondering if an RC Snubber across the switch might correct the issue.
I was thnking of something like this:
https://www.mouser.com/ProductDetail/Nichicon/PPA2301680KJ?qs=vmHwEFxEFR81NOpAvTkPCA==

with the correct resistor might do the trick.

Patrick

Hi there,

I've designed a fist-sized programmable resistance box QR10, it can be controlled via USB-COM port or keypad.

I've thinking it for quite a long time that does it helps in audio field if I modified it as a potentiometer? Then you might control it automatically for high-end loudspeakers or as a smart gain control for op. amplifiers.

Advantages compared to the semiconductor resistor chips or mechanical potentiometers:

• High accuracy, output resistance tolerance can within 0.01%. Accuracy of potentiometer ratio can be even higher.
• TCR as low as 15ppm/25ppm so no worry about temperature drift.
• High resolution. A model that range from 1 Ω to 630 kΩ is with 0.1Ω steps, which means 6,300,000 steps in total (you can calculate how small a dB steps it could reach)
• And low cost.

Some additional work still need to be done to modify QR10 as a potentiometer. But I think it's also possible to DIY one by adding an external fix value resistor.

Any comments or suggestions will be appreciated.

Hello all,

I am looking for MJ15018/19 pair of transistors or MJ15020/21 pair of transistors. I need few of each.

If anyone have any combination of these and know that their are not fake, please contact me. I live in Europe.

Thank you,

Ivo

I have a question about matching four of the same drivers in one box [here, a vertical array]. The system they are going in is exotically expensive.

I purchased eight SATORI 6.5" MW16TX-4 TeXtreme drivers and measured the T-S parameters for each.
They of course are not matched perfectly from the dealer.

Look at the data and see if you can create a working theory for their matching.

I originally considered that Fs would be the deciding factor but when Fs is close, another parameter is not [e.g., Qts].

I plan to put four drivers in one box [two boxes for stereo of course] but I want to match them up to get the best matched performance from both boxes.

I've even created a speadsheet plotting all parameters against an average [including standard deviation] and still nothing obvious leaps out of the data.

One could just arbitrarily stick drivers 1-4 in box A and 5-8 in box B, but I'd like to take some time and use logic to match them better than just by chance.

So, what would YOU do to match these drivers and get balanced sound from both?

Pic of the measured Data for each of the eight drivers PLUS the parameter average is attached.

Hi everyone!
This is my first post requesting help... 🙁
I am looking for the datasheet of 2014 tube. I really was not able to find it in the internet!
The only information that I found from a substitution handbook is that could be a replacement of 6CL6.
Any help is welcome,
dario

As title says: What to do with a bag of 200 pcs. original Toshiba 2SJ74GR? I bought them in ca. 2010 from a renowned german distributor, being in the drawer since then and I don't remember what I once intended to do with them ...
So what besides of selling would be a good use?

I picked up an Onkyo SKW-530 subwoofer the other day, not because I wanted it myself but because I could get it cheap along with a couple other things and I intended to resell it. So, I know it's not a great sub in any case, but my problem is I'm not sure whether it's working as it should. I was cleaning it and testing it and I noticed that the output seemed much lower than I expected.

I tested it hooked to the sub out from my Marantz SR-7002, without changing any settings on the receiver that under normal use is sending signal to my Klipsch SW12.

I played an old CD that would normally have some good low-end to try it out, but wasn't hearing (or feeling, when I put my hand on it) much of anything. I turned off my L/R speakers completely (using the A/B speaker switch), and then I could hear the faintly working sub. I turned the volume on the sub unit itself all the way up and then I could hear it a bit. I went into the EQ, where Audyssey had previously lowered the sub level significantly, and raised it up to around 0 db in the EQ, and I also turned the receiver volume up. At this point I could hear the sub, but the amount of sub sounded to me like what I should be getting at 50% or lower on the volume dial, not with the sub's gain all the way up and all these other things, too.

So I turned the music off and just went into my receiver's audio setup, and I can definitely hear the test tone rumble through the sub (though nowhere near the level of the Klipsch, even with the Klipsch at 1/4 gain). That rumble does go up/down as I adjust the gain on the sub, so I know that at least that part of the sub's amp is working, and from what I can tell the actual sound reproduction sounds "normal" - not obviously distorted, not cutting in and out.

I've got audio measuring apps on my phone - if I run that test tone through the sub, is there a way for me to know roughly what level of output I "should" be getting from it? Like I said, all I really want to do is make sure it's working before I sell it to someone as a working sub. Otherwise I'll sell the driver and trash the rest, but it would be a shame if it was all working fine.

At the same time - if I have to turn the sub's gain all the way up just to hear it, something tells me it ain't working fine.

I've taken out the amplifier and had a look at all the parts. Zero expertise there AT ALL, but did not see any obvious signs like burnt components, leaking caps, loose connections, etc.

Scratching my head thinking about the best way to make a phase plug for a 12" midbass FLH. I need clean linear response from 150 to 800 hz and plan on using a tractrix profile for the low distortion to mate with a 2" coax CD crossed at 500 hz. The midbass driver is a B&C 12MH32.

Any thoughts on a simple method of making a good phase plug mold or using other method of construction?

Hi all,

Being personally fond of the sound of my UGS modules - nobody's perfect -, I wanted to test some ideas we've been talking about with my friend Philbyx, who owns the paternity of the idea 😉 His sharp neuronal connections made a link between the UGS schematic and the Zenquito, a french PP amp which seems to be quite highly appreciated from French DIYers (http://perso.orange.fr/jm.plantefeve/sche.html). So we decided to take this concept further and, stimulated by the Aleph-X brainstorming, we began to test and build something.

A word of warning here : this beast uses the now infamous dual japanese JFets 2SK389/2SJ109. Shame on me, I know... But I had some of them... What else could I do ?

So here's what I finally came upon (see attached pdf)... Nothing very fancy or new, just basic building blocks put together. The front end is a basic current-mirror ugs, with increased current output capabilities to properly drive paralleled mosfets. The current mirror transistors have been changed to handle the power requirements, and I used 2SC2911/2SA1209 for their nice characteristics.

The front end is followed by a classical Vbe multiplier for biasing a typical triple pushpull of mosfets. I'd rather see this biasing stage as a kind of shunt regulator, but that's only a matter of words 🙂 I kept the resistor network at the output of the UGS to be able to test the difference between global and local feedback (using jumpers).

The power supply is built from a 2x18V 400VA transformer (1 per channel) with a CLC filter, giving a rough 23V at rails. The front end is powered a bit higher (30V) using a voltage doubler (à la A75 😉 ), followed by discrete regulators, to provide more output swing from the front end.

The mosfets are idling at 250/300mA each for the moment, and on the bench, the max output power (before clipping) is close to 100W in 8 Ohms. More than I need 😉

I've added an on board DC protection, but I must confess I'm amazed by the offset performances : under the mV on diff offset, and under 5mV on absolute offset... 😎

It's only at prototype stage for the moment, and I began the listening sessions comparing them to my Aleph X.

How to say... In a few words, the AX is rounder, warmer, softer... The U.P. is sharper, wider, faster. The AX gives a "classical" class A feeling, with warmth and lushness, but a bit of fuzzyness. The U.P. seems a bit "cold" in comparison, but seems to be more "balanced", in the sense that it does not seem to emphasis anything. It goes higher (some veil on the AX in comparison), is faster. The bass has a better slam than the AX which seems a bit slow here. And most of all, the U.P. is awesome on the spatialization. Not that it widens everything outrageously, but it's much more accurate and refined, much less fuzzy than the AX. Every frequency seem to be much more isolated and located...

Well, I'm not good at putting it in words, but both AX and U.P. are marvelous amps, and I will have hard times to chose. I still have some test to do on the feedback path (still haven't done extensive listening on the local feedback configuration).

Some photographs of the beasts at work :

The AX/U.P. challenge


One channel :


Closeup :


Once again, a zillion thanks to Nelson Pass for guiding us to the stratosphere 😎
And warm thanks to Philbyx 😉 Feel free to post your version here 🙂
Thanks also to Marc and Ninio, for supporting this without listening 😀 You'll be rewarded 🙂

You know you are into a hobby a little too deep when you are watching tennis one evening and the only think you can think about when you see this -



is that you really need to build some speakers with some SB Satori Textreme drivers. LOL

Hello everyone,

I have this idea of making a minimalist audio interface that will have great THD+N specifications of -120dB / -117dB.

On the analog part of things I am thinking to have two input/output channels switchable between balanced/unbalanced XLR / RCA connectors. Both input channels will have Gain controls. Output channels will have Volume control (better if digital, but analog is fine too).

On the digital part of things I am thinking about USB, COAX, and maybe TOSLINK inputs. It would be nice to have the ability to switch between those with a button and three LED confirming the input.

This is the part where I lack knowledge in terms of how to make it work in a simple manner. I know of CM102 chip that allows to connect DAC chip to the PC via the USB - https://www.cmedia.com.tw/products/USB20_FULL_SPEED/CM102AplusSplus.
But it only allows 48kHz sampling rate. Also, how would it all interact between DAC chip and ADC chip at the same time? I also have seen XMOS chips used for tying USB and DAC/ADC together?

If this project is successful I plan on designing DIY friendly board and posting Gerber files as open source for diy community to enjoy.

Any help will be appreciated.

Thanks,
Pavlo

I bought some B&C 12MH32s from a private sale and have some suspicions regarding their authenticity. The thing that concerns me is the packaging, specifically the tape used to seal the box and the banding around the inside folded cardboard. I've never seen a B&C driver packaged this way so I'm a bit nervous.

The boxes do look authentic compared to other B&C drivers I've purchased. There are matching holografic stickers with part and serial numbers on the boxes and drivers. The part about the driver which worries me is the spring loaded terminals compared to the ones in the stock picture from the B&C website showing they're faston terminals. A quick Fs measurement confirms Fs do be within 3 Hz of published spec before break-in, although on the lower side of the range. A high power sine sweep sounds very good, clean and no weird noises. Without comprehensive TSP measurements I won't know for sure. I just want to hear what you guys think.

Has anyone seen B&C drivers packaged this way before? Is this possibly a fake?

I wish to revive an RCA radio from 1949 that contains a number of waxed paper capacitors. I have found film capacitors on the Digikey site for all but one value - .005mF. Inside this radio are five of them but of different voltages. One is 200 volts, two are 500 volts and two are 600 volts.

Any suggestions on what to use for substitutes? If there are .005mF capacitors still available, and are 600 volts, that should cover all of them.