Is VFet an internet influencer thing that makes no sense at all?

I admit I started to buy vfet amp years ago influenced by some "internet influencers" they make me to believe they have unique sound, like no other trasistors can compare.
So, I got Yamaha b-2, Yamaha b-1, recently a Sansui ba-1000. Sure they sound similar sweet and smooth and i am satisfied.
I also found that not all vfet amp is so sweet and smooth for example hitachi ha-500f. I haven't heard one but according to kitr's review and listened to some youtube recordings I feel it sounds a little dull maybe on par with sony ta 4650. lacks some airy and live feel of ta-8650 from videos recorded in same environment. So I feel circuit design is more important.
I wonder is that true no other transistor can achieves sound like vfet? what about mosfet, high current mosfet like uhc or lapt, SiC depletion jfet? Anyone know any non-vfet amp with similar sound?
I am also curious about how vintage vfet compare to modern lowest distortiin class ab amp(
topping la90) or a GaN fet class D?
IMG_20230213_114752.jpg
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SpicyTL - Transmission Line Simulation Model

Hi everyone,

SpicyTL is online and available for download.

SpicyTL is a simulation model, based on the electrical circuit theory, which predicts the electro-acoustic behavior of transmission line loudspeaker systems. It is the evolution of the AIRDAMP model, whose development (and the principles that allow its operation) is illustrated in the article Progettare una linea di trasmissione utilizzando SPICE (Designing a Transmission Line with SPICE), published in issues 408, 409 and 410 (April, May and June 2019) of AUDIOreview. Compared to the latter, SpicyTL has a modular structure composed of several acoustic blocks and can also simulate tapered and expanding TL's.

The model, particularly suitable to design a TL damped with polyurethane foam, allows to simulate frequency response, phase response, group delay, electrical impedance, speaker cone excursion and air velocity at the TL output.

Operating instruction and download at the following address:

SpicyTL-english – Transmission Line Speakers

An in-depth look at the software, and its use for designing the Ikigai monitors, can be found in the article Progettiamo una linea di trasmissione con SpicyTL (Designing a Transmission Line with SpicyTL) on the 427th issue (January 2021) of AUDIOreview.

Kind regards,
Andrea

I have chosen to restrict access to transmissionlinespeakers.com for users from Israel. This decision is a response to the current events and human rights violations in Palestine. I strongly oppose any form of violence and injustice, and I believe this measure is necessary to express my ethical dissent.
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Yet another Volume controlers and source selections

I will post more projects that was made by me :
1. Volume controller / source selector for motor potentiometer ( like Alps blue velvet ) :
-schematics
-avr atmega8515 soft
-pcb
On atmega8515 you have to write only the fuse-bit named :
" Internal RC oscillator 8Mhz "

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Quad 99 Pre > CDP Quadlink - Only Standby Pin Required

Hello all

Apologies if this is in the wrong section.

I have a Quad 99 pre amp and 99 CDP. I found it sounds best using RCA interconnects vs the Quadlink cable. Of course this disables the standby mode function, which is now missed.

Has anyone tried to alter a DB 15 cable so that only the necessary pin for bring in or out of standby is connected?

I've seen a few forum posts that state pin 7 is responsible for this function between the 99Pre and the 909 (but that would be the Ampbus connection), I need the pinout for the pre>CDP (Quadlink not Ampbus)

99 pre pinout

Image 002.png



Does anyone know which pin i should be targeting?

Thanks in advance!

1100a² #2

One channel of amp distorts when the volume is turned up. It's goes from clean to distorted around ½ power.

The voltages across r206, 207, and 208 are identical to the voltages across r306, 307 and 308.

Audio remains clean at c304 a and c201

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XTC, PolkSDA-style, up-cycled, full-range build tips.

I am looking to build a new arrangement into existing pair of gifted cabinets of about 6-8ltrs. I want to run two identical (ideally 4 inch, wideband) drivers on each side in a (polk SDA-esque) crosstalk reduction arrangement. eg; LEFT; R-+, L+-. RIGHT; R+-, L-+ The drivers will be about 8-inches apart, horizontally, in each enclosure.
I have this arrangement set up in my room with 4 separate speakers and I love it. But the logistics of this are cumbersome, to say the least.
I will be running a sub from 100-150hz
I'm posting here simply to get tips and suggestions as I go through adapting the cabinets.
Any recommendations for good and affordable (ideally 4" to approximately fit the current cutouts) wideband drivers, bearing in mind cost of buying 4x
Also, my current set up uses two amps to achieve the "effect". Amp suggestions would also be useful.
I'm not particularly technical in terms of electronics/physics language.
I'm not particularly interested in a debate about the merits of xtc approach. Just hoping for experiences and tips.

Melted speaker?

Blaupunkt 5x8 voice coils melting tweeter post? I would think the coil would burn up before melting something. I measured the output for DC and it had .5v offset. It's a Chinese head unit with a (marked) TDA 75xx class AB chip. something strange going on here. Any ideas?

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An attempt to upgrade my push-pull VAS amp

I am now embarking on a perilous journey on this DIY audio hobby. I am pushing the limits of my audio knowledge acquired here at DIYAudio. This project is an attempt to repurpose my pcb board intended for a CFP amp. LTSpice says the schematic will work, that is according to my understanding of how the circuit behaves.
The push pull VAS concept was taken from Keantoken's Aurum X amp with the output topology also being taken from Roender's FC-100 amp.
I ask you if anyone has made a similar design please comment your feedback. If this amp will turn out to be a firestarter 😄 I will not hesitate to toss the boards in the trash bin.

Thanks y'all!

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Hello World

Hello fellow diyadio members🙂
My name is Lew, I am a student of engineering sciences in Germany. Normally I am hesitant to post online and like to just read along, but in this case I decided, after a few years of "passive membership" to come out of the shadows and to become part of this amazing community.
So far I have built four pairs of speakers, for friends, family and myself and two class D amplifiers out of modules. Now I would like to transition into class A amplifiers by first using one of the provided schematics from this forum. In the future I would like to be able to design something not terrible myself. Thanks in advance for helping me on my journey😉

ICEpower1200AS2 single ended subwoofer amp with miniDSP 2X4

I’ve built Ripols with 2 x 12” 8 ohm Dayton woofers. I believe that 1200AS2 should give enough power into 4 ohm. I want to use Mini DSP 2X4 with them as I have not built the crossovers. I have a balanced preamp and I want to implement the following setup:

Pre->MiniDSP-> 1200AS2-> Ripols.

I’m not sure if I can run 1200AS2 with single ended input, also, 2X4 mini DSP has 0.9 V output which is not enough for 1200 AS. I can get 2X4 mini DSP balanced that will solve the problem with the input but it’s output is only 2V so it is probably still not enough. Are there inexpensive buffers that I can use with ICE1200AS ?

7x HT12/HT12+ fully assembled $2359.54

HT12 is epic, but sold out!
https://www.diysoundgroup.com/ht-12-kit.html

Why wait for them to be back in stock when you can buy a set of seven fully assembled!

Look at how the HTM12 measure!
https://www.erinsaudiocorner.com/loudspeakers/diysg_htm12v2/

Now, HTM12 will be slightly different from HT12, HT12 is a cheaper model.
Most of the offaxis should be very similar between the two though.
Also, two of these are "HT12" kits with crossovers built by me.
The other 5 are "HT12+" with improved (much higher component count) crossovers purchased fully built by Matt Grant, mtg90.
If you have Dirac or Audyssey the improved crossover is a bit overkill, but wanted the best starting response possible. \
If you want, individual frequency traces flat on its back on my deck are available.

Why am I selling? Because I am a madman and built 6x 8' tall line array of ND91 drivers.

Hoping for a local sale in the San Francisco Bay Area, technically, I am located in Marin, near the Larkspur Equator Coffee.

Cost is $2500, equal to all the kits, crossovers, screws, terminal plates and wire.

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DAC-END 24 (AD1865 NOS DAC )

DAC-END 24 (AD1865 NOS DAC )

AD1865 is a dac chips with natural sound and inspiring.

ago, Andrea and I had success with DAC-end, now I have upgraded dac-end 16-bit, 24-bit up-end DAC.

it receives the input frequency of 192 KhZ- 24bit and special use Amanero to use USB audio source from your computer.

invited to view some links:
DAC End - the AD1865N-K with single ended vacuum output stage
http://www.diyaudio.com/forums/quan...287-dac-end-best-dac-ad1865-updat-2011-a.html
DAC End 2 - the AD1865N-K with single ended vacuum output stage


DESRICRIPTION DAC-24

NOS DAC-AD1865- 192Khz
Quanghao-Andrea

Description:
DAC AD1865 unit accept 192KHz input with operate NOS (non-oversampling) mode.

  • Structure:
Digital input:
1 x coaxial input,
1 x Optical
1 x AES/EBU- Balance in put
1 x USB Amanero
1 x selected in put
1x AD1865
1x Cs8416
1x HQ-01 Transfomer supply: 0-6V*2
Analog output : R IV passive
Digital section : using AD1865 as digital receiver CS8416 + Amanero USB.

DAC module assembled and tested with all components


  • DAC kit Design construction:
1. NOS (non-oversampling) application
2. 3 x low noise, low Z-out shunt regulator
3. 2 x LT regulator for Digital + USB
4. 4 stages regulation for digital section
6. Audio grade components were used.
7. OSCON capacitors used for digital section
8. The composition resistors Vishay error of 1%,
9. 1 x Transfomer supply: 0-6V *2

PCB size 216 mm x 110 mm, 1.6 mm double sides board with gold plated.

Optional: You are welcome to ask if you have any question on the assembly and more information.
Shall not be reproduced without consent!

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Reverse Engineering 4-way Speakers

Hi everyone.

I was lucky to score a nice external crossover from Accuphase with fixed crossover frequencies. The crossover is equipped with modules to feed a a 4-way design. 3 frequencies modules and 4 line preamp modules. I have 4 stereo outputs.

The frequencies are 130, 350 and 5000Hz.

I am struggling to understand the configuration of drivers for such frequencies. I am not sure if I am willing to build a 4 ways speakers as I just finished a 3 ways speakers with digital crossover using minidsp. But I am curious.

Does everyone have an idea of a typical design with those frequencies? Typical driver size and configuration. Any compatible speakers where I could bypass the internal crossover and feed directly the drivers?

I asked the previous owner. No response yet.

Thanks!

What to do with Altec 420A bi-flex speakers?

Hello,

I have a pair of Altec 420A bi-flex speakers that I would love to use in a project.

They are the black-framed, "16 inch" type - refer picture. I think they are the same as the relatively common 15" white/silver 820A / 820-8As. I know they are different to the 820B.

These things have a huge Vas (~470L) and, unfortunately, large enclosures are not an option for me.

I was hoping to for some ideas as to what to do with them.

I was thinking (/hoping) that some form of open baffle or smallish ported enclosure might be acceptable. Ideally two-way with the 820A on bass and mid duty.

Any guidance appreciated.

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Help!! is my WinISD response any good? need advice from pros

Hey guys! i recently decided to build a new trunk enclosure for my 2 " Pioneer TS-W304R" powered by my brand new "AudioControl EPIC1000" amp and decided to use WinISD to try and get the most out of my enclosure before i go through all the work to build it, ill attach the screenshots below. i watched a few videos on how to use WinISD but im not quite sure on how to optimize it. thanks guys! let me know if you need any more info, im very beginner, any advice helps!

car audio 1.PNG


car audio 2.PNG

SL1200 M5G Ti arm tube swap

Getting prepared to do an arm tube swap with Cardas litz wire on one of my 1200s using the titanium arm tube from an M5G model. I've done this swap before, but this time I want to add some very fine lambs wool wrap around the wire.

I notice with some of the rewired arms I've done previously the top end took a slight hit in overall detail and the midrange sounded a bit peaky. I would attribute this to the stock alu arm being more lively and resonance prone in the HF.

I have an SME109 arm on one of my 1200s which was upgraded with ceramic bearings, but otherwise stock aside from wiring. This arm sounds very balanced and detailed with a VDH3 MC cart but it doesn't have the bass extension of the M5G arm with an AT-OC9/2 cart.

So my ultimate goal here is to have the new M5G arm setup along with a fluid dampener and cardas wire, so I can sell the SME along with its mounting base or use it on another custom TT built from a Corian plinth with SL1200 drive. On this one I'd like to use my magnetic bearing setup with the dynamic height adjustment. This would be the test bed for the bearing setup.

Has anyone done anything similar to their SL1200 using the magnesium or titanium arm tube? I also have a graphite powder resin composite coated arm tube which weighs about 0.3 g more than stock. This one also sounds very good but also has duller top end.

Having big trouble with my dual 15" Ripole design

Hi Everyone,

Edit: Issue resolved. My amplifier was nowhere near sufficient enough to drive my subwoofers adequately. After strapping a Crown XLS2500 across the drivers, things are great. Cheers!



I built a dual 15" Ripole subwoofer using Dayton RSS390HF-4's. Everything works great as expected until transient bass notes hit, then it sounds similar to the suspension/coil bottoming out. The issue is that I hear this when there's only 3-4mm of excursion. My ignorance on the physics of this exact topic leaves me just with some bad feelings, as if the suspension is just not okay in this application, but I'm having a hard time identifying which parameters I should be looking at. I've measured the THD to see if there's anything obvious going on, like resonances from the slots, or significant dominance of even vs odd harmonics....etc. Nothing pops out. The THD plots are clean at 95dB 1m, generally less than 1% in the sub range.

Here's a recording of the sound.
Recording of "blatt, blatt, blatt" sound

I'm wondering if there's a slight imbalance of pressure on the cone from the cavity geometry, since the backwall of each cavity is curved (to reduce the Q of the 1/4 wave resonances), which may put an uneven load on the cone, which might be able to "rock" the cone to one side during its stroke, and maybe the coil is actually rubbing on the magnet / pole piece during these louder, transient signals? I'm somewhat convicted in this belief at the moment since the driver in free-air excurts significantly more before I start to hear the suspension doing suspension-y things.

This is the impedance sweep of the ripole.
1736875696685.png


And this is the THD measurement @ 1m of the Ripole, DSP already applied (dipole correction boost, cavity resonances reduced, 20Hz and 105Hz steep crossovers, room correction
)
1736875936608.png


And here's the THD measurement of the full integrated system measured at 2.5m, listening position:
1736876301061.png


I did a quick experiment. I "sealed" the front cavity on carpet by rotating the enclosure 90deg, leaving only the dual rear cavities open, the output changes to that of a monopole, and is expectedly more efficient. It gives the sound you would expect of a large driver in an undersized enclosure. But the strange suspension sound diminishes quite a bit, if not completely. However, since I don't have a way to directly measure the "sound" of the suspension / coil noise, I suspect I'm just experiencing a higher "signal to noise" ratio, given the expected efficiency boost of no longer being a dipole flapper.

Anyways, I am curious if others are having this issue. I am looking for help identifying root cause / physics of the issue. I would also love to be able to better select a subwoofer driver for this application. I followed all the info I knew at the time of purchase for this application, aiming for a final Qts of 0.7 in enclosure, choosing a driver with low distortion motor desgin, high excursion, and designing the enclosure cavity sizes scaled up from my previous build, where I used 4x Peerless SLS 10" subs, chambered. (they sounded good)

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A Big'un - the Audio Nirvana Super 15

I decided it was time to open a thread to capture my experiences with this driver and the box I'm going to make for it.

It's a big driver. Until you have one in your hands, it's hard to imagine.

More about the driver here
Edit: the ferrite version is no longer available, I recommend the Super 12” Ferrite as the affordable alternative to the 15” Alnico. Also, I wouldn’t go smaller than 10”, where the Super 10 Alnico is my recommendation as the preferred option for a vented box design enclosure when smaller size is a priority.


Edit: this design eventually became an open back box with a light level of stuffing behind the driver. It's not an open baffle dipole in the traditional sense because the highs are mostly radiated from the front whizzer cone and the stuffing absorbs these frequencies behind the driver. The bass however is able to radiate backwards from what is essentially a U-frame, perhaps tending towards cardioid.

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  • Locked
SanKen Transistors..

I have various SanKen transistors as follows :

2SA1215 X2
2SA1216 X2
2SC2921 X2
2SC2922 X2

I bought these a few years ago to put in my Sony 550ES but never got round to it. The Sony has now gone and these are no use to me.

The 1215's and 2921's I believe are genuine as they weigh 18-19 grams each, which is what the SanKen datasheet says.

£8 per pair

The other 2 pairs are lighter so may be fakes. I haven't looked into it much further than a quick Google and I'm not knowledgeable on the subject.

£4 per pair

Free postage in UK PXL_20250116_211510478.jpgPXL_20250116_211559054.jpgPXL_20250116_211604302.jpgPXL_20250116_211613335.jpgPXL_20250116_211617214.jpgPXL_20250116_211731045.jpgPXL_20250116_211757099.jpgPXL_20250116_211810598.jpg

Stax Pro voltage

Need a little help here.
To cut a long story short.
A friend asked me to look at his headphone gear. Stax 007 and SRM-T1S headphone amp. The story is, that this
never worked as expected. And he´s right. The sound is dull, boring and nowhere near, what you would expect from STAX.
He hiself had replaced the tubes and all the elctrolytic cap´s. This made no difference.
Visited another firend, who has a set of Lambda phones and an older solid state Stax headphone amp.
On his setup, the 007´s sound perfect. A slight more laid back than the Lambda´s but totally acceptable. On the SRM-T1S far from as good.
So I´m confident, the error must be in the SRM-T1S
Can´t see or measure any faults in the circuit........... other than maybe this one..........
People write, that the bias voltage is 230V for normal and 580V for Pro.
Where to measure this??
No matter if I measure from L-, R-, L+, L- or ground to bias in the plug I end up with approx. 150V in the normal plug or 360V on the Pro.
I called the Lambda guy and asked him to measure his amp, and he got the same results....... 160V vs. 355V.
Am I on thin ice here, or is this measurement taken between "bias" and some unknown ref.??
I only have a really bad copy of the schematics, so if anyone can scan the org. service manual from Stax, it would be very much appreciated.
I found the adjustments on Headfi.org and adjustments are done within less than 1 Volt (Stax acceps +-15 or less).
Thanks in advance for any info...... and no...... I´m not gonna convert it according to the CCS mod´s 😉

"Haudegen": MM-Phono-EQ

Why not be bold again and simply reach into the component box?

No weeks of thinking, just have fun like a child. Just like in the old days, simply tinkering and listening to music. It's not quite that simple, because there's so much at the back of your mind ...

Design frame:
-Two-stage
-Voltage divider that bends back the cutting characteristic
-Gain about 40dB at signal frequency 1kHz
-One coupling capacitor at the input and one at the output
-Rebuildable, if it works
quite simple (KISS) and the hope that it doesn't noise - and above all works!

This problem has kept me up all night, being a 19 year old again. It's not easy to throw everything overboard and deliberately break some design rules.


Today:
I only listen via MCs, but if it works, I'll buy the nicest and cheapest MM my store has in stock.

Here's my idea, the simulation results are perfectly fine - if I haven't made a stupid mistake.
You never know.

😉

Always be gentle,
HBt.

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ClarityCap PUR/PUR-S series

Hi guys, I would like to know if anyone here tried the ClarityCap PUR version capacitors? How would you rate the PUR against the Jantzen Silver Z-Cap or Jantzen SuperiorZ-Cap?

Also there is a PUR-S version, is there any noticeable difference from the PUR version? Could not find any review information on these PUR version from the net, so any input is appreciated, thanks.

Dröttnar- A SEAS U18RNX/P and 27TAC/GB 2way with exceptional clarity...

In the Spring, I finished the Missing Link boxes for the InDIYana 2022 xover theme with the added objective of using it for more upcoming 7" 2way type build stand-mount projects.
Most of you have likely seen the first use with the SB29RD(A)C and Dynavox LW6004HRN. This is the second one, using the Seas U18 and 27TAC Titan tweeter. The 50 Hz tuning and stuffed 15 ltr box was good for the U18. If I tuned lower, it would sound a bit anemic, but the highish tuning used supported the bottom end fairly well.

I used the first of my 2 pair of Titans in the EMP build recently in the last couple years, and this is now the second implementation of a tweeter I really enjoy a lot. What is posted here for FR and HD is prior the addition of a 1 ohm resistor after the xover on the tweeter to adjust to taste. It was flat before, but just a hair bright, so just eyeball about a -1dB tilt above 2.5k.
The xover ended up being LR2 at 2.5kHz, which is one of only a couple times I've managed that result. Because of this, the toe-in needs to be just a bit more than normal to get the image right and not shift when moving my head. I literally can look down the outer left and outer right sides of the respective channels when seated in the right spot. To get the drivers to comply at LR2 took a single coil on the woofer and an overdamped 3rd order on the tweeter with an additional Fs LCR compensation. The TAC has no ferroFluid, and this is a likely compensation circuit to add in most cases.
Some funny/interesting things that occurred over the course include;
  • I heard more details in some tracks I know really well, and that is saying something. This is a VERY clean design. It has lower HD than a lot of my other builds, dominantly -50dB for the most part.
  • I had some sssibilance initially when I added the second resistor, which was a MOX Lynk type. I changed to a sandcast, and the problem subsided.
  • I have some very small Russian polystyrene caps of 0.002uF. I paralleled the tweeter caps with them just to hear the results and had no expectations. They are very small in value. Well- I removed them after a first listen as the tweeters sounded broken, disjointed, and distorted badly. I really thought I might have had a bad tweeter because of what I was hearing. After removal, the relaxed and detailed sound that I heard prior returned as I had left it. I'm not really sure what caused the bad interaction, but it was awful!! I am not a supporter of bypassing for the most part, but since I had them and they were doing nothing I gave them a shot.

Listening impressions? I feel they are really revealing without being harsh. That said though,they are okay with rock, jazz, country, orchestral, classical, etc; but I would not recommend them as metal speakers. Being that they reveal the flaws, the source material really needs to be clean to sound right. Those of us Metalheads know that over compressed and gritty can be quite common for the bad mixes. With well recorded metal, I would still think they'll sound quite good. Nightwish's 'Dark Passion Play' was fine for example, but Born of Osiris was not.

For the parts used, in case you're curious, I used ASC X387 poly-in-oil caps for the tweeter, and I hand wound a 16AWG air-core for the woofer that I triple-dipped in polyurethane. Sandcast resistors used throughout.

Thanks for reading my long winded intro to my 'King' Norwegian design.
Wolf

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For Sale ICEpower 80am2 LOT and other interesting stuff.

Hi,

I have for sale a lot of stuff I want to get rid off.

I sale all as one lot, as I don't have the time to ship each part in its own.

So, whats included??

-. 24x ICEpower 80AM2 amps plus assorted cable set.
-. 4x untested Hypex UcD180
-. 42x PCM 5242 I2S DAC boards (see pictures for specs)
-. 1x ADAU1452 DSP (the one sold in Aliexpress por USD$70 aprox.) with the proper AD1938 CODEC board.
-. 2x extra AD1938 CODEC boards (see pictures for specs)
-. 1x USB to I2S 8 channel board
-. 1x SIGMAlink Programmer

Price for everything: USD$250 80 plus shipping. Depending on the country, shipping thru FEDEX would be around USD$50-70 80 extra.

Feel free to ask any questions.

Cheers.

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What's a better way to split DAC signal?

I'm building a PCM1794 DAC, planing to have line-out, headphone amp, maybe fully-differential low-pass filter for balanced output, with line-out it's just datasheet low-pass filter, but headphone amp I'm planing to use TPA6120, which requires differential input, would it be a good idea to split the signal after I/V stage? if current output isn't an issue(using opa1612), one goes to opa1612 as in datasheet, one goes to TPA6120 without buffer?

what's normal way to handle multiple output?
thank you all

IV stage without buffer.jpg
tpa6120.jpg

WTB Locking for pushbutton switches for an old Marantz Receiver

I'm looking for three of these switches.
These switches are installed in an old Marantz receiver, originally from the company Toko, which has now become C&K.
I read in a forum that the C&K type F4UEE should fit - but I can't judge that.

Does anyone happen to have such 12-pin versions to give away? I would need three of them.

If you'd rather like to swap your switches, I have some good components to dive into.

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Help in choosing a potentiometer as a "Passive preamplifier"

Help in choosing a potentiometer as a "Passive preamplifier"
============================================================

This is more of a guide as I already found the answers I need. To begin a potentiometer as passive preamplifier does not amplify anything. It is just a potentiometer used as a potential divider in a box between the source and the amplfier. It is really just an attenuator but becuase of its position and the box it is called a passive preamplifier.

You might think, well I have a single high voltage source with 2.1 Volts, no interest in microphones, vinyl, balance or (dread) tone controls - so I think I should just use a 10K logarithmic potentiometer to get a nice pure sound, because after all - less stuff in the signal path must be better than more stuff.

But if you try it, you may not get the sound you expect - it might exceed your expectations but could possibly sound terrible. If you ask about it, you are met with claims of high frequency attenuation, lack of dynamics, lack of top end, lack of bottom end, lack of sleep!, balance problems, changes in the sound as the volume is changed, too low volume at max and that by violating a 5 times or 10 times voodoo source and load impedance rule can make the sound gods angry and get you punished with bad sound.

Before I dispel some of these claims and explain the validity of the rest let's get some terms out of the way and see what a preamplifier actually does.

When you see the word impedance it is usually accompanied by one of the words input, output, load or source. In the case of a preamplifier the input impedance refers to the inputs where you connect your sources, for example your CD player. The output impedance refers to where you connect your amplifier.

Great now what is the load and what is the source?

Looking at the chain left to right CD > Preamplifier > Amplifier, you can say that the source impedance is the output impedance of the box to the left and the load impedance is the input impedance of the box to the right. But you have to qualify this with whose source or whose load you are actually referring to.

So the source impedance of the preamplifier is the output impedance of the CD player and the source impedance of the amplifier is the output impedance of the preamplifier.

A preamplifier serves several purposes:
1 It amplifies low voltage sources such as microphones or record players
2 It fixes that audio curve thing for record players
3 It provides a balance and (dread) tone control
4 It provides low output impedance
5 It provides a high input impedance
6 And it looks nice and impressive, gives a warm and fuzzy feeling inside, has a headphone output and a remote control too if you are lucky then some blue LEDs....

1 Low voltage sources must be amplified before sending the signal to the amp, otherwise they may not be loud enough. An amplifier has an input sensitivity value in Volts. A signal with this voltage will be amplified to the limits of the amplifier i.e. the rated power in Watts.
2 Look elsewhere for an explanation
3 I try to sit in the middle anyway and my sources are usually have a good balance,
4 The output impedance refers to the ability to drive a heavy load. A heavy load is one with a small impedance. The smaller this figure the better. More on this later.
5 The input impedance refers to the load imposed on your source by the preamplifier. The impedance should be a high as possible (= a light load) so it does stress on your source too much.
6 Look elsewhere for a warm and fuzzy feeling inside.


High frequency attenuation
Capacitance in the interconnect cable combined with the output resistance of the potentiometer make a low pass filter. There is a good explanation on the DACT website in the technical section two PDFs and also an Excel file to calculate it. This can be a problem if the output resistance of the potentiometer is high and or if your interconnect cables have a high capacitance and or are very long. If you choose a 100K potentiometer and 10 meters of high capacitance interconnect cable then you will get some high frequency attenuation and hear it too. If you have a 10K potentiometer and a short run of interconnect cable then you won't. This low pass filter problem does not affect preamplifiers as they have a very low usually 50-100 Ohm output impedance. With preamplifiers you can use meters and meters of interconnect cable and not fear this high frequency attenuation low pass filter problem. My L-25D amplifier has a 1 nano Farad capacitor connected to signal and ground as part of an optional RF Filter and the capacitance of the RCA interconnect cable is probably around 300 pico Farads. Plugging these values into the DACT Excel file shows that capacitance is not a problem. Be sure to read the PDF which describes how to use the DACT excel file.

Apparent lack of dynamics

Well if you are experiencing the attenuation above that would go some way to explaining the lack of dynamics. The rest of the way is found by calculating the load imposed on the source by your potentiometer and the input impedance of your amplifier in parallel. Let's look at how a potentiometer is wired and do the maths.

Figure01 shows how a potentionmeter is wired to make a passive preamplifier. The ground is the outside of the RCA interconnect cable and is usually connected to one of the outside pins on the potentiometer. The inside conductor in the RCA interconnect cable is the signal and is usaully connected to the other outside pin of the potentiometer. The signal for the amplfier is usually taken from the middle pin of the potentiometer. This is the wiper part of the potentiometer and its position determines the values of R1 and R2 in Figure01. The formula on Figure01 is used to calculate the output voltage from the potentiometer. This formula is used to calculate the volume curves seen in the figures of graphs seen later. Figure08 is a LTSpice file showing the simulated circuit. R1 and R2 are represented here as variables with a step to simulate the rotation of the potentiometer. Simulate the circuit using a 300K RLoad and you see that the VOut becomes very linear.

Figure03 shows that the maximum impedance seen by the source (the load imposed by the potentiometer and amplifier) is the impedance of the 10000 Ohm potentiometer. This occurs at the lowest volume setting -infinite dB.

Figure02 shows that the minimum impedance seen the by source is the impedance of the potentiometer and the amplifier in parallel which is calculated 1 / (1/Resistance of potentiometer+ 1/Impedance of amp) = 3230 Ohms. This occurs at the highest volume setting -0 dB. This is a rather heavy load for your source to deal with.

Well not all sources are equal. My DAC has a bunch of NE5532 opamp chips in its output stage. These chips are rated to drive audio loads all the way down to 500 Ohms. One review tortured the output stage with a 600 Ohm load and it performance was barely affected. Your mileage may vary - it is unfortunate to say and I hate to say it depends, but it does. Some sources have highly integrated chips that also incorporate the output stage opamps and a bunch of other stuff into one chip. These are the bad guys that you want to avoid if you venture into the world of passive preamplifiers. They are not designed to drive heavy loads and likely will cause you problems.

So now you should be thinking: high Ohm potentiometer = potential low pass filter problems, low ohm potentiometer = too heavy a load for the source.

Balance problems
These are caused by poor matching of the two traces in the potentiometer for each channel. This effect is magnified in the low volume area of the potentiometer where the traces approach 0 Ohms. Here it is very difficult for the manufacturers to get good matching. The matching varies with potentiometer position. This is not really a problem with good quality pots, but might still be a problem if you have a low input sensitivity high voltage gain amplifier with high sensitivity speakers. Good quality potentiometers from TKD have a 0.1 dB usual to 0.3 dB maximum deviation for a stereo potentiometer. This corresponds to a 1 - 3.5% deviation in the levels between channels. Good luck hearing that difference.

Changes in the sound as the volume is changed

This is due one or two of the two problems explained above. It could be that at high volumes, you are stressing the output stage of your source too much. Also if you have a symmetrical source the balance problems are magnified. If you are worried about your four trace potentiometer for your symmetrical source then used a stepped type with make before break resistors.

Too low a volume at max

This is a problem of a source which has an output voltage which is too low or the input sensitivity of the amp is too high or both. Not really a problem for me with my DAC as a source with 2.1 V and an amplifier with an input sensitivity of 1.6 V

Hour of power
The next problem is not listed above but I will explain it anyway. It is the effect of turning the volume control and it stays really quiet until close to the end where all the power seems to be in the last "hour" of the volume control between 4 and 5 o'clock. The problem is the input impedance of your amplifier in relation to the value of the potentiometer and how this affects the logarithmic(ness) of the curve of the voltage output from the potential divider. Using the Excel file in the attached zip you can plug in some values and understand the problem better than I can explain in text how it manifests itself. The Excel file will show you the minimum and maximum load imposed on the source.

Figure04 shows my system, details of which can be gleaned from this text or seen in Figure01. The output Voltage of the potentiometer is on the left and the -decibels(attenuation) on the bottom. The red line represents the ideal curve and the blue line is the actual curve.

Let's try a 100 KOhm potentiometer instead in Figure05. That is just terrible, all the power is in the last hour as I mentioned above.

Let's leave my potentiometer as it is and increase the input impedance of my amplifier from 3.23 KOhms to 100 KOhms in Figure06. Much better, but I don't feel like buying another amplifier with higher input impedance and I cannot modify my amplfier to have a higher input impedance.

Let's leave the amplifier as it is and change the potentiometer to 1 KOhm in Figure07. That looks much better than my 10 KOhm potentiometer, but now I am possibly stressing my source with 764 Ohm load at maximum volume(formula is above) Most sources will have given up at this stage and be on their knees begging for mercy.

For my system a 5 KOhm potentiometer would have been better, but the 10 KOhm is quite acceptable. A 50K potentiometer or heaven forbid a 100 KOhm potentiometer would be no fun at all with the capacitance problem and the last hour of power problem explained above. With a input sensitivity of 1.6 V and a maximum power of 350 watts @40hms, a source with 2.1 V and 200 Watt 4 Ohm speakers - my maximum power is achieved @ 0.7 Volts which is at the -4.5db mark. The perfect result here would have been -9dB.

The 10 times or 5 times myth
Well this is just a general rule to keep the load impedance high - there is no real maths in there. If you read the above then you know already keep the load impedance high so as not to stress your source. You might get claims of: well your potentiometer has a output impedance of 10 KOhm and your amp has an input impedance of 20 KOhm and that this is bad because it breaks the rule!!. These people don't understand that the potentiometer is not a source. The potentiometer and the amplifier ARE THE LOAD.


I hope my little rant here is helpful to someone. Pointing out mistakes I made is welcomed, pot shots (pun intended) and hit and runs are not welcome. Claims of: "Well I tried one and it sounded terrible" without any details of the components involved and their input and output impedances are not welcome. If you read the above then you will now know that you cannot just drop in a passive preamplifier into any system and expect good results!! I tried to deal with all the factors involved and did not intend to scare people off. Armed with this knowledge you can put togehter a system with a passive preamplifier and get better sound than any preamplifier.

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Stereosystem compensation för stereo loudspeakers

Because of the distance between your ears , the frequency response when listening to two loudspeakers in stereo should not be flat .

The Harman way of doing this is to simply ignore that there’s always two stereo speakers at normal listening to music , and just concentrate on one speaker in mono - with a flat response. This is correct when you listen to only one speaker in mono.

But what about stereo ?

Below you see two frequency response curves . The curve at the bottom is the ideal frequency response according to the research from Harman , without stereo compensation at listening position 3-4 meters away. We see the boost of bass needed below 100 Hz .

The upper curve in blue shows measurements 1 meters away from one stereosystem compensated loudspeaker . We see a peak centered at 1,7 kHz , Q=3. And a dip at 3,5 kHz .

If using two such speakers in a listening triangle L - R - listening position , the sound with good stereo material will be holografic , free from each speaker and very natural/real sounding .
This can easily be done using a dsp with PEQ function .

IMG_0844.jpeg
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Differential Probe - Reverese Engineered

Hello,

I use a number of Differential Probes at work (like the TesTec SI9000-1 found in RS). The need to calibrate & repair them necessitated a reverse engineering exercise. Not at all easy, all ICs (and some of the 'transistors') have had their markings sanded off. However, some numbers could be made out.

I have to hand 4 of these probes, the earlier version uses through-hole components and the later models use surface mount components. Both types are revealed here:

Through-hole version of the probe:


Notes:
Guessed op-amp used for U3
Q3=Q5 (markings sanded off). Guessed this device as TL431
Q4 (markings sanded off). Guessed at BF451

Surface-mount version of the probe:


Notes:
U3 (markings sanded off) - can just about make out '74' so guessed at 741
Missing: SMPS chip. Could not make a guess at that!

------

Anyway, comments, discussion, errors, improvements etc. PLEASE! 🙂

Andy

The SAMBUCO tonearm is may baby. It's a few years that I concentrated my work on tonearms.....

THe inspiration came from reading a few books reading a few books about the design of musical instruments, particularly those made from wood. There is a similarity between tonearms and violines: You want to avoid high Q resonances, or "Wolf-TOnes" in the violine language. THerefore I tested the design principles on wooden pipes. Funny detail: It turned out that my SAMBUCO tonearm is made from elderwood, the same as Harry Potter's magic wand.

Geoff Husbands articles were part of my research work, and now I believe, I can contribute a chapter to the design of tonearms: www.sembuco.net will give you an introduction to my approach.

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Fourier Sans Pareil OTL

I have a friend's pair of Fourier Sans Pareil OTL monoblocks for diagnosis and repair. I understand he and a friend have swapped them back and forth over the decades and they have low hours. Apparently, they were glorious, and then they weren't. And one isn't currently working. The other one is ???? I haven't put power on them yet. My bench has a preamp project in progress, so I'll get to the OTL's when that's done. I'll start with information gathering.

Attached is a schematic, but I don't have any other documentation.

I'm already planning to replace 360V caps with 450 or 500V caps. I read that these are underspec'ed, and with 340V on secondaries, that's obvious.

Some of my initial questions are:
•Where can I get a manual? Operator manual or service manual?
•How to bias? Procedure?
•What are the target voltages at the Test Points?

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Is my Cambridge Audio Alva Duo the weak link?

Hi.
Can I give you my story for context.

Equipment :
Project 1 X-pression with carbon arm and Ortofon 2M black.
Cambridge Audio Alva Duo
CR developments Kalypso PP EL84
FHXL / Mark Audio MAOP10

The whole system apart from the turntable is fairly new to me and I'm still getting used to it.

I'd had a 2M Bronze for about 4 or 5 years and had a feeling the stylus might be worn, hence the 2M Black. Records seemed to play ok, but were lacking a bit of sparkle. The top end definitely sounded rolled off.

Anyway, after fitting the 2M Black, I was still underwhelmed with vinyl replay..

Now, my digital source is also new to me and is a wiim pro.
I've been messing with the EQ etc simply because it's there, and found the sound I like has that "smiley face" shape on the sliders, with more emphasis on the top end. The FHXL's are quite bass heavy.

So.....

Can I do better than the Cambridge Audio phono stage?

Is it a weak link?

As always, any advice appreciated.. Further info available upon request lol..

Oread - a DIY MM phono approach

This a new thread originated from the Haudegen thread . The initial design evolved on so many and different ways that a split was necessary to keep the overview what's cooking here 😉

Before we can dig any deeper I need to summarise all the info that is used in HtBs Oread.
Firstly some placeholders will be put in here.
Depending on the spare time - have kids they said, it will be fun they said! - the blanks are filled.

The spec goals:

  • MM phono amplifier
  • Gain: <xxdb>
  • exp. THD <>
  • discrete only
  • RIAA implemented in negative feedback
  • RIAA tolerance <xx%>
  • economic friendly (no OOP parts included)
  • proper PSU design
  • more TBD!

Some general thoughts:

Any help and suggestions are more than welcome to achieve a proper enjoyable circuit for everyone.
Depending on the interest a group buy for PCBs could be initiated - but firstly we need to check the schematics/datasheets/simulation and do a nice test build with real world measurements.
I guess till end of this year we do some brain work and after the dust of xmas-warfare cleared up
we can freshly heat up our soldering irons and have some fun in the analogue world.

Hi, introduction before going to the actual topics ;)

Hi! My name is Bartłomiej and I come from Poland in Europe.
I started here when I decided that I want have amazing speakers without that price tag that's so commonly found on them. I started by looking on open baffles, but then moved towards semi-closed designs. Right now it seems I aim to build the Anthologies II by Jim Holtz and Curt Campbell - I will most definitely have some very noob questions on them 😉 Aiming to get a lot of the design made by a CnC router so I will have to learn CAD as well.

Best regards, I look forward to speaking with you 🙂

Ayon Spark Delta 100Hz spikes in left channel

Hi all, I'm attempting a repair of an Ayon Spark Delta. It ran for several hours without load connected and has since developed a 100Hz humm/rattle in the left channel. It's faint but noticeable and it shows up on the scope as spikes at a 100Hz interval. Besides running with no load, it also had 12AX7's instead of 12AU7's in the preamp section, causing oscillations among other unwanted audible issues (e.g. increased noise floor)

It's not an overly complex amp for the most part, but it does have some proprietary electronics for auto-bias, startup sequence, etc. It's not immediately clear what's causing the 100 Hz spikes. Power supplies are shared between channels and the right channel is clean, so... Without a schematic, it's rather hard to get my head around. Mainly because it's a 52kg beast, it takes very little to upset the auto-bias and auto-shutdown and it's a mystery to me what topology Ayon used in this amp (besides it being single ended in the output (paralleled 6C33 triodes).

All B+ and heater voltages measure fine and clean.
Switching out the tubes changes nothing.
Tested the preamp tubes to be sure on a tube tester and they check out fine.
Amp is able to auto-bias the output tubes, but in the defective channel, one 6C33 gets set to a much lower grid voltage. Also after bias settings are renewed.

Before continuing, I wondered if there's anyone out there with more info, schematics or experience? Not an amp we see everyday, so might be an interesting ride.

Better V-reg for Parasound A21 amp?

Been doing some poking around and measuring in my Parasound A21. Reason for this is higher odd order HD at lower load impedance (4 ohms). Mind you this amp already has low THD but odd order HD dominates. Its a fabulous sounding amp but it has some difficulty retaining its composure at high current demand peaks.

I traced much of the "problem" to the dirty vreg solution of a simple 1.5k 5W resistor feeding 15V zeners to drop 65V (!!!). Along with producing tons of heat (running at 250 degrees F !), it allows alot of garbage to modulate down into the VAS and introduce distortion. I injected external clean +/- 15V into the first input stage and it made a huge difference in sound.

Any suggestions of a better approach to regulating the +/- 15V ie. mosfet series reg or is it smarter to run separate standalone supplies instead?

Even a simple 317/337 series vreg would be way better than the dirty garbage zener approach, but there aren't any 3 terminal regs I'm aware of capable of surviving the initial B+ difference at turn on. I would like to do something like separate op amp corrected series regs for both channels.

Any suggestions?

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Designing and end-to-end fully-differential headphone amp?

I came across this circuit and I was wondering about its benefits, and how it could be integrated into a headphone amplifier.

But before we start, I want to clarify one thing: is a standard 4-pin cartridge a differential output device? I suppose the answer is "it depends on how you connect it". In a 3-pin dynamic microphone, you certainly have an explicit GND pin since the moving coil is center-tapped, so it provides two 180 degrees out of phase signals. But in a voice coil with two pins, such as your standard phono cart, the signals are balanced depending on how you wire it?

In the case of the circuit I posted, I understand the input is referenced to GND via R7/R8. Normally, one end would be tied to GND and the other end would be treated as a single-ended input. In the posted configuration, the differential signals are referenced to GND via the R7/R8 resistors. I suppose those resistors have to be precisely matched to avoid distortion, but if they are, the common-mode noise will be rejected better.

First question is: does this configuration provide better rejection against common mode, in comparison with the usual single-ended configuration?

Then we have the gain stage. A headphone amp chip such as the TPA6112A2 has differential inputs, and it would be straightforward to just feed this preamp, with its differential outputs, into the gain stage. This should provide a very clean signal path and lots of headroom.

so the second question is: is this assumption correct?

Finally, IMO, any preamp really needs some sort of tone control. I understand some people will prefer a "flat response" but this is an argument I don't want to go into. I'd like to have some sort of tone control. But I'm confused as for how to implement such a thing

I haven't found any differential tone control circuits. I wonder if it's even possible to implement such a thing or if i'd basically just need two controls, one for each leg of the differential path. Which i think would be a disaster. It would be impossible (or very expensive) to eq both signals exactly the same.

So the practical way to do this would be to have a single-ended opamp, which will basically render the initial circuit (balanced output) moot. Might as well just use a single-ended output opamp and keep the differential input, and everything else single-ended

final question: is there a practical way to implement a balanced (differential) tone control?

Ceramic Capacitor Kit 1pF ~ 0.22µF 6.3 ~ 50V Surface Mount, MLCC 1400 Pieces (28 Values - 50 Each)-- NH USA --

Make offer on both in this thread if interested.
1 KIT is sealed and 1 KIT is opened and missing a row of caps.

The description below seems to be similar or the same.


S-0402
Johanson Dielectrics Inc.
CAP KIT CER 1PF-0.22UF 1400PCS[/td][td]

Ceramic Capacitor Kit 1pF ~ 0.22µF 6.3 ~ 50V Surface Mount, MLCC 1400 Pieces (28 Values - 50 Each)

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JEROn][nMOS Power Amplifier

Hi all
In last two years I have been simulating an amplifier that I believe to be a High End equipment but I need to stop to do this in my life. So, I decided to donate my idea to DIYAUDIO community.
Any of you who find it interesting I hope you have fun.
And yes, the output of this amplifier is from the Gamut amplifier, already presented here in previous years, with some modifications made by me. As the days go by, I can respond to those who ask why each component.
I still need to design in Kicad the low noise power supplies and capacitance multipliers of this power amplifier and show here.

Regards,
Edvaldo

Mccormack DNA-1 R1 input circuit board bad

Hello.
My love , the Mccormack DNA-1 Power amp is bit sick..

Its drops voltage that make the speakers kind of push out when its power
on. also got extreamly hot. ( very hot.. couldnt touch).

My tecnician told me the problem in the input circuit board.
(in fact, 2 tecnicians said same..)
also got the scematics, i was told the parts in it are very hard to find and
better try get new board like this.

Steve Mccormack was amazing kind and help me, also directed me to conrad jhonson.

Conrad johnson (new owner of Mccormack) had similar board that can fit
but "sorry, out of stock.. no more :-( " .
its calld " R1 circuit board".

As im in love with that amp.. am not going to give up...
Does any body have idea of where can i get board like this ??
Any ideas ??

Thanks...
Efi

Faulty Adam A77X amplifier

Just a noob from Germany ready to learn because of recent problems with my speakers.

My Adam A77X stopped working recently.

Both speakers are having the same issue.

They create a popping noise like a metronome. Audio ist still Pasing through them though.



I have a friend who could repair them but he told me he is helpless without schematics.



Could anyone provide schematics for this speaker? Would be highly appreciated
.



Thank you

WONDOM ADAU1701 Without Extension Board

Just got the the WONDOM ADAU1701 APM2 without the extension board so I could basically use it as a pre-amp. But I'm unfortunately unable to find a wiring diagram to connect the APM2 in a 2.1 setup without the extension board, would anybody be able to help?

Thanks!

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The Sound of New Transistors in Old Amplifiers

Hi everyone,

I’d love to hear your opinions and insights about how new transistors affect the sound in vintage amplifiers. I know this is a complex topic, but after years of repairing and refurbishing old amps, and performing numerous tests, I’ve come to believe that transistors have their own sound signatures. This is especially true for transistors in the early stages of the VAS (Voltage Amplification Stage). However, there are times when I notice no difference at all. This inconsistency has been driving me a bit mad, so I wanted to share my observations and get your thoughts.

Here are a few examples from my experience:

  1. Tone Control / Preamp Section:
    When I replace transistors in this part of the circuit, if I use KSC1845, the sound becomes overly bright, and the lower mids (especially vocals) seem to recede into the background. This is something I’ve noticed with most KSC-series transistors. On the other hand, when I use BC-series transistors, the sound is much more balanced, with everything sitting naturally in the soundstage.
  2. VAS Stage (After the Differential Pair):
    Here’s where things get interesting. If I replace transistors in the first stage of the VAS, the effect is reversed: KSC transistors sound perfect, while BC-series transistors seem slightly off.
  3. Differential Pair (Long-Tail Pair):
    When I create a matched pair of KSC transistors for the differential stage, I don’t notice any tonal difference. The sound does become cleaner and more detailed, but the tonal characteristics remain unchanged.
  4. Preamp Circuit with Adjustable Voltage/Current:
    I built a preamp circuit where I could adjust the operating voltage and current to use the transistors in their most linear region. Even in this setup, the differences in sound signature persisted: KSC transistors sounded bright and less harmonic, while BC transistors were much more pleasant and musical.
  5. Driver Transistors:
    The differences are even more noticeable when using these transistors as drivers for the final stage.
  6. NOS NEC Transistors:
    I’ve also tested a batch of NOS (New Old Stock) NEC transistors. Most of them sounded terrible—cheap and uninspiring, with the exception of some power transistors, which were passable.
  7. Output Transistors (Final Stage):
    For output transistors, I don’t notice as much of a tonal difference, but I do have a personal preference for Toshiba 2SC5200 and 2SA1943. These seem to deliver a more harmonic sound and often provide a better soundstage in many amplifiers.

My Questions for You:​

  • Have you noticed similar sound differences between different transistor types?
  • Do all transistors have specific sound characteristics? If so, what causes this?
  • How do you choose the best transistor for a given application?
  • In your experience, what are the best-sounding transistors for audio circuits?
I’d really appreciate your feedback and experiences. This topic has fascinated me for years, and I’m curious to know if others have noticed the same patterns or if my observations are more specific to certain circuit designs.

Looking forward to your thoughts!

For Sale LM1875 Transamp

I have for sale a stereo set of this beautifull small amp, the LM1875 transamp. Fully build, and tested. Worrking perfectly. I have to many amps so this must to found a new home.
For free also 2 bare boards.

I am looking for 30eur + shipping, payment paypall.

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Coaxial HF + 4x10” synergy

I’m looking for some small single-cabinet options that will go a bit louder than a PM90, playing down to 80-100hz to meet a handful of TH18.

I’ve seen some people discussing coax HF (eg bms4090) and that seems great for a diy synergy/MEH, maybe with 4x LF.

Before I get too ahead of myself, are there any existing similar threads, or has anybody tried this? I’ve tried search with no luck.
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troubleshooting SymAsym

Hi,



I am new to this forum and ended up here in my search for a solution for my broken DIY amplifier. The amplifier consists of two SymAsym 5.3 modules, see attached circuit. It is basically the original SymAsym design with added speaker protection and 2x 4700 uF smoothing capacitors.

Both modules functioned well for about six months. The left module then started showing problems. In the beginning I heard cracking noises and occasionally a loud pop. If I turned up the bass or volume further, the crackling went away. Some time ago the module went silent.

To investigate the problem I wanted to try out my oscilloscope with function generator, so I connected the left module to a 100W 6R. Unfortunately, R2 burned out when I placed a 1kHz sine signal on the amplifier input. I replaced R2 but it burned out again, even at low sine amplitude. So probably another component must be defective. I have desoldered all the transistors and measured them with a multimeter (diode test) and they all seem OK.

Because I have little experience with electronics, I would like to ask for your advice here. Could it be that a transistor is defective even though the diode test is successful? Or could it be other components that are causing problems?



Kind regards,



Jonas

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Falling for dome midranges

I have a chance to listen to the vintage American speakers: ADS L980/2. I’m literally impressed them so much especially the midrange sound. They utilize two-inch dome for the midranges. I as a DIY person have a plan if I’m going to use the two-inch domes in my new projects. However, as I listened to them carefully, I found they had unique sound. Well, I have been searching for more info about this driver and I could find some info.

Material: woven soft-dome with proprietary damping
Voice coil: 50mm., wet wound, single layer
Magnet: 1.38 Tesla, 1.1 X 10^-3 Weber
Re: 6 Ohms

From provided info, has anyone ever found any modern drivers that have close specifications to them? Or, in the worst case, no longer find the units with those specifications, could it be possible to special order to nowadays driver manufacturer to make it? Is the data provided sufficient?

Roksan Kandy MkIII and Scope advice sort

I have a Ka-1 amp that's new to me. Just testing with a poor speaker, I hear both channels sound different. One also warms up politely, while the other is cold. I can swap the L&R pre to main audio cables, and the sound difference follows them. So I believe the issue is within the preamp section. I need to test again to see if the other channel then warms. I presume so, but am testing for that, now I'm aware of the temperature discrepancy.

Lets have a look
20241230_113023.jpg20241230_113035.jpg20241230_112949.jpg20241230_112631.jpg20241230_112726.jpg20241230_112741.jpg20241230_112804.jpg20241230_112818.jpg20241230_112826.jpg20241230_112853.jpg20241230_112913.jpg
idfk.jpg


I'm impressed I can post so many pics in one go. Though some have spun around, disorienting even me, and I have the thing in front of me. I hope it's not too much hard work for people.

The final image, I posted larger, is my initial thoughts. Firstly, I seem to have too many relays. I feel the Red one is outstanding, and might route everything through it, even after individual switching. It seems odd, but wishful thinking suggests an easy target. I have a phono amp in yellow. I wouldn't be surprised if the pink amp was part of it, or the two green 'Dished' caps. The green amp a buffer for the tape loop. The black one, the only one common to the couple of different inputs I have connected my CD player to, and seen the same fault. So have I narrowed it down to a very small area.. maybe. It has two dished caps I guess are on power, but shared by both channels. Then two more on the output side of things, I could swap over.
I have metered the preout, and it looked.. like I need to finally buy a scope. I see 35khz on both channels. No AC. DC of 2v and dropping. In about 30 seconds, they were down to a volt. One about 0.2v higher and I did a cold start a few times looking at this.
I guess I could now bypass the preamp completely, and inject my cd player straight into the power amp. Or bring the preamp signal out, to another amp. I have noted the main amp sits in protect if the preamp isn't connected to it. So this DC means something. I don't know. I have only an inquisitive mind. No training.

I have fancied a scope for years. Maybe 50. I have no funds or space for a desktop item, but see Ali has things from £20 now. Some are a basic multimeter at a glance, but are doing something. Has anyone messed about with these. I don't need 27mhz anymore. Just something to look at audio waves comparatively. Will anything do really? I would like to eye up a CD player I guess, but this item will live forever in it's box, and maybe never meet a real signal generator. I don't imagine I need much?

Passive radiator

Simple question:

Can a passive radiator incorporated into a subwoofer enclosure increase efficiency? Or is adding dynamic mass to the system, rather than just moving air, a reduction in efficiency but perhaps beneficial for directionality?

Is an "omnidirectional" passive enclosure -- for example, a 12" subwoofer mounted in a front baffle and housed in a framed "sealed" cubic enclosure with a compliant skin of some sort, e.g. latex or something with a relatively low modulus of elasticity, stretched taut across the five other sides worth experimenting with?

Seeking advice on kits

Long time since I posted but I am seeking opinions on speaker kits . Haven't built a kit in about 20 years but had good luck when I last did it ( gr research towers going strong)
I have a large room 18ft wide by 38 ft long with high sloped ceiling( 22ft to peak) in front half where the speakers will be placed. Room has alot of hard surfaces.
Trying to decide between Neil Blanchard transmission line Towers, Troel Gravensen fatal 10 or 12-430, sb acoustics gema, or finally the css audio 2tdx or 3tdx tower kits.
From an aethetics perspective I gravitate to the classic box speaker shapes ( TG speakers, Gema) or the Neil Blanchard design.
Will be feeding with a yamaha as2100 or classe audio ca-150 ( need a new preamp for the classe)

I listen to blues, classic rock , jazz at low to moderate levels.

Neil Blanchard designs most cost effective and get good reviews . Wondering if anyone heard any of these speakers in person and might pass along their opinion.

Toshiba 2SA1943 / 2SC5200 Power Transistor Upgrade - recommendations?

Looking for a power amp upgrade - currently using the Toshiba 2SA1943 / 2SC5200 what's new and available that would be an easy upgrade for those parts.
I have looked at the Sanken 2SA1216 / 2SC2922 and Sanken 2SC3858 and 2SA1494 but they don't seem that available from trusted sources.
Any other recommendations that could more or less drop in and are available?
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