I found this decoder by Pavouk, from long ago. Seems like is still being used in recent equipment. The thing that bothers me is that 7-12V AC is being used with a bridge rectifier instead of using just 7-12VDC. Is there an obvious reason for doing so?

BTW is this really still being used? I mean even AMB gamma 1 and Twisted Pear S/PDIF Transceiver uses similar chips.

Hi, I'm going to trash an audiolab 8000p if I don't replace 2sk389v.

Who has this jfet???


Thanks a lot!

It should be simple to get a basic NE5532 stage to work but I've been trying for days, built 2 different regulated supplies and 3 circuits and can't get anything to work. Loud buzzing and distorted sound on everything. Frustratingly, a chi-fi single 12v supply pcb fed with 15vDC from my bench supply works perfectly - silent and good sound. Pictured in red. No schematic available.

I have +-15v out of the regulated supplies, standard book circuit. The whole system is grounded through my tube amp via the interconnects. I also put 1uF in series on the input to the NE5532 stage in case there was DC offset from the DAC (output stage there is also a NE5532). Didn't help. I could try 1uF on the output of the NE5532.

I've built a lot of tube amps but I'm unfamiliar with chips like this and have no idea what's not working. I need help!

Ex audiophile in my younger years (in taste, not budget) in the early 1990's. I was a musician/singer with a passion for all things high fidelity. Also an ex audiophile grade hifi salesman in my much younger years (circa 1988 - think Dahlquist DQM-9's, Carver & Crown).

Moving around quite a bit over my life I have more or less shed the large speakers and separate A/V components for smaller, more practical solutions, like wireless bluetooth headphones and small compact speakers. I sold my last proper hifi equipment back in 1998 (A Yamaha natural sound integrated amp A-1020 & some KEF 103/4 speakers KEF - Reference Series Model 103/4 (1992-95) - United States

I am now in a position where I find myself missing a proper home audio system and want to take up this hobby again.

This should be interesting, as I am currently in a 800 sq ft apartment with a small living room. I have a 60 inch 4k Samsung tv and currently just use the built in speakers, ugh. I live on a very busy, noisy street. Though I want some speakers that sound just as great at normal or low volume (neighbors) as they do at higher volumes. Yeah, so... Need to remedy the situation!

When I was younger I had a keen interest in speaker building and did have some projects back in the late 70's and 80's. I want to either build a new set of speakers (larger bookshelf sized on stands) by doing one of the following:

1. Refurbishing and rebuilding a set of vintage/used speakers

OR

2. Building a custom set from a flat pack knock down kit (I have no space or tools for woodworking so need to buy cabinets already built).

I have some electronics engineering knowledge dating from the late 1980's, so that should help. The kind of amplifier solution I put in place will be largely determined by the type of speakers I build: powered or passive... Not sure about whether it will be a simple 2 speaker setup or a full blown dolby atmos surround experience. I am leaning towards an excellent imaging 2 speaker system with a good soundstage that will serve both music and video duties.

Anyhow, great to meet you all!!! 🙂

Hi

Looking to buy a used UMIK-1 USB microphone. Let me know if you have one, want to use REW and this to equalize my room and also learn more about equalization

I am located in California.

Thanks

S

Guys. Which modules U recommend to be used to input Instr in stereo on windows machines?

Hello,

My ESS Sabre 9018 eval board is for sale.

I would like $250 USD (OBO) for it.
Shipping in the US will be around $50, due to the length of the board.

Please message me here, or email me directly at steven.osugi@gmail.com if you have any questions.

Thank you very much,

- Steve O.

As title. I'm in the UK.

Cheers

Hi all,

Is anyone familiar with Clearaudio emotion motor?

Is it serviceable, meaning can be opened so someone have access to the internals?

Is it full of epoxy or sealed some way?

Thank you

Stay safe

Everybody hates Bose. Even I think their stuff is grossly overrated. Nevertheless I bought a Companion 5 sub-box with all the electronics removed for around $20.00.

As is typical Bose, the cabinet cannot be disassembled. To access the driver the entire unit must be destroyed. My original for this was to connect a Sony 40w plate amp but now the box has actually arrived - the plan has changed. The power supply for the original amp and electronic is contained within the sealed compartment containing the driver. It almost impossible to obtain specifications for Bose equipment but the stated power consumption is 300w - which kinda makes sense. How else are you going to get anything resembling bass from a dual coil 5.25" driver?

As I already have a transformer, my new plan is to replace the missing electronics with a TPA3116D2 and a low-pass filter. For this to work I have to assume the output from the original PSU does not exceed 25v - Based your collective experience - how likely is this?

Have a JL Audio e1800DM monoblock that has a protection/shut down issue. The amplifier was playing normally until a very heavy note was required, then the protection light would flash very briefly, the amp would power down and restart and the whole cycle would repeat as long as that level of output was required. If the volume was turned down just under that level, it would continue to play uninterrupted.

I tested it on the bench with a 2ohm dummy load and a 50hz test tone. Output was continuous and uninterrupted until the voltage at the + input block read under 9v due to my small power supply and mains current limiting bulbs, which were full brightness before the amp would restart. I checked/resoldered all the legs of the heatsink mounted components and solder joints but yielding no results.

I tested the amp in the vehicle afterwards and monitored the + input terminal voltage but the amp still shuts down on heavy bass notes even with + input voltage not dropping below 12v. The rail voltage was around +88v with 14.3v on the + input block. I have no idea what can be causing this.

A self inflicted wound occurred while trying to check the rail voltage during testing. The meter probe slipped in between the buss bars and contacted the board which blew up Q31. I can only read IRL___ and the rest is missing. It is likely a logic level FET.

What is Q31? I can only begin troubleshooting when this is replaced but possible causes of the initial issue would be welcome now.

The upper right side of this article shows how to current regulate a filament using a voltage regulator as a CCS.

http://www.tubecad.com/july2000/page6.html

Let's say the tube specification draws 800ma at 6.3 volts after reaching temperature. So I set my regulator to 800ma. Will the voltage across the tube be 6.3 volts after its hot even though the current source IN is 10 volts here? Because, well, ohms law. It's really not necessary to regulate both the voltage and the current as long as I'm regulating one of them the other will automatically adjust to what the tube needs, correct?

I want to limit current so the filament will last longer due to no inrush, because a cold filament has very low resistance. Most filaments like light bulbs burn out at start up.

But was wondering if my basic ohms law is right here.

I've recently become interested in trying to put together a Linux based computer for processing audio into different streams - a PC crossover.

This is a new direction for me, moving away from the DSP hardware solutions that I have been using for the last few years, but I have some experience in C/C++ programming and have found a few examples on the web of what is possible and these are very encouraging. I have an idea for a new way to PC based loudspeaker crossovers using my Active Crossover Designer tools (ACD).

I have discovered the LADSPA plugins that will, for instance, execute one of a number of filter blocks on a stream of digital data, for example a second order Butterworth highpass filter. This is way too specific of a filter implementation. I have been working with IIR DSP for awhile now and I'm quite familiar with IIR biquad coefficients. My crossover software, ACD, calculates the IIR biquad coefficients from more familiar filter descriptors and makes them available to the user. What I am looking to create is a new plugin that takes the biquad coefficients as input directly. The code for the plugins that I have looked just calculates the coefficients from other parameters, like the type of filter (second order high pass Butterworth) and one or more parameters (filter corner frequency and gain). My "generalized biquad" plugin would simply take the five (normalized form) coefficients as parameters and then would use them directly. These are the same five parameters that miniDSP uses, with a few sign changes, so miniDSP biquads could be imported as well.

Once this new plugin has been created, the doors are open to implementing a complete crossover in Linux using an approach like what is done here:
Digital Crossover/EQ with Open-Source Software: HOWTO | Richard's Stuff
Richard Taylor's efforts have paved the way, but his plugin set is very limited. Using the same approach, but with a generalized-biquad plugin that can implement any first or second order filter, would make Linux audio better suited for loudspeaker crossovers.

I'd like to chat privately with someone who has worked with LADSPA before, as well as ecasound, about some practical aspects of my approach. If that's you, please drop me a line.

Otherwise, I will try to post my progress here from time to time. I'd love to hear people's thoughts on all of this as well.

Hi!

A friend of mine bought a thermometer for remote measurings temperature of working valves,

he reads 242degC at EL34 in PP OPS,

someone told him that the temperatures >220degC might be danderous for valve lifetime.

Do you know what should be optimal/max temperatures of the bulbs?

Hi

I have just completed a guitar amp build with built-in digital effects and it's gone fairly well. This has been a combined electronics, woodwork and software project. I'm pretty pleased with it but I am struggling to debug a noise issue and was wondering whether anyone on this forum might be able to suggest any ideas.

Bascially I can hear an annoying warbly tone, more so when I increase the amp gain to get distortion. So I guess the noise is getting in right at the input. I can provide more details of the design but essentially it's as follows:

- enclosed switch mode power supply module adjusted to give 24V DC from 240V mains. 0v clamped to earth on PSU.

- Opamp buffer stage to give high impedance input for guitar 1Mohm.

- Teensyduino 3.6 + teensy audio module.

- approx 10W Class D power amp module (loud enough for home use) running from 24V into 8 ohm.

- Switch mode step down module to give 5V for tensyduino from the 24V

- 3.3V LDO for analogue stuff.

Any help appreciated, thanks.

I have a pair of one and three-eighths inch circular Dayton waveguides and was wondering whether there would be any point in "modifying" these to accept a 4-inch full range driver (FaitalPRO 4FE32 4").

Or should I instead buy some compression drivers that work down as far as say 1500Hz so I can mate these with a pair of RS275P 10-inch 4ohm Dayton woofers in a sealed enclosure? Will the Daytons run this high or would I need to think about 12-inch or 15-inch pro woofers instead?

I will use an active crossover I have that's set at 250Hz if I use the Faital 4-inch drivers. Otherwise I can use another active crossover that's variable.

I have a pair of 400wpc into 4 ohm NCore monos and will drive the Faital Pros with a Burson Bang (27wpc) stereo amp.

Hi,

I'm building an amplifier with the TAS5518 processor.
I wish to use also the PWM output from it for the headphones.
Can somebody tell me how to convert the TAS5518 Differential PWM Output to analog to be used with a standard OpAmp?
Thank You very much.

Hello.
I want to bias the Long-Tailed PI for the lowest distortion, so it should be biased as symmetrical as possible.

I have a 360V supply and I allow 90V for the tail. I am left with 270V for 100K RLoad, 2.7mA. Now the loadlines :


Is is right to bias at 177.5v since it's the middle point of the voltage swing, for the lowest distortion (and IMD) ? Or Am I doing it wrong ?

Thank you.

This is not exactly a question about the "BEST" HT set up. It is a question about design objectives or duties for each of the speaker positions in a 5.1 HT set up.

Does anyone know where I might find the specific frequency ranges and "impact" that one expects to see at each of the 5.1 positions? I understand frequency can be limited and/or trimmed by the receiver. I just wanted to know if there was a published spec. Right now I assume they are all full range unless you make it something else. And, I assume that most systems are what you can get by with and not what would be optimal. But I don't know.

So, while I read redundantly that the center channel sees most dialog, it is not clear to me if, in addition, it is seeing the same low frequencies as the front mains.

Likewise, do the surrounds need to have the same frequency, low extension, as the front mains?

There is also the issue of "impact." Do the center and surrounds carry the same responsibilities for sound "impact" as the front mains?

I can’t find much about home theater that isn’t marketing BS and euphemisms. Thanking all in advance.

Hey folks! So my home studio currently has 3 sets of monitors in use and things are cluttered enough as the studio accumulates more stuff. One pair are Avantone mix cubes for lo-if critical midrange dial in. Another set is a well regarded 2 way 5.5/1” dome. The other is a DIY 2.5 way tower using Scanspeak drivers that extends to 35hz. All in all, cross referencing across all three has provided outstanding results that translate to hifi systems, small portable mono devices and car systems.

So why mess with a good thing? Clutter in a nutshell. And second, the need to accomplish something.....self fulfillment and ergonomics....One set of monitors on the bridge....minimalist but ful functioning.

My goal is a horizontal three way monitor with an 8” woofer flanked by a vertical mid/ high compliment with a 4-5” midrange and tweeter above. There are no constraints on the tweeter type. I’ve pretty much settled on the Dayton RS225 for woofer duties though.....unless someone can talk me out of the fantastic performance and relatively low cost? I’m all ears.

So here’s the kick.....and an essential part of the design....I NEED to be able to flip a switch on the front of the monitor that will send only the full range input signal to the midrange driver, bypassing all crossover components inside. The purpose of this is to mimic or improve upon the revealing character of the Avantone Mixcubes. In that, the midrange driver will need to cover 400hz to 8khz flat and then acoustically rolloff....no breakups above the 8khz range but a smooth progressive descent to 12k or so.

Ideally bass extension should have an f3 in the low 40’s. Room is very well treated so I don’t expect much help from room gain. Most tracks of even the most aggressive bass devices are almost always highpassed at the board at 35hz or higher anyways in the real recording world.

Thanks again for the help!

I recently bought a UK version of the Hafler IRIS preamp to replace my dead IRIS preamp. I just rewired the supply to change the unit over to USA current. It was working beautifully. I added my IRIS tuner to the chain and after I plugged that into the preamp switched outlet the preamp went dark and quiet.

I found the preamp internal power fuse was blown. I replaced it with the fuse from my original USA preamp. And still nothing. A test with my DMM showed the replacement fuse was open. The sticker inside my old USA IRIS preamp states "use only direct replacement 3/16A 125 slow-blow MDL/3AG".


The two bad fuses don't match the sticker, and they don't match each other:

UK fuse: 1/10A 250V 313 (resistor inside it)

USA fuse: 3/16A 250V 313


So is this a 3 amp fuse? A 16 amp fuse?
How do I find the correct fuse? The big online vendors are no help at all.

I've been searching Mouser, Digikey and Newark for an hour, and I can't find anything with "3/16A" in the search fields. Oddly, if I do a web search for "MDL 3/16" I can find a reference to that type of fuse in the Eaton docs, but the fuse shows up as "obsolete". I found an updated version on their site, and searching for that part number via DDG takes me to a Digikey page for that very fuse, which can't be found by using their online search tool!

I've had trouble using these search engines before, especially when trying to find capacitors. If anyone could please explain to me the right way to search for this fuse I'd appreciate it.

Hi, I resumed old project. Box, front and rear panels are ready.

I post the block diagram here.

Only the ground path is drawn - the green lines between the modules.
The ground path follows the signal path, module to module.
All modules are separate pcb's to be build.

Mby the volume control, TC 9459 would be one pcb with two vol IC's for 4 channels. Anyway, it does not require analog power supply. Signal path is not connected to the microcontroler psu.
Analog and microcontroler psu's have direct ref. point to one another (red circle on diagram).

The main idea is to use the input selector board for ground return distribution.
Signal and power ground would go along through the signal cable shield.

1. Output DAC buffers would get analog supply ground through the input selector.
p.s. Note that input selector switches ground.
- That would also switch dac buffer circuit on and off.
- That would connect analog psu ground to the DAC digital psu ground.

2. Crossover unit would get power ground through the input, via the signal cable shield.

3. The next modules (subsonic, linkwitz transform) are connected in series and would get power ground via the signal cable shield.
Analog +-15 v would also go on a bus, following the signal/ modules layout. So modules would be connected in series.

All that means that the power return path (for dozen opamps) would go along with the signal path. Power consumption current in this device is small compared to a power amp. But still few times higher than the signal current.

IS it ok to run signal and power ground together, or I shall run them on different paths?

Regards - Emil


img host

When laying out a PCB, what is the best order in which to place the input resistors and caps? Should the RF filter come before the DC blocking cap, or the other way around? It seems like a good idea to put the RF filter cap as close to the first input device (
op amp, JFET, etc) to minimise the trace length that can pick up noise.

I've attached an image showing two different orderings, there are other combinations possible. Are the identical, electrically speaking?

I was playing around with FET compression circuits recently, after I read this article - VCAs (section 7). The FET circuit does indeed work but I was only using a fairly general purpose 2N3819.

Does anyone know any resources I could use to further my knowledge specifically in FET compressor design? i am interested particularly in finding how to select FETs for great performance (even though I realise they are now hard to find) and how I can modify the design. I noticed in particular these designs are quite noisy and the switching can be a bit rough, it would be good to know how great FET compressors get around this

any master FET selectors out there? 🙂

Hi,

I'm a mechanical engineer, father of 3 and music lover. I love DIY and have built several interesting things, from a headphones amp (OpAmp based), sll cables I use in both guitar and audio rigs, several guitar pedals (clones of Fuzz Face, Big Muff Pi, Dyna Comp, Electric Mistress, Phase 90, A/B Switch, etc.) and small accessories to my guitars and DSLR rig.
I own a Fender Stratocaster, several pedals and a Laney Cub12 (with mods) and a simple audio system:
Sources:

• Garrard SB125 Truntable (1970's belong to my dad) + Cambridge Audio Azur 640P Phono Pre amp
• Sony CDP-590 (modded by my friend Tube Dude from Aveiro, Portugal)

Amp: Prima Luna ProLogue 2 (2x12AU7+2x12AX7+4xKT88)
Speakers: Visonik Revo 4

So I am very glad to be here among the giants, while attempting to roll up all of my skills and one of my major interests into a satisfying, repeatable experience. JAMMING THE HELL OUT TO MY FAVORITE TUNES and hearing some new ones.

Recently I sat through a series of SynAudCon courses. Prior, I have messed around in dsp land a bit. Working on a small club system, hoping to design and build up a pair of tops to complement my sub setup. Also, I need a more than decent set of desktop speakers. Classical, Jazz, Nuevo bossa, King Crimson, some of the flavors I crave... .

I would love to hear what everyone thinks and does, I am backing out of other interests to focus on the audio as it ties in closely with my profession. I will be lurking here quite a bit, and might pop up w some questions.

Peace yall!

How much does a power amplifier development cost if you order development? And which documentation package should all of this include? For example, a computer model, circuit, drawings, documentation for the board and transformer?

I have an NHT SubTwo with a history of blowing outputs.

Read more about it's history here:

http://www.diyaudio.com/forums/showthread.php?s=&threadid=43199

That was two years ago, and now I have another blown output. This one occurred when I was moving the sub a few inches to reach an antenna wire behind it. The sub was in standby mode at the time, upon being moved I heard the two 10" drivers get shorted to rail voltage, followed very shortly by the fuse pop.

Damage is a failed original (non-counterfit) 2SC3281. The previous failure two years ago was one of the 2SA1302's.

I am wondering if there is a problem in the jack for the interface cable to the controller, and that moving it caused a connection failure. I'll be inspecting it carefully.

But, alas, on to the reason for this post:

Now that the 2SC3281/2sa1302 combo is no longer available, I am looking at replacing all four outputs with the newer 2SC5200/2SA1943 devices.

Question 1: Does anyone sell these in matched sets? The amp design has no emitter resistors, so I'd really like to install a matched set.

Question 2: If nobody is currently selling these as matched sets, is there a relatively benign way to match transistors? I don't mind buying a dozen or so of each and building a small circuit to match them, and my DVM has a hFE test mode as well. I don't have access to a curve tracer, though.

Question 3: are there now any better output devices on the market that would be a drop in replacement for this design? (see schematic in thread linked above)?

Question 4: any suggestions on what's causing them to fail? Poor amp design is my #1 suspect.

Jlsounds ak4493 dac + oled

i have for sale jlsounds ak4493 dac +oled. Used a few times.
60€ included shipping in europe.

prefer shipping in europe

Toshiba 2sk389v replacement

Who have this jfet?
Please contact me!

Has anyone played around with vertical deflection tubes (6LU8, 6LR8, 6KY8, triode with power pentode) for audio purposes?
A pair of 6LU8 looks promising in a small push pull setup: you'll get 2 high mu triodes, one for input stage and one for phase splitter (split load) and two beam pentodes with Pa of 14w each. Looks like a better candidate in terms of power than in example ECL86 and friends.
But I cant find much information on the net using those kind of tube for audio -- any obvious pitfalls using them?

NAKAMICHI IA-2 AMP HELP

Hi,

I just purchased a used Nakamichi IA-2 amp. Nice piece of gear but there is a very loud hum from LH channel when switched to Phono and there is no signal source. (All other selection are silent.)

Also, the trouble and strife says the reproduction is metallic...

Could somebody kindly advise about troubleshooting?

Many thanks!

Ftpols

Hi!

So... because of different reasons, I'm having trouble realizing projects and instead I'm spending time planning and brainstorming around different project. Not the most productive times... And the building phase of projects works better on my sometimes troubling anxiety/depression.. Anyway, a new idea came to me the other night. After a quick session in sketchup:



Features:
- 3 - 4 inch full range driver in narrow open baffle

- 6 - 8 inch woofer in vented box
- passive crossover

Me/room:
- Small room, about 8 m2 I would guess
- listening distance about 1,5 m
- rather short to back wall
- very wide taste of music with focus in metal, acoustics, rock, country and cinema score

I have no experience of open baffle, but I think this would be a fun try. What I'm looking for and thinking with this design is a open and room filling sound. Could that be achieved?

What would you call this? Hybrid? Semi open baffle?

Suggestion on drivers? I'm considering a pair of mark audio. Have been looking at them for other projects. If this one fails I could use them in another build. But that is true for all drivers I guess 🙂

Crossover point I guess I have to see when I have decided on drivers, but I'm thinking 150-200 Hz. Thoughts?

Is it going to be tricky to power these with a single amp and passive crossover? I see a probability of the woofer needing more power than the FR driver. Depending on driver off course ...

Thanks you all.

Hi all. It starts!!

L and R power amp dual mono based on the 500. And added a center channel with a NXV201PS to hopefully get a better center channel than the denon AVR4520.

The case is quite large but it's still a tight fit.
Center channel is trigger activated only from the AVR and has LED on front panel to show it's on. LR dual mono main amp has push button on front with LED and also trigger option from the AVR as well.

Hi,
I previously built an ACA 1.6 biased at 1.2A with a 24vdc switching supply.
Sounded so good, I wanted a more powerful version for my 4 ohm speakers.
I built a pair of ACA monoblocks. Scratch built, not kit.
I used larger heatsinks, and a beefier switching power supply.
I bought a pair of Meanwell 36vdc@4.44A switching power supplys, and my plans were to bias the amps at 18vdc,(midpoint), and 2.2 amps current, by adjusting R1,R2,R3,R4 to get 2.2 amps current.
Am I correct in assuming power delivered to my 4 ohm speakers will be about 19 watts peak? Is the correct calculation: (I^2)*Z or 4.84 * 4 ohms = 19 watts peak? If so, would RMS power then be .7 * 19 = aprox. 13.3 watts?
I cannot find a clear,(to me), description of power calculation for a class A amp, based on voltage and current at the output.
Any help appreciated.

1 amazing horn.

No shipping.

I can meet you in waterloo, cedar rapids, or iowa city.

I may be up for a trade........


$75 cash

Hey all!
Have some miscellaneous parts and boards boards I no longer need.

1. Sigma 11 (30V, The σ11 regulated power supply) $50
2. DIY dual bridge rectifier (with 2x 1000uF 50V BG caps, and 8x MSR860 diodes) $50
3. 2x Auricaps (1uf, 200V) $30 the pair
4. Caddock MK132 resistors - 2x 2.7K, 1.5K, 330 and 75 $15 the each, $60 the lot
5. Audioquest angles IEC adapter $12, new in box
6. 4x 1000uF 25V Blackgate NX caps (test strong) SOLD
7. 2x 10uF 50V Blackgate NX caps (test strong) SOLD
8. 1x 47uF 50V Blackgate NX cap $50
9. 1x .47uF 50V Blackgate NX Hi-Q cap $50

Ok I have not even stated assembly yet and found I made an error. For R3 I have a 10k 3W resistor, same as R14 and R24 instead of a 1W resistor. Any issue with using a 3W in R3 besides some additional bending to get it to fit? Just want to make sure I am not upping the wattage on a resistor designed to be a weak link in case something goes wrong.

Hello,
I wasn't sure where to ask this question, but since I'm working with a tube amp, I'm asking here.

OK, I have drawn (copied) an SET tube amp schematic in LTSpice.

Now, I would like to run a simulation so that I can measure DC voltages at the tubes, specifically the voltages at the cathode, plate, and screen grid.

I have absolutely no idea what to enter into the fields in the screenshots I've attached.

Thanks for help with this. I have spent many hours Googling, reading how-to's and watching videos, but I cannot find any examples to copy.

I've just finished a clone of Fender Champ. Voltages were fine, so I proceeded to other tests. It sound OK, but there's hum. It's still not in the chassis, since I'm experimenting with that. Since that's my first working tube amp, I really have no much experience, so I need advices and help.

At first I was running it with full wave SS rectifier and AC filament supply. But the hum was really strong, so I tried with DC filaments (used battery with reference to circuit ground for that). Not much improvement.

Then I tried to use high voltage regulated power supply I've build recently (using MOSFET for regulation, I think I've found a schematic for that here, called Elvee PS) for B+ and AC for filaments and hum reduced significantly. Tried the same with battery for filaments, and hum increased a little!

Then I noticed that my guitar's volume pot contributes to hum level. It's smallest when volume is at zero, then it increase dramatically when pot is at the middle point, and then decrease when pot is from 9 to 10. No idea why.


How to troubleshoot this?

Hi everybody, I´m put my hands in a hammer dynamics super12 FR kit, the novo the sound is great, very clear and detailed.
my question is, on the web all agreed that the best damping solution is asphalt layers cover by bubble wrap and a deflex layer behind the driver. unfortunately beside the wrap I can not find the rest. I just found a Dynamat extreme sheet (for car isolation), can i use it? (i believe is the same use of the delex and asfalt), my concern is to not "eat the bass" instead of order it.
any experience? thoughts?

thanks in advance

Does anybody still have any of these? I'm only interested in the bigger panels(8"x4").

Does anyone ever tried to open SSG silver mica caps?
Thanks for tips. I want to put them in a wax/wood.

Hello folks,

Doing my first recapping here and a have this large smoothing capacitor next to the 6CA4 rectifier. It has 5 pins on the bottom. It seems to hold two caps (one of 40 uf and of for 10 uf). Can you recommend a good replacement & manufacturer that you beleive would do a good job?

Many thanks for your help,

Romain

I'm about to cut wood...but I wanted to get a second opinion on slot loading two 18"s vs 1 facing me and one floor loaded (3" off floor).....

Anyone feel strongly that one offers more sound quality than the other?

Hello, inspired by thoughts on one point speakers I took what was lying around from other projects and came up with these great sounding speakers...
Fane 15" 400 paired with a no name tweeter
Separated at 3000Hz with 1st Order Butterworth 2 way crossover design



The funnel combines all frequenzies to a very satifying hearing bath...
It also serves rizing the mids very much...
I hear the Tannoy Autograph approaching...🙂










Next summer I will try a simular enclosure but with open back...like here:

Hello guys! My first post here 🙂 Never want to join a party without bringing something, so I thought i'd offer a tutorial on Asynchronous Sample Rate Conversion. I know it's been discussed many times, but I'd like to devote a full thread to an in-depth technical review of this technology. Are you guys game?

Rough outline would be something like :

1. Focus on the easiest case, modest upsampling from say 44.1kHz to 48kHz.

2. Review the details ... in both time & frequency domains ... of common filter structures used for ASRC, focusing first on "jitter free" environments.

3. Describe how the system responds to jitter on incoming data, compare to a PLL-based clock recovery system.

Sound like a plan? I know you guys know nothing about my credentials 😉 but maybe we'll get through that as the thread develops, cool? Of course, I'll welcome all inputs from the experts on the forum.

Last weekend I heard a pair of Cain and Cain Abby’s and was pretty blown away by what was possible with a single driver. I left inspired to try one.
I know there are tons of “what should I build threads” and I’ve read a lot, but everyone’s goals are so different so I do apologize for another but I want to make sure the design is good for my space.

My goals:

Room- 11’x18’, the speakers will go on the shorter wall and because there is an entry door in one corner they have to sit about 3-4’ off each wall. Also because it is an entryway, they really can’t be more than 2-2.5 from the front of the speaker to the wall without crowding.

Size – tall is great and as big as they need to be, not compromising size for sound.

Shape – I love the look of the front facing double horns, but sound is the most important with the space I have. Ideally they could sit against the wall. Kirshma looks great, Venom too. Silbury if rear-facing. Or a tall Voigt shape like the Abby.

Sound – I don’t know enough to say, but airy, spacious, great mid range. Clear bass, as much as possible without getting boomy. I listen at lower volumes and love hearing the detail in music.

Driver – completely open, I was leaning towards Alpair 12p, but hearing smaller drivers like 10p are sometimes preferred. Around 6”, around $300 /pair.

The room seems like the biggest factor and some of the more popular designs might not work well with it.

Thanks all

All items in perfect condition and full working order.

1) One pair of Pos & Neg SuperTeddyRegs from Teddy Pardo. These are set to 15V but can go from 12 - 22V with fitted resistor. Simply change R2 for higher output.

£60 for the pair free post UK - EU & USA £10 shipping


2) Ian Canada Raspberry Pi Hat Isolator board. Unused and new. Will throw in Ian's Battery supply board.

sSOLD


3) DIYINHK XMOS 384K I2S board. Brand new and unused.

£20 free post UK. Overseas please enquire

4) Talema 70051K 15VA 2 x 9V toroid transformer new and unused.

£12 free post UK


5) PEECEEBEE PSU board from Shaan. Unused

£10 free post UK

Hi

Im trying to centralize some information. This is because after maybe 5 years of building fw amps and others, there are still (simple) things that i cant control or fully understand. Im just a musician 😉

So, in this thread im hoping that the simple little big things come to an understanding for those that like me, are dummies in a good sense.

So,

I've built the DIY sony vfet (2 transformer/psu in the same chassis). It's working ok.

I've built the Ba3 pre amp with dual PSU also in the same chassis. It's working ok.

I've built the F5, working ok.

I've built one of the APEX audio amp's. Working ok.


On all of this projects i had some doubts. Regarding fine tuning procedures, cables, ground noise/loop and other things that for some of you, are obvious things to do and because of the money invested here, and high expectations about sound, im afraid of something really obvious missing.

Will attach here some pictures of the amps.

One thing is certain. I have noise. I want to get it out of the amps...if possible.

Have an OB system with MA speakers and eminence.

I thought that going with dual Psu's would solve my noise issues with sensitive speakers

Please, feel free to point everything that you find silly or wrong. I just want to improve.

Thank you for everything

I'm in the process of building a diy DAC/Preamp.

I'm looking for a way to add volume control using a remote to my system.
Preferably a physical volume knob aswell as a remote using IR or RF.

I looked around on the ebay and found a few options, please tell me if these are even worth trying.

Any other suggestions are very welcome.


Double Digital led Potentiometer Remote audio Volume control,Amplifier dc 5-12v | eBay

9V AC/DC Infrared Pre AMP Motor Audio Amplifier Volume Remote Control Board Kit | eBay

Warm greetings, gurus of DIYAudio! Would like your advice on what to do with these (47X25X30 cm) Monitor Audio R300 oldies but goodies, which I picked up second-hand for £60. 🙂



Currently working OK, but I've had to strip out the dried-up ferrofluid gunge from the tweeters with a blotter to get any top-end, and they sound a bit harsh and sibilant at the top, and a bit lumpy in the Infinite Baffle bass for my bluegrass style tastes. 😕

The front end, Rotel RA-931 25WPC Amp, RT-950BXTuner and RCD-965BX LE CD player:



The speakers have a simple inductor on the bass, and a 2nd. order RCL filter on the metal domes, with 2.2 ohms series resistor and 3.3 microfarad capacitor plus coil, @ 90dB efficiency:



I could just slot in a 8" Fostex FF225WK fullrange unit and a 65mmX190mm bass reflex tube in the tweeter hole and there is little carpentry involved according to my measurements with a cabinet volume around 28L, or just replace the 94mm tweeter with a Morel Classic CAT308-94. I COULD fit a reflex port on the back if I stick with existing bass unit, but know nothing about what would be a suitable size or whether it is feasable. Because I like bass reflex, and the way it sounds more alive. Or perhaps I should just replace the tweeter ferrofluid and get it sounding smoother?



Any thoughts or advice? It would be much appreciated. 😱

Hello guys, my sennheiser gamer one is louder in the right side than the left one, I figure out that there is a way to bypass the volume potentiometer fixing the problem, but I don't know how to do it, someone can help me? If there is other way to fix too
Sorry for my english, Im from Brasil.

I just ordered one of these from eBay. Thought I would pass on the info.
High current toroidal transformer w/ 46.5v and 23.25v secondaries approx 1kW | eBay

Hi Everyone,

For quite some time now I’ve been working on asynchronous USB audio signals. I would like to introduce you the final version. The XMOS firmware is changed so it can play 32-bits PCM audio and to play DSD audio.

Features:

1. XMOS 500MIPS 64-pin chip for transmission of asynchronous USB audio data to I2S line.
2. Sampling frequencies: 44.1kHz, 48kHz, 88.2kHz, 96kHz, 176.4kHz, 192kHz, 352.8kHz, 384kHz
3. Resolution: 16-, 24- and 32-bits.
4. I2S or right-justified data (external pin configuration) in 32-bit frame.
5. Two low jitter oscillators for audio 45.158MHz and 49.152MHz.
6. Input for external master clock.
7. LP5900 ultra low noise linear power regulator (6.5 uV RMS).
8. No switch mode power supplies in the PCB board.
9. Galvanic isolation (outputs, two oscillators and reclock are after the isolator).
10. Galvanic isolated output with open collector for DAC configuration in DSD mode.
11. Automatic detection of data type (PCM or DSD).
12. Master clock outputs: 45.1584/49.1520MHz and 22.5792/24.5760MHz (I2S 22.5792/24.5760MHz are reclocked by edge from 45.1584/49.1520MHz)
13. Three different ways to power the board, possible configurations:
13.1 Power by USB bus.
13.2 The USB part is powered by USB bus, but the generators and the reclock are fed by a separate external power supply.
13.3 Both the USB part and the generators with the reclock are fed by two different external power supplies. (The selection of a configuration is done by jumpers)

DSD Audio over PCM frames
DSD64(2.822400MHz) and DSD128(5.644800MHz) by method DoP with 0x05/0xFA markers.
In DSD mode outputs are DATA Ldsd -> DATAi2s, DATARdsd -> LRCKi2s, BitClockDSD -> BitClockI2S.
SPDIF output (through DA101C transformer) – 44.1kHz, 48kHz, 88.2kHz, 96kHz, 176.4kHz, 192kHz (16-24 bits).

Additionally, I developed a DAC PCM5102 PCB to the XMOS PCB for monitoring purposes.
This PCB is made for indicating sampling frequencies. DAC PCM5102 accepts 16-, 24-, and 32-bit audio PCM data, but it can not play DSD signal.

Kind regards

Hello,
Some may remember my problem with DIY android tablet head unit for car from other posts.

Long story short: I get ground loop noise when powering android tablet with buck converter (stepping from 12-14v to 4.2v). I added audio ground loop isolator that made an improvement, but noise was still there.
I bought cheap oscilloscope and discovered that I have up to 600mV ripple when under load on power output of buck converter .
I managed to minimize the ripple to about 30mV with LC filter and thats the best I could manage to do playing with different capacitors and inductors. That also helped alot with the noise.

Now to the point: Left alone the power source, what can I do to audio signal path to quiet down the noise.
If I google ground loop isolator I can see that some of them have resistors and/or electrolytic capacitors. But mine has only 2 transformers (stereo).
I also remember from DIY amplifier project I did a long time ago that someone suggested me to add some resistors from signal towards the ground to bring down the noise... it helped alot... but I can`t remember values...

So what components can I add to the audio path to hopefully see some improvements?

Thank you

anyone reading thank you for your time i need help fixing this amp i got next to nothing for the one picture is of the fried side of the am on the opsing side is the other picture it looks to have the same componets of the side thats fried

Hi! I stumbled across Charlie Mimbs youtube videos a few days ago. Wow!! Until then, I've just assumed that ESLs were out of the reach of DIYers. And, in fact, this is my first visit to "Planars & Exotics". There was something about the word "exotics" that made me think they weren't for me. But maybe they are? And having spent the last few days learning about ESLs, I think I'm starting to understand the basic concepts.

My query is:
Is there a thread here, or a site, with a simple guide for making an ESL that you would recommend? Ideally. one that gives a good description of how to complete the electronics side for both the power requirements and dispersion?

Thanks!
Dan

I have designed a board with a STM32H7 and USB and I2C DAC and ADC. But what clock crystal should I use?


The USB needs 48Mhz, and I want a common sample rate like 44.1khz 48khz or max 96khz. And something that can get somewhat close to the 480Mhz maximum core rate.


14.7456Mhz crystal could have worked is the frequency of the core could get a high value.


Or is a impossible mission? Maybe I can't get a common sample rate and have to resample the audio to compensate.


Any idea?

I have bought someone else's project. The Tonearm has no wires, and has no way to hold the wires connected to the head shell.
Is there a source for the end plug? Has anyone made one before?

Years ago I bought a satellite/subwoofer speaker system (with passive crossover) the sound of which was never completely satisfying. It had good low bass, upper midrange, and treble, but the upper bass and low midrange was never quite sufficient. It had a "cold" sound instead of the warm clarity that I prefer. I obtained other speakers and relegated the sat/sub system to storage, eventually changing the crossover and getting closer to the sound I wanted but not putting it into regular usage.

I am in the process of converting a small barn loft into a combination workshop and sound production studio, and part of this process is installing a couple of speaker systems, one of which is the sat/sub. The loft is finished in painted drywall and has hard reflecting and diffusing surfaces with a plywood floor painted with oil-based floor enamel. Due to space considerations I placed the sat/sub against the short wall of the rectangular room in between some other items which made a sort of niche for the system. I hooked up a receiver and a CD player to run some listening tests and on all the music I played the system had an unpleasant hard sound, almost hornlike in its projection of midrange frequencies and dull sounding without treble clarity. Some instruments on familiar recordings were barely audible and there seemed to be almost no stereo separation, no spaciousness to the sound. The bass sounded all right, but the other effects made the speakers nearly unlistenable.

These effects were noticed at a listening distance of about two meters on axis. When I moved closer to the speakers (inside one meter) the sound improved considerably and had the clarity and pleasant tonal balance I remembered and preferred. "Aha!" thought I, "The room is at fault!"

I looked online at room treatment products available from various sources, and my impression was that they were quite expensive for what was supplied. I considered other materials for room treatment that were not specifically made for that purpose and settled upon memory foam. In my area (central United States) some of the big box stores sell what are known as mattress toppers, made from memory foam. The toppers are simply mattress-sized pads that are placed on top to provide more cushioning and comfort for the sleeper. The stores I shopped carried toppers in both 1.5 inch and 3 inch thicknesses. I bought the 1.5 inch topper which was manufactured with geometric patterns embossed into the memory foam, somewhat similar to acoustic treatment products. The topper manufacturer calls these patterns "Comfort Zones", and I thought that combined with the memory foam construction I might get a useful bit of both absorption and diffusion.

I trimmed the toppers to the dimensions needed and set to work. My setup involves the satellites resting on top of the subwoofer boxes which are hard against the front wall and have sharp-cornered structures on either side. I believe the satellites, which have small baffles of about 10 inches by 7 inches, were interacting with all of the near hard surfaces and refracting off all the corners to give an unpleasant quality of midrange hash to the sound. Basically I used one thickness of the topper material (pattern side out) to line all the reflecting and refracting surfaces in proximity to the speakers and added an extra thickness to the flat surfaces immediately to the outer sides of the baffles. I also rolled up some of the material and placed the roll lengthwise between the satellites (about one meter apart).

This treatment made a very noticeable improvement to the sound. The midrange hash is gone at the listening distance of two meters. The treble has a clarity from being in proper proportion to the midrange. Reverb tails are more audible. The improvement that leaped out at me, however, is the stereo separation. The instruments are now clearly discernible in the mix and their left/right positions are easy to identify. Now, the only question is: Do I want to treat the whole room with this stuff?

This is the first time I have applied sound treatment to a room (and the first time I believe it was needed). I have read many references to the effects the room has upon the sound of speakers, but since I normally listen at close range I had never before heard the room influence demonstrated so clearly. I am a convert!

I would be interested to hear from other members to know if anyone else has used memory foam material as room treatment. If anyone is interested in trying it, the mattress toppers I have seen are available in twin, full, and queen sizes, and I would not be surprised to know that king-sized toppers exist. The larger sizes are more economical as they provide more material for not proportionally more money. The price I paid was under $20.00 USD for a queen size topper.

It worked for me; maybe it will work for you.

I'm an idiot and repeatedly kicking myself. Here's the story.

Earlier this week I accidentally fried my Creek 4330 being careless while swapping out speaker cables. Shorted one of the channels by not noticing that the spades were touching. Loud HUM when I turned on the amp, and I did not kill power soon enough.

Upon disassembly saw that one of the output caps was obviously blown, and one of the output fuses as well.

This morning I replaced that cap and the fuse. Powered it up while it was still disassembled on my test bench, and power came back on fine.

Like an idiot, instead of taking my time (and replacing the rest of the caps as I had planned to) I reassembled everything. (It's not important but my excuse was that the kids were causing havoc and I wanted to wrap up the project & help corral them 🙂 )

Upon reassembly, power would not come back on. When I took it back apart I saw my mistake - I hadn't even clipped the legs off of the new cap before reassembling, so I obviously shorted it on the chassis.

So - what did I blow up by doing this? It's not obvious to me; all of the fuses look fine and I'm not sure how to go about troubleshooting. Any advice (other than to not be an idiot in the future?)

What I finally get about the FF85WK

Like I said I really came into the forum to discuss the 4" but I'm happy to see what you guys think of the setup and anywhere I might be going wrong, unreasonable expectations,etc. I feel like I've given them enough time to get used to a lot more detail.

Brightness might be the wrong word. Sibilance is zero issue. Cymbals and hi-hat sound great. The upper part of female voices sounds great (upper chest, neck, voice box, enunciation). Perhaps brash is a better word. I knew they would be a bit forward and perhaps that is part of what I hear, but I feel like I'm separating out the forwardness/shouty from my perception of the sound signature.

The other night I listened to them (critical listening) for about six hours. Not full songs, hopping around while playing with running them with EQ, wide open, different volumes, sub in, sub out, X-over points, sat back a bit, etc. At the end I was disappointed with how they had worked out for me. (not blaming the speaker)

The wall to the right of me could probably use some sound absorption which is about 2' away from the speaker. But other than that the room is pretty much mixed surfaces, carpet on the floor. The Speakers are about 8" away from the (bare) wall behind them. The room is about 10x12. The back wall is about 6' behind me. But I'm not trying to fill the room. But I did notice that a good solid foot back from where I sit there is a sweet spot where they sound the best. I have tried pushing them back closer to the wall, and further out on the desk to get them further away with mixed results.

Outboard gear

I'm using the usb out of my computer to a Topping VX1. The headphone out of the topping goes to the line in on the sub. It's one of the reasons I got the VX1 is because it can output both speakers and headphone at the same time.

TOPPING VX1

The drivers are in ufonken cabinets, exactly according to the plans. I was very careful on all steps especially fitting the brace. Touching, but not too tight. I put in all the deadining material as close to the plans as I could.

They are on a monitor bridge and then on 4" stands that weigh 1lb each. As I sit the speakers are just about ear level and I have them facing straight out into the room in a roughly 3" equilateral position with me at my desk. I have played with the position a lot. Though they sound pretty good pointed dead at my head.

The sub is built around the TB 5 1/2" sub speaker built in a 9l sealed box. F3 was about mid 70ish. I forget the Q but it was in the .7s

I crossover the sub at about 160hz. Though I've played with it anywhere from 180 to 120. At lower volumes I like to feed a bit of lows to the speakers. Trying for that balance I guess that Bob mentioned.

I mostly use Foobar player using the WASAPI driver. Files are 192kbs or better. I have a folder of all 320k and Flac that is my 'library' for foobar.

That's the story and I appreciate you guys taking an interest in my enjoyment of the speakers.

Not sure if this is the correct address to post but was looking for some recommendations for some in ceiling speakers that will provide near hifi sound (if even possible) that will be driven by one of the tpa d amps else where on this forum?

Sent from my LG-V510 using Tapatalk

Hello everyone. I didn't want to back trace my other post. So I hope everyone can chime in and help me. I will list what my friend bought for his project and go from there.

1. Drivers. Fe206an drivers ( has ).

2. Vifa ring tweeters ( peerless ) XT25BG-60-04 1" inch tweeters.
( has ).

3. I am buying the wood. So that's takin care of. ( ( getting ).

4. The amp and pre amp I can get a model number if it would help if needed? ( has ).

5. Subwoofer will be add later this year's or the beganing of next year.

So the speaker wire and binding post are easy to get. The stuffing He needs to get as well. I have all the tools but, I am limited in time and design. So that's where I need help. I did research and there is a ton of designs for this drivers. I am a one man show and I would like to do a simple ported box of possible? A subwoofer is in the future plans for sure. So If the ported box doesn't work out I wonder if the mark audios pencils could be modified to work with the fostex fe206an drivers? Just throwing things out there? So that's all the info I have for now. If anyone could help me that would be great. Thanks Jeff

Amp is using BUZ102 for the outputs .

Anyone know of a good replacement for them ?

Morning!

Just finished the build of the enclosures and am listening to them now.

Have done some rootamentary damping, wondering how folks have approached theirs?

Using Fostex FE108Sigmas in the enclosures.

Thanks for any thoughts you might have to share.

P

Great sound with these dacs chips.

R2 40€
S1 Single Crown 90 €

S1+R2 110€

shipping costs from Spain by the buyer.

I am doing research for an amplifier that I hope to eventually build. What I have noticed is something called, "damping factor" in specs of some amps. Is there something somewhere that is written that explains: what it is, what is a good #, bad #, and especially, what in the amp determines damping factors, and how do design this into my amp. Any help would be greatly appreciated. Thanks!

This circuit is inspired on two background ideas. One is from John Linsley-Hood's ClassA 10W design, and later modifications; the other is from Mullard's 10-15W Class A circuit which was published in the second edition of their Transistor Audio and Radio circuits, which showed a differential driver to a Darlington output and driver pair for each output half.
The similarities are slight. This design uses a more modern approach, with a differential input stage, a high impedance transistor pair which sets the current in the differential driver-VAS combination and a driver-output transistor pair.
The quiescent current in the prototype was set by choosing component values after measuring the gains of the output and driver transistors. I happen to have obtained high gain 2N3055's from ON Semi, so as this is a development circuit, it may be that other devices will need adjustment. The easiest approach would be to use a pot in the CCS of the input stage.
Simulated distortion was 0.002% at 1kHz.
I measured the prototype with a bias current of 1.5A and supply rails of 20V at just before clipping (which turned out to be due to the PSU with a hard clamp at 2A; will repeat with capacitor filters) at 0.005%. Circuit is shown with distortion image.
Temperature stability of the bias current is a question, but because the driver stage is set by the Vbe's of Q3, Q4 then as the unit warms up the current will actually drop, so essentially safe. But there will be a signifcant change as the transistors warm up. These are mounted on a 0.5K/W heatsink, but so far the current has not shifted too much.
More info as I get the results later.

Hello all!

I got one of my crazy ideas again.

I have a few R085 to play with, but have mostly stayed clear of them because of the negative bias supply requirement.

I get that one solution is to place the R085 in the top of the output stage like the DEF amp. That, however, requires it to be run in common-drain mode which as far as I know would drop the gain to 0 and also have a low output impedance (I would prefer a single stage amp and I also want a current source amp with high output impedance).

If I remember correctly, however, in the F6 Nelson uses the transformer in a nifty way to run the upper N-channel mosfet in common-source which got me thinking...

Does this trick work in general or is there something specific with the F6 topology? Could I build say a F2-ish amp but put the CCS on the bottom, a R085 on top and use this trick with a transformer to make the R085 operate common-source? Or dive deeper into crazy and use a P-channel mosfet on the bottom and configure it as a mu-follower =)

And if it does work, would it perform comparable to a normal common-source configuration with a negative bias supply or would it be inferior in some way?

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I just traded a bike for someone else's old DIY project. I got some Fostex FE206EN with the horn cabinets suggested in the specs. One driver plays slightly quieter and has a mechanical sounding buzz or rattle in the mid range. I have cleaned contacts, inspected and compared both drivers, swapped cabinets, but the issue remains on the single driver. I've never dealt with a whizzer cone arrangement before so I'm treading lightly.

I'll be back tomorrow but tonight this has me whooped.

I have a sweet peach amp (FU-50) and it has been working fine. I started having two issues...

1-getting popping sounds
2-the right channel started going out.

As for 1...i noticed i could clear it up if i tapped on the center tube. I took it apart and found that the center tube socket was pulling out of the circuit board...ie the pins of the socket were pulling out of the solder joints. So i cleaned up and re soldered the socket and that solved that issue.

As for 2...i did not see anything unusual but did notice that i would hear audio out of the right channel when i first turned it on for a second or two and when i switched it off...also if i left it running for a while it might pop in. I did try swapping the two large tubes and also some other ones i had. It did seem to follow some tubes. When i had a pair that seemed to work, i notices that the right channel tube was nuclear hot...visible glowing red significantly more than the left side...also if i swapped two working tubes...what ever i put in the right side would glow like rudolfs nose. (see photos of the two sides compared)...i also noiticed that the two circuit boards used for the two large tubes were populated differently....the right one, (the one am having trouble with) is missing R2, R3 and has no ground wire soldered to it....is this normal? (see photos).. Any help would be appreciated...i am afraid to use this now.

In one channel I mistakenly used a 2sk1530 instead of a 2sj201 on the minus supply.Switched on and of course got no sound from that channel.Replaced with correct 2sj201 but still getting no sound from this channel.Have I damaged something else in the circuit.

Dear audio enthusiasts,

I read a lot here and enjoy the brilliant expertise shared by the forum members.

Recently, I came across an interesting thread discussing one of the countless variants of what I just called the Hitachi style push-pull VAS in lack of a better term.

Keantoken presented his design and BesPav suggested a few improvements: Keantoken's Aurum-X 300w Amp with LatFETs

I don't want to hijack the other thread as the decision was to basically stick with the original design and not to implement most of the suggestions.

Using a minimized model amplifier, I simulated the suggestion to replace the tail resistor of the VAS LTP by a current source and while this is an interesting idea, I can't see the big improvement, but new problems coming with the CCS.

Below you can find screenshots and also the LTSpice file.

Did anybody succeed with using a CCS for the VAS LTP?

What would be the possible improvement?

Best regards,

Lee

Hi guys does anyone have schematic on this amp i got it like this from previous repair