Hello everyone.

I was looking for solution to fix one issue with my speakers and google said 'look at diyaudio'. Well, I checked the topic and, honestly, I don't know if it can help me, but I decided to post a comment. Since it doesn't let me post until I introduce myself, I decided to make a post here. To make things looks even worse for me, I created this account 4 years ago, but my post count is zero. I guess I tottaly forgot to introduce myself back then. 🙁
Even though I probably deserve to be kicked out for such thing, I hope that will not happen. I'm sorry guys!

I'm kindly asking if you can allow me to post comments on this board.

Thanks!

Do you have such a module,I want to buy it.

Need a pair of 2.2 uF film capacitors for tweeter filter.

Voltage rating lower the better, new or used.

Paypal f&f, or if you want to trade, I have a ton of great film capacitors in 0.22 uF range and a ton of great tubes (12AU7/5814, 12AX7/5751, 12AT7, 6SN7(yikes), 5U4G, EL84/6BQ5).

The range of quality I'm looking for may include..:

Audience Auricap XO
Clarity Cap CSA, MR, CMR
Audyn true copper
Jantzen superior-Z or above
Jupiter or Miflex copper foil
Mundorf silver/oil or above
Rike audio Q-cap, S-Cap

Thanks!

Hi, I am a Pysics professor, now retired.
By several years I made amplifiers from scratch, including pcb design, following several diagrams, includind the famous Quad 33/303. Usually friend wanted to buy these amplifiers, and no one failed...
This was a long time ago. I'm now almost 74 and my usual amplifier is now the Audiolab 8000A.
Regards.

Hello, my name is Mihai, I am here to find repair info for some car navigation unit I have been working on. Used to be interested in audio some years ago, built some loudspeaker enclosures, power amplifiers, etc 🔉

Nice to meet you!

Thought this is a good idea for members that just need a few parts without building an order to meet the minimum requirement for free shipping.

Anyone here can offer to add to their order or request parts to add to someone's existing order from Mouser / Digikey / Farnell

If you need a few caps or resistors, anything, let me know, I will be ordering soon.

🙂

Hallo guys,

this is my first thread after reading a lot for years on this website. So thank you for all your input. I dont speak english very well, so sorry for that. Now I want to design a new speaker and it should be my last one (at least for some decades)...

Points to consider:

• My room is 5,5x4,5x2,6m. Listening Distance is 3,8m. My avg. (for me) loud listening at 3,8m is about 75-80db avg. with 85-90 db peaks. So the speakers should produce around 85db. avg. with 95-100db peaks, to have some headroom.
• Musictaste is very wide. Only thing I dont listen to is stuff like heavy metal, large orchestras...
• It should be a 3-way. No 2-way. No Di'appolito.
• It should be activ with dsp.
• It should allow to swap amps (so no Hypex but miniDSP)
• It should use the T25B and M74B (I already have them)
• Passiv Notchfilter will applied to lower distortion of the resonance peak of the M74B
• It should have a nice dispersion, but dont use waveguide (because my son is 1,5y old and I dont want to have beryllium exposed to him (atm using a TW34BWNG in my speakers that have to go because of it)
• Room is my homeoffice, music, homecinema, gaming room. So room treatment is an option. A little bit is done already and I will continue.
• It should looks like build by a company, not a diy dude. Similar to Rockport Orion
• It should cost overall arround 5-6000€ without amps.
• Solidstate Amp for woofer and Tubes for the Midrangdome and the Tweeter.
• It can be a bigger floorstanding speaker but no refrigerator.
• I want to do some fancy stuff, even if it wont help much (Tuned Mass Damper at the top of the speaker)
• Building time should be arround 6-12 months.

Tweeter is BlieSMa T25B crossed at 2750hz LR4 passiv.
Midrange is BlieSMa M74B crossed at 520hz LR4 activ (considering my needed max SPL and LR4 I think its fine).
Woofer will be most likely the new Purifi PTT10 crossed activ. Closed.
DSP will be miniDSP Flex or Flex 8 depending if I go with hybrid crossover or not.
Amp for the Woofers is a Accuphase 303-X (I inherit from my grandpa, so selling it, is more or less no option)
Amp for the Midrange I consider Tubelab SE with single-ended 300B
Amp for the Tweeter will be the same (if hybridcrossover) or a Tubelab SSE with single ended KT88 13-15Watt (to come closer to the max SPL of the M74B with 7-8 Watts from 300Bs)
I have a UMIK1 for measurements.

I dont know if I wanna go the Tube route with a Hybrid Crossover or not... Or just use 3 Amps... one less amp is tempting...

The plan for hybrid (of couse can only be finaled with enclosure build and measurments taken):



The plan if going full activ:



The enclosure plan:






Volume for the Purifi is exactly 24,8 L like Purifi datasheet and WinISD consider for 0,7. (You can get the mm² from sketchup for every object and than get the volume out of it. Volumen displacement from the Purifi is also considert.)
It maybe look like the enclosure for the purifi goes all the way to the top. But there is a plank under to M74B without any holes.

The bracing you can see in the pictures.

The front is in total 30cm wide. At the mid of the Midrangedome its 142mm wide and at the tweeter 88mm wide. C-T-C speacing between tweeter and midrange is about 1,2. (I follow some of HifiJims (and his friends) recommandations, I found at diy audio.

It will be build ot of 22mm good quality MDF and all parts will be glued with SikaBond T1 Purform. I have a fully equipt woodshop in my basement with a format circular saw, routers and so on. It will take time to do all the parts, but that's what make the design possible for me. You can print every piece in Sketchup in 1:1 and than use a router and flexible curve ruler to get it perfect. Than use the finished one as template for the same pieces.

The Tuned Mass Damper at the top will build out of steel with a total diameter of 110mm. It will be damped with small adjustable oildampers you can buy for RC cars. And it will be adjustable in height. I consider to make it out of aluminium and hollow to adjust the weight with damping oil. But this will increase the costs and chances are, the oil will escape.

I want you to ask, if my design is completely ******** or are there some mayjor design flaws?
Should I skip the hybric crossover and take a miniDSP Flex 8 with 3 Amps instead and for what reasons?
What I could maybe improve?
I just tried 2 passiv Notches for the M74B like found on this forum, but cant see much improvement over just one with slightly diffrent values. Is there a reason to use two? The Peaks are to close in my option to get much benefit out of it. Peak of resonance is -54 dB down at its highest point.

Thank you very much, be nice and greeting from Germany.

My name is Hans en I’m working as a linac engineer as professional.
During my life , I have build a few amps. But now will start with a tube amp.
Hopefully you can advise me if necesseray.
Kind regards, Hans

This is about an AER Compact Mobile2 acoustic guitar amp

Recently it started to blow fuses after a few hours of playing guitar. Until a few days ago that it started to blow them instantly when powering it up

There’s a mains filter inside the amp that it is shorted (fried). Bypassing it the amp plays just fine. But I don’t know if I can find the same one

Should I replace it or leave the amp without one? Is this a common 250V 4A mains filter. Do I have to find exactly the same one?

Hi!

Been looking a bit into building a high efficiency two-way with a 1st order filter. In other words, as simple as possible.
Are there any good HE drivers out there that works well with a 1st order filter?
I see Zu Audio are doing it with a 10" wideband/full-range.

Maybe a Dayton PM220 wideband woofer with a Dayton AMTPRO-4 or something?

Any suggestions? 🙂

High output.
What do you have?
Cash or swap for several items I currently have for sale:

AE TD10m drivers x2
KEF R8 META
AE TD15x Apollo X2 (in sealed cubes)
KEF 3005SE.eggs x12

(Red)horn response, (purple) is the mic @ the vent exit and some random irrelevant spl.

This is a new thread as suggested by Moray James. For Moray James: What I did was to simulate the enclosure with two speakers in Horn Response. I simulated it as you would an Altec A7 type of speaker but with a front chamber on the horn. This front chamber would be the plenum if I am doing it right. The plenum opening was the horn throat. I rounded the opening a couple of centimetres and used this as the horn. I did do the combined sim to take into account the vents. As I said in the other thread, I don't know if this is anywhere near accurate.

Any input from those who have done this type of design would be most welcome.
jamikl

In the near future, I'm going to put up for sale some small (4" full range) drivers. I don't have the original cardboard. What are some good ways to pack them up?

Hi all, after some years (10ish) I decided to revive my first hifi set. Sadly my cd-player didn't stand the test of time very well... So that's why I'm here to see if I can can everything back alive. I'm looking forward to it!

Hi to all from the RSA !
Does using layers on pcbs create a capacitive condition ?
If one layer is positive power supply and another layer is negative power supply, should they be on the same side or on opposite sides of the pcb?
Will they interact with ground plane layers ?

Thanks in advance for your comments !!

Hi,

I have two sets of PCBs to give, just pay shipping and they're yours.

1 - Set of VSSA Concept from Andrej Lakner, pcb designed by Jason Kuetemann

2 - Set of AKSA Quasi comp Mosfet by Ranchu

Best!
Do

As a newbie, I would like to introduce myself briefly:

I have been a HiFi enthusiast since 1978 after buying two Pioneer devices (SA-9500II and TX-9500II)!
The two devices formed the basis of a collection that has since grown, mainly of Pioneer devices and a few other manufacturers.
As a long-time electronics hobbyist and a qualified electrical engineer, I see myself as being able to repair everything myself and, if necessary, develop everything myself.

At the moment I am busy with a replica project "TAD TSM-2" and that is exactly why I ended up here.
I hope to find some additional information or to be able to give information to others!

I look forward to a lively exchange with you here in the forum -
best regards
Pioneer_Exclusive
(Juergen)

I want to measure the parameters of a lavalier microphone like this one:



The data I am most interested in are sensitivity, s/n ratio and frequency response.
To measure the frequency response, do I need an anechoic chamber?

What particular equipment do I need for a scientific measurement?

Thanks.

Witajcie członkowie diyAudio!

Hi everybody
I recently got a vintage amp and have been using it for a couple of months now. It works and makes a very pleasant sound to my ears. Though there is one concern. Almost all the time I hear that one channel is slightly louder than the other. Of course, it may be my hearing. I did the typical cleaning of the unit, including the balance and volume control pots and bias adjustment.

Question: Can I use a tool/equipment to measure the actual levels of the signal coming out the amp to both channels, to confirm or negate my subjective and personal suspicions?

Thanks.

I have been very interested by Merlin book on Tube Preamps and I decided to build the RIAA amplifier, however I decided to replace the cathode follower in the output stage by a MOSFET source follower (I already used this solution in a Baby Huey amplifier with a very good result) to save one tube and because I wanted to avoid the big voltage difference between the heater and the cathode of the output tube...
As I don't like point to point soldering, I have started to design a PCB. If someone has any suggestion about this project, please let me know.

Do you know if the PC900 can also be a good device for this project ? The pin-out is not compatible however 😡

Cheers,
Marc

Looking at some photos of open baffle stuff and this one comes up... One calls it slot loaded, another calls it horn loaded.

Technically I suppose I'm curious of:

• what the ring type relief is called?
• does the phase plug simply avoid the flat slapback of leaving that central portion untreated?
• is this slot or horn loading?

Maybe it's neither etc, I was just curious "what the folks at diyaudio called it" etc. Thanks

-SOLD-

Built using full kit from ZenMod with Antek AN-10425 transformer, Randy’s W12 CRCRC PS board, DIYAudio Softstart, WBT binding posts and RCAs, KimberKable wire and custom 4U 500mm chassis with UMS hole pattern. Sounds fantastic. Only selling because I purchased a PassLabs Int-25.

Asking $1,000 with shipping within continental US included.

Hello everyone,

I put some effort into designing an output filter for a class-D IC. Here I am using Coil Craft's and Murata's SPICE models to calculate the filter with an 8 Ohm load.

Lines:

• Red line is the capacitor impedance
• Green line is the inductor impedance
• Purple line is the filter response

Filter:

• -3dB cutoff is around 25kHz, just outside the audio band
• The filter works until about 10MHz where the parasitics of the components have a resonance.
• Sure the values can be tweaked, but this is not a bad starting point for now.

Post filter feedback

The IC that I am using TAS2780 has a post filter feedback (PFFB) feature and I thought that would be good to use to keep the THD as low as possible. So I started reading about the PFFB requirements.



It seems crazy to me to specify the filter cut-off frequency to be greater than 10MHz when this device is switching is 380kHz.

What could they mean?

1. The rise/fall time of 1/10MHz = 100ns. So maybe if the rise/fall time of the IC is around 10ns or less, the filter could filter that out?
2. The cut-off frequency they mean where the parasitics break the filter? This is around 10MHz for the L and C parts I am using.

They have an application report SLOA260 that was not that helpful, if focuses on using ferrites.

Questions

1. Can anyone experienced with Class-D and filter let me know if a -3dB cut-off just outside the audio band is sensible?
2. Has anyone used TAS ICs from Texas Instruments before, or the TAS2780 specifically? Can you suggest a sensible filter with or without the PFFB feature.
3. What the hell are they talking about in the data sheet and application report when they specify the cut-off must be greater than 10MHz?

Hi All,

I am going around in circles trying to decide how to proceed on my next build. I have a pair of lovely TAD2001s coupled to a beech plywood tractrix horn that I have made - very happy with the sound at home. Below it I am using a Supravox 285 in a ported enclosure as a 2 way monitor style. They seem to work very well together but the low end is limited in extension and the output of the woofer is slightly limited for high spl fun now and again albeit super detailed and fast in mid bass. The crossover is 2-way 2nd order at 800hz - that suits my horn fc and what TAD recommends. Using active crossover right now.

So I am now thinking how do I take this to the next level to accommodate more output without ruining quality but also being durable. I was considering either going for an alternative single 12” driver or building a new cabinet with MTM design using twin 12s per side OR going for a single 15” like the JBL 2226H. I did loads of Hornresp simulations of different drivers but can’t make up my mind. The 2226H in every sim I did didn’t really get adequately to 800hz. 700hz looked max but I realise many use it above that.

I do want to be able to play this system really loud on occasion so durability of woofers is also important. I am not too fussed about content well below 40Hz as I can augment with sub. Importantly I do want high efficiency.

Any advice much appreciated!

Anyone in the UK got a spare pair of DIY Audio Store Universal Power Supply they're willing to part with?

Hi y'all! New here but not entirely new to DIY audio projects.

My friends and I are looking for replacements for Toshiba 2SC3381 and 2SA1349 BJTs. There are forum posts with single package replacements over the years, but even those replacements are out of production. I'd like to avoid eBay or sketchy transistors if possible. If I'm not mistaken, the parameter we are having difficultly with is the DC current gain (hfe) classification we're looking for a minimum of 350.

Any help is appreciated, thank y'all!

When I have reached retirement, I can finally dedicate myself to diy audio, which I have been doing for 50 years now, especially in speakers, both on magazine projects and designed by me, also thanks to the forums on the web.
To write in English I have to help myself with the automatic translation of the web, so sorry for mistakes.
I have also been passionate about environmental acoustics for many years, because of a friend acoustic engineer specializing in concert halls, acoustics in public places, companies etc.
I finally made a special listening room of 7x5x3 meters designed and evaluated by him with a final written report. This is because for years I have been convinced that 50% of the final quality depends on the environment, especially for the bass and high levels of listening that I like with the right music.

Again greetings to all.

Rack Case, front aluminum panel size 3x17in.,box depth 12in.
Sale $28.

QA401 in excellent condition.
EU only
330Eur + shipping

Hello,

Im new here, hope this is correct subforum. I plan to test my working integrated amplifier Marantz PM-750 DC, want to try their preamp.
I already research that its doable. I want to make sure, is this the correct signal path? Here is the zoomed schematic, and below is full schematic.
What i read is right before power amplifier board to make an output.





Red is right channel line
Orange is left channel line
Blue is where i plan to cut.

And second question, if i plan to test the amp, should i put E-CAP bipolar 10uF 25v in series each channel, just like what i see in my Marantz 1180 DC right after "MAIN IN"?

Thank you

I'm designing something in which audio plays a small but vital part. A DFPlayer module will be used to play a short audio file and the output will feed an external PA system. The easiest way to do this is to use one side of the BTL outputs as a Line-Out rather than create a balanced-to-unbalanced converter

With this arrangement, one output pin will be open and the other pin will face an unbalanced high impedance load to ground. Is there any possible issue with this arrangement, in terms of stability and distortion? None of the documentations I've seen about DFPlayer is very forthcoming with this kind of information.

As the DAC nearly newly released maybe somebody will have benefit for it. There was error during asembly and many (I checked photos in internet- same error) or probably all cdp.3 have wrong capacitor soldered in filter - 15nF instead 1.5nF. I accidely found after tried listen to it. high frequencies was rolled of quite badly at 10khz already -3db

once replaced sound become transparent and airy.

i have expierended same ~2011m on m2tech evo usb conveter were highs had -3db at 15khz and 1.5db at 8khz

hope this helps

Ok as the title says I have the red Sanwu tda7498E version from Amazon. I soldered the 4 jumpers together on the bottom of the PCB and input a signal into the LEFT RCA only from my low pass filter preamp. I also put both dip switches that control gain to the max setting of 36db and the output level is approximately half of what I get in btl mode! I usually use this amp in btl with (2) 4-8ohm 2-way bookshelf speakers and it gets very loud. I measured 26vac average clean with my oscilloscope while using a 34v 10a PSU. While the amp is in PBTL I'm using a 4ohm GRS 8" high excursion sub from parts express. What's going on, am I doing something wrong?

Q: Can tubes (2 final tubes in a pre-amp) share the same DC battery filament supply, assuming sufficient current and voltage from the battery? This would be using LED cathode bias. Or, I could go with filament bias. Not sure if there will be crosstalk between the tubes as they share the same DC supply. Thoughts?

Can anyone help me identify any problems with this TT DHT - looking at the silver grid, it seems 'blackened' (maybe heat/current) . Is that normal?

Tube is a Hytron 210

I am trying to mate some SB Acoustics drivers and passive radiators together. They are literally made for each other, however, when I enter it into WINISD it is simulating what I assume to be an improper frequency response. Part numbers below:

Driver:

5×8″ SB15SFCR39-4 / Paper
https://sbacoustics.com/product/5x8in-sb15sfcr39-4/​

Radiator:

5×8″ SB15SFCR-00 / Paper
https://sbacoustics.com/product/5x8in-sb15sfcr-00/​

These are small 5x8 drivers so I would think they wouldn't require a huge amount of space.
Can I get a sanity check? Yellow is the passive radiator. Blue is it vented. The vented looks great. I would think the passive radiator made for this driver would look similar.

Greetings! Thank you for the opportunity to share the audiophile journey with the DIY Audio community. 🙂

I love these repair/ restoration videos!

Login to view embedded media
A lot of people do them and they are becoming more popular, makes everything seem easier than it is.

Maybe I am getting old and don't like the new stuff anymore?

I'm making a headamp and I'm using a 1mm steel enclosure. I want to use wood for the top and am thinking of "attaching" it to the enclosure using small magnets.

Rationally this shouldn't be a problem but...will the magnets cause any issues in terms of EMI or something else?

Thanks in advance.

I'm looking forward to learning from this group about recapping older speakers and designing and building transmission line enclosures for full-range drivers.

The Hawaii, has left her home on Vancouver Island for the final time.
https://www.cbc.ca/news/canada/british-columbia/hawaii-martin-mars-last-flight-1.7291472
I had the pleasure of visiting the second to last unit, and parts of others, (the only flying one was anchored off shore), on a trip to Vancouver Island about a dozen years ago with then still somewhat spry Father. He left in 2015 but having him there, that day, that time, is something I am so glad we did. We had a couple of extra hours before the ferry took us back to the mainland, so what better way to spend time with your aging Dad, than doing things that had a big interest for him.

I did not want to be the 'Cat's in the Cradle' son.

Sorry for the picture quality. Maybe, it's just me, getting a little blurry eyed. 😉
She really was the queen of her skies.

I heard from a discussion about driver measurements. The topic was about generating xover filters, and about the best compromise measuring distance. I consider this an interesting playfield, and this motivated me to perform some simulations. It’s all about anechoic, direct sound and the influence of the measuring distance on the baffle diffraction step (BDS).

Assume a linear monopole point source on a circular baffle. When the pressure wave reaches the rim of a baffle, its’ room geometry changes from half-space to full-space. This is a change by a factor of 2.

In the time domain, this change theoretically will cause a negative impulse. As the room geometry changes by a factor of 2, the magnitude of this negative impulse might be half the amplitude of the initial positive impulse. In the frequency domain the response will show the well-known BDS and some comb filtering towards higher frequencies with a first minimum corresponding to the baffle width. This is widespread common knowledge.

Most models and graphs for the BDS assume an observation point in infinity. For the infinity case, in time domain the negative impulse occurs after the time needed for the soundwave to propagate from the centrepoint of the baffle to the periphery.

Instead, theoretically approaching the observation point to null distance from the baffle plane into it’s centre will double this time: The sound has to travel to the periphery first, and then return to the centre where it gets assessed. In time domain the negative impulse occurs delayed by a factor of 2 compared to the one assessed at Infinity. In frequency domain, this leads to a combined squeeze/shift of the typical frequency response along the frequency scale. The first minimum of the comb filtering pattern will occur one octave lower than in the infinite case.

In analogy to these both extreme cases, it becomes evident, that even without taking any room response (such as first reflections, echos and the like) into consideration, measuring at 1m and listening at 3m will not be the same in terms of frequency response. Therefore, relying to a measurement from 1m distance to optimize a set of xover-filters, and then later on listen to the result at 3m may lead to a mismatch between the filters and the actual frequency responses and thus to an awkward and unwanted experience.

Let’s do some simulations to explore this one. For the following graphs sampling rate was set to 192kHz. For the first three graphs the baffle diameter was assumed to be 34.3cm which corresponds to a wavelength corresponding to a frequency of 1kHz.

Curves show measurement distances of
green: 0, theoretical measure flush on the baffle's center, e.g. spot-on the point sound source
brown: 1 == 0.343 measuring distance (in baffle diameter length, 1 == 34.3cm)
blue: 2.9 == 1.0m measuring distance
grey: 8.7 == 3.0m measuring (=hearing) distance
black: Infinity theoretical measuring distance
Red individually shows the resulting magnitude difference between two test distances.

The time domain graph. The negative impulses of the baffle diffraction occur with different delays from the initial positive impulse:




Frequency domain: Red shows theoretical null distance vs. infinity BDS model distance




Frequency domain: Red shows the frequency response difference between 1m vs. 3m measuring distance:




Frequency domain: Difference 34.cm vs. 3m measuring distance … measuring very close is no good idea at all …




And now, what about a more narrow baffle at 17.15cm diameter, e.g. half the size of the former one?

Time domain for 1m vs. 3m measurement:



Same 17.15cm baffle, but in frequency range
Red 1m - 3m
Blue at 1m (5.8 times the baffle diameter)
Grey at 3m (17 times the baffle diameter )



Small baffles, as expected by geometry/trigonometry logic, show less severe shifts at the typical measuring/listening ranges of 1m/3m than large baffles. It’s evident, because relative to the smaller baffle width, the distance/bafflewidth quotient get bigger. By the way, this might be a contributing reason why baby baffles sound more pleasing at any listening distance than adult ones.

Caution: Take all these simulations all with a grain of salt. These graphs show a theoretical workout of the subject, highlighting a possible issue. In practice you will ideally not have to deal with central point sources on circular baffles. But the take-home message seems clear: Do not measure too close in a false idea to best possibly exclude room interferences. This will jam your measuring validity range. The closer you measure, the messier the result gets for more distant locations. Same goes not only for x-over filters, but also for whole speaker system measurements.

Does anyone know of a good headphone amplifier project or kit that is based on a discrete A/B output stage and includes negative feedback of the output signal in order to reduce crossover distortion? I need something beefy as my target is to drive some demanding planar headphones

Hello everyone, I have looked up this old post and I want to do this on my TNT first generation turntable: https://www.diyaudio.com/community/threads/diy-vpi-based-turntable.185469/

Anyone know the whereabouts of this turntable? Also, the materials to build this turntable?

Thanks,
Ken

Hello I'm Bazza
I've just bought the above amplifier and really looking forward to having allot of fun playing around with it however, I have never owned or used a quad amplifier. No user manual. How do I connect a source, adding speakers/amps and all the other amazing options available.

Dear forum,

It is not a new idea, but everywhere where I saw something like this in the internet, it was never accomplished and the thread just ended up with no conclusion.

I want to raise a topic of a Guitar pickup system, where it is possible to connect them in any possible way (parallel, series, out of phase) with the help of a microcontroller onboard. This was done ones commercially by a Music Man company in their Reflex Game Changer guitar. I was thinking on what it takes to re-make something similar.


QUESTIONS

1. As the guitar pickup output has high impedance, and in general guitar output has a high impedance, when wired canonically of course, I am very concerned about preserving this high impedance (if it is at all possible). As I understood, the analog switch, even with the lowest on-resistance will affect the output of the pickup and lower the output impedance.
   Q1: is it indeed like that and it is not possible to carefully select such components, so the original impedance of the pickups is preserved?
   Therefore, I see recommendations to use a buffer, e.g. op-amp, between the pickup and the switch, so the sound characteristics are preserved, but the signal is then "transformed" into a low-impedance environment. As I understand, this is a compromise to use digitally controlled switches. Moreover, some people prefer the guitar to output a low-impedance signal, as it eliminates a lot of problems with long cables, nasty guitar pedals, etc. In other words, there's an opinion, that low impedance is not bad at all.
   Q2: if the buffer is needed, is it enough to put it after the pickup before the first switch for every pickup? Or does this chain needs more buffers?
2. Buffer: I know there are a lot of buffer options with JFET and other transistors. I think the choice here is: as smaller as possible, as less components as possible, the best sound preservation. For me it looks like Op-Amp is the option to go, even if it can be costly.
   Q4: what do you think?

I've been lurking around for quite some time now, I guess introductions are in order. As the title says, I'm from sunny SA. Got my HiFi feet wet recently when I bought a secondhand pair of KEF Coda 7's. I'm powering them with a Aiyima A07. The plan is to upgrade all of that as time allows. I enjoy listening to blues and a bit of jazz but still have a lot to learn about music.

I picked up a nice looking old 70's Kenwood that was missing parts, so I retrofitted it with a new PS/amp/preamp and I'm trying to come up with the best way to power the legacy tuner/phono stages in the unit. I'm very tight on space so would prefer not to add another transformer.
3 voltages are needed, +28v, -19v, and 13.5v.
I was considering an lm317HV/lm337 connected to the main +-40V supply to derive the +28v & -19v.
For the lm337 I saw this thread which uses a zener across the input-output so you can use it with higher voltages.
For the +13.5v I was considering using the positive half of the preamp supply but wondering if pulling from half the supply is a bad idea.
I'd like to keep the legacy components functional, even though realistically they will probably rarely if ever be used.

I have two unused Elna CERAFINE 12000u/80V.
Cream of the crop in lytics.

$200 + shipping at cost.

R

I got scammed a few weeks back on face book market place.

Recently I found my email address and password got hacked.
Stupidly I used same password for lots of different websites.

I managed to get in quick with ebay and changed my email address and password before any money was taken.

I got caught out on Ali Express as theye didnt ask for cvv number before making a payment.
I was down about £300 which HSBC are sorting out for me. Again got in quickly to block the card but still some damage done.

I guess the lesson is use different passwords for each website you buy from or use.

Since I sell this stuff for a living, , I use a lot of subjective terms/descriptions of what I hear. One of those is "transparency" and I wonder if this is in fact a low noise floor?

MSB describe their amps as having a class A gain stage that provides all the voltage gain, followed by a class AB stage that only supplies current. This sounds a lot like a current dumper design. Anybody know if that is indeed the case and has anyone heard them ?

Hi everyone I'm triyng to repair an harman kardon ts2bq/ts11/I don't know how many name this subwoofer has...
The first problem was literally a hole big such a finger in the amplifer board with shorted mosfet and blowed up capacitors. After having fixed the amplifer board I've tied to turn it on, the results are two shorted mosfet and another problem.
After several tests with the tester I'm stuck, if I put the tips in the positive and negative of the diode bridge for a few seconds the tester tells me that there is a short circuit, after the value continues to rise up to 1560, after a few minutes it goes back to how it was before. I did these tests by removing the diode bridge (to check that it wasn't that) and removing both the amplifier board and the jack board.
What component can have this behavior? Could someone help me?
This is the datasheet

Thanks

I try to fix what I can

The cheapest Martin Logan ESL is a good value for the money. Can we make it a little better with a little effort? These are the things I successfully tried:

1. Linear power supply. You don't even need to buy that fancy 'audio' power supplies. A normal, stabilised, linear (with a transformer) power supply also helps a lot. I tried something for $10, 15V, 1A, with a 5.5/2.1mm DC plug, plus in the middle.​
2. Bass is probably the part where ML ESL is the weakest. We can at least try to mitigate some problems here. Adding a weight to the speakers could help. I had granite coasters laying around, that almost fitted perfectly. Also, it seems that the enclosure has some resonances, so I added self adhesive bitumen pads inside. That was very helpful for quality of the bass. When being at it, changed the filling to wool.​
3. Unfortunately I couldn't place the speakers far enough from the wall. The back wave was messing the sound too much. Ideally it should be diffused behind the speakers somehow. Probably with some acoustic system on the wall. Anyway, I couldn't find anything aesthetically pleasing enough. I had to damp the back of the panel with an acoustic mat. It worked well for the purpose.​
4. Next step was checking if bypassing the caps in the crossover can be beneficial. The result was also promising. Caps bypassed with 1% of the original values. It helps with resolution and smoothes the highs.​

Do you have any other modifications that can be easily tried?

Edit March 2024: You want parts? Read this post carefully. Do what it says, all of it. Otherwise your PM might not get an answer.

I have a bunch of Sony VFETs 2SJ28 and 2SK82 parts for sale. All parts were tested for gate-source voltage (Vgs) at ±1.6 A drain current (Id) and ±20.0 V drain-source volatge (Vds). There are full curve traces for all parts (Id vs. Vds at Vgs = 0, ±2, ... , ±16 V).

The parts were matched by "curve matching" as explained here.

If you are interested in getting some of these parts:

• Take a look at the parts lists with parameters (Vgs, gm, μ), curve matching, and curve plots:
  • 2SJ28 data is here
  • 2SK82 data is here
• Select your parts:
  • I recommend to first look at the curve-matching list available at the links above. In short, a low delta-Vgs means a good overall match. Read this to learn how the curve matching works.
  • The parameters list may also be interesting.
  • This post may be useful to learn about parts matching based on their curves.
• Send me a personal message (PM) here on diyAudio with your part numbers and complete contact details (see below) and I will reserve the parts for you
• Payment: PayPal preferred
• Once I receive payment, I will send the parts to you

Price per part is CHF 135.00 CHF 170.00
Shipping and handling is CHF 22.00

Contact details (I need all of this, and I may ignore reservations with incomplete information):

• Shipping address (first and last name, street / building / number, town, ZIP, country)
• Your telephone number (including country code)
• Your email address

Hello, I recently made a PCM1794 DAC. The output of the I/V circuit was set to 2Vrms. I'd like to add a headphone output jack and a volume knob here. NE5532 based headphone amplifier Are there any problems driving low-impedance earphones or suitable headphones such as HD600?
Experimenting with this circuit in LTspice, I think the power is too large.
And why do you need a high-pass filter?

Since I decided at the last stage for another preamplifier, I am forced to sell this one. It's definitely a great preamp but for capacity reasons I'm forced to sell it.

The preamplifier has a set bias voltage of 170mA, ideal for low-ohm headphone. All parts are genuine from @Tea-Bag

Sold only together DCG3+DCSTB. (Price is negotiable.)

Additional parts I can offer:

• inteligent on/off module
• Large heatsinks with pre-drilled holes.
• 2 pcs toroidal transformers from (toroidy.pl), 50VA, 240V, 2x18V
• Khozmo attenuator 25K
• AC Power Entry Modules Shurter

I only send within the EU. (paypal, wire transfer)

I can give a discount when you buy all parts.

I had a look at the IRS12092 datasheet it seems to me there's only 1 input.

Also the amp seems to use 2 opamps to convert unbalanced to balanced (or maybe I misread the PCB labels and that it converts balanced to unbalanced). thoughts? Thank you.

https://www.aliexpress.com/i/32884021133.html

I’m visiting a friend and his boys…they are in the early stages of building a speaker…it’s their first one and they don’t have hardly any tools.

I have a twenty year carry on luggage only streak I don’t want to break and would like to bring the following with me….any problems?

Mylar dome tweeter
Speaker wire
Speaker terminals
Aux extension (3ft)
Aux to RCA
Wire stripper and crimper
Soldering iron and solder

The ones I’d think they might have issue with are the soldering iron and the wire stripper/crimper….but please inform me if there are others…or if you’d think no problem with any of the above.

Hello everyone, I have looked up this old post and I want to do this on my TNT first generation turntable: https://www.diyaudio.com/community/threads/diy-vpi-based-turntable.185469/

Anyone know the whereabouts of this turntable? Also, the materials to build this turntable?

Thanks,
Ken

Hi there,

does anyone know at which frequency the inductor values are measured?

In all these calculator for cross overs are values for the inductors. But the value of an inductor differs a lot over the frequencies right?

Hey guys,

Long time lurker here. I play around mostly with Altec speaker stuff, but I do a reasonable amount of DIY audio things. I'm glad to be part of this community and the wealth of knowledge that is here. I'm a graphic designer by trade, so I worry more about the aesthetics then I probably should on my projects... but it comes with the job.

Thanks for having me.

- Woody

Hello, I’m Anthony. I’m an electronics technician turned music equipment repair technician (on the side)

Hi!

As in the title, looking to buy a 1200/1210 mk2/mk3d/mk5 for use at home.

The ideal is a 1210 (black) that hasn't seen commercial / DJ use!
Local pickup in SoCal is best but open to shipping if needed.

Thanks!

Attached is my attempt at updating my Harmon Kardon Citation II amplifier to a PCB version.

It took me 7 versions,and a learning curve to understand the schematic & pcb software.
A friend of mine said that I should sell it as a kit, but I'm too old, 71 and I don't have all of the knowledge required to do so.
I do have a BOM built, and have determined that the cost to do this amp is about $650 for parts , minus tubes. KT88 cost about $400 for 4. The 12BY7A tubes price varies depending on where you buy them.
I'm curious if you or anybody else might be interested in my project.

If you are interested, please feel free to contact me.
Thank you for looking at my project.

Today after 10 minutes of operation the NAD 3045 internal digital went amok resulting in a rapid rise to max volume. I rushed to turn it down and pull the AC cord which I will not connect again. In future, I will avoid integrated computerized amps because they have more failure modes with less reliability than classic amps or chip amps. I am upgrading an old (vintage?) NAD 320 to replace the 3045.

I dislike multi-turn volume controls because they compromise usability.

history: https://www.diyaudio.com/community/threads/nad-d-3045-overheating-allegedly.379438/

Help diagnosing Shuttle 6 Pre-Amp

Help Diagnosing Shuttle 6 Pre-Amp

I resoldered the major caps, all of the solder joints on the vacuum tube pcb, and the input and output ports. That doesn’t seem to have fixed the clipping, but the output did have a hole in the solder joint, which could be an issue, and I cleaned up some crappy solder joints on the vacuum tube pcb with holes too.

It could be an EQ thing where it’s just not going to sound right unless you play with a bass since it’s optimized for lower frequencies - that’s my main thesis as of right now. I do notice playing all higher notes on the bottom three strings the noise fades away really fast or clips, as does the same thing on the entire upper side of the fretboard so I think this one is likely.

If you keep the gain all the way up and volume low you can make playing with a guitar sound bearable, but it will clip on the highs and you will get that background distortion/rumble behind it no matter what, to varying degrees - mostly when hitting multiple strings and on long notes the sustain doesn’t seem right, but again, maybe EQ is to blame.

It’s also way too loud to be pleasant through the headphones unless you turn down the volume via the headphones themselves, I find that odd.

I do notice the input signal seems to be off, like I mentioned the his will cut off; the lows do too at at a lesser rate - but it will come in and out during long sustain notes, cutting off slightly early it seems. Not sure if it’s an EQ thing as mentioned.

I noticed some oddities with the input signal by listening and watching the light. The input light flickers as the sound cuts off - which could indicate a bad port input port, if, again, it’s not an EQ thing.

So perhaps it could be fixed with a new input port?

Howdy-

Apparently I'm not allowed to post anything until I introduce myself, so here goes.

I'm wparks from Lewiston, northern Idaho. Lewiston is a small town at the confluence of the Snake and Clearwater rivers, an actual ocean-going sea port way over here in the Idaho panhandle. We're at 700 feet elevation and deep down in a large valley so the weather is extremely mild. I'm a retired electrical engineer who specialized in microprocessor design and validation for HP and Intel until I burned out, sold everything, and ran to the off-grid lifestyle in the rural countryside. Best move =ever=.

Now that I know what I want to be when I grow up I'm moving into more discrete electronics. I cut my teeth on monochrome monitors before the internet was even a thing, so getting to learn about all of the newest technologies available now to the amateur hobbyist has been very empowering. I have a growing passion for high quality audio, and have begun learning, assembling and experimenting with my hi-fi gear. I am limiting myself to a low budget, building for myself as much as possible, and maximizing fun and experience rather than making some blind jump right at "end game" equipment- crawling before running and all that.

Music that is flowing in my veins: mainlining Portishead, freebasing Tool and more Tool, snorting Black Sabbath, smoking Pink Floyd, and popping Gorillaz

Some gear I own and am learning/having fun with:
Sony SSCS5 speakers - my favorite starter speakers and best sounding I have ever owned (so far . . .)
Philips SHP-9500 Headphones - Inherited from my father and discovered on my own just how sweet they are- Thanks Pop!
1961 Magnavox 9302 15Watt 6BQ5 tube amp - A $40 console pull I reworked into an audiophile contender with Dave Gillespie's mods from Audiokarma
1978 Magnavox R344 12W white trash transistor chassis from a $20 console I turned into a pretty stand-alone wooden cabinet integrated reciever
Denon DCD-1500ii CD player - my only piece of audiophile pedigree bought for $10 from an estate sale that just needed new tray belts (knowledge =IS= power)

Recent projects I have built or are building:
"Combat Audio" +/- 15V 1.5A regulated power supply in an army surplus ammo can for powering all of my diy audio gear
Phono Preamp based upon Rod Elliott's ESP project 06 Hi-Fi Phono Preamp that is the best sounding I have heard (comparing across 4 others)
Most excellent headphone amplifier based upon Rod Elliott's ESP Project 113 that drives my 32 Ohm cans to tantric ecstasy. Oh so good!
Modifying a several crappy Fisher "Studio Standard" 🤣 (B.P.C. era) amplifiers into something somewhat respectable
Repairing and cleaning junked stereos and boom-boxes for dontation to local charities
Crappy Acoustic Monitor PSM-1250 12" 3-way retro design tower speakers I am gutting and redesigning into something respectable.


I have learned an incredible amount about electronics design as supported by my education, and a lot about what makes good audio. I have a voratious appetite for new knowledge, and I love to teach what I have learned. I'm here to listen, learn, and teach when I can. Looks like a fun group.

-Warren

have for sale:

2 (qty) DALE IT800 - Isolation Transformers (Medical Grade <100uA leakage)
Each unit contains an actual 900VA nominal rated Toroidal Transformer
Capacity - 6 Amps input
Capacity - 5.7 Amps Output
Hospital-Grade Power cord with 2 receptacles (Green Dot)
I used one for Analog sources and the other for digital sources to enable total isolation of digital " noise" getting into the analog system

$129.00 USD each with Free Shipping to US & Canada
Take BOTH for $199 USD with Free Shipping to US & Canada

ISOBAR 12 ULTRA 12-Outlet Power Bar - 19" Rack mountable,
15 ft. (4.57 m) Cord,
3840 Joules, Diagnostic LEDs,
1U Rackmount
NOTE: I used Stock pics for the ISOBAR

$99 USD with Free Shipping to US & Canada

Hi everyone, i'm joining to seek knowledge about Daphile and LMS music server. Found this forum to be the best on Daphile and have been reading for a while but now I would like to be able to also post questions. Thanks.

It all started when I decided, just to be a contrarian, to do a hardware simulation of a plugin, ha! I've used 112dB's Redline Monitor for mixing since it came out some fifteen years ago, and I'm sure there are better crossfeed solutions out there, both software and hardware, but it's what I like and am used to, so I thought, why not see if I can make an actual circuit that does the same thing.

Although I've never seen this topology and I arrived at it after lots of trial and error, I'd be surprised if it hasn't been done before, in fact I would assume that the plugin developers based their code on some existing circuit or other, which could be this one or something very similar... In any case, I was delighted that I could pretty much nail the plugin's response with such a simple circuit. So far I've only tested it using LTSpice's awesome WAV processing feature and, although it doesn't null perfectly with the plugin, it's more than close enough and I doubt I'd be able to tell them apart in a blind test.

The plugin has three parameters to adjust: a "center level", which adjusts the phantom center from 0 to -3 dB, a "soundstage", with selects the simulated angle between monitors from 30 to 90 degrees, and a self-explanatory "distance" from 0 to 2m. After some experimentation, I settled on, respectively, -1.5dB, 60 degrees and 0m (any other distance introduces FR ripples that I don't care for), so that's how I've always used it and what I've attempted to implement. In the first picture below you can see the left and right response when only the left channel is active, both as shown by Plugin Doctor and as simulated in LTSpice (I don't show the phase responses but they aso match). See also the schematic and two LTSpice circuits, the AC simulation and the WAV processing simulation.

I'll have more to say about component value selection, gain, opamp choice, etc. if there's interest, but in the meantime you can feed it your own WAV file and see how you like it, just change the file name in the voltage sources and place it in the same folder as the .asc file. Note that, as well as attenuating the output by the inverse of the gain, I knock off another 3dB to prevent clipping, which will happen if your WAV goes to full scale, so adjust accordingly when comparing the before and after. You can also adjust the duration in the .tran statement for the output file length (in seconds) you want. Anyway, have fun and let me know what you think!

Edit: typo in the schematic, it should say "move left to bypass"...

I drawn this schematic with my imagination yesterday. Seems rather promising for testing in real. It can provide 4 watts rms output with THD >5%,

What are the advantages of having a phase plug for a mid/woofer?

Guys,

Yesterday I DJied in a club
That I use to play sometimes

Humble old system below
2 Yorkies LS608's driven by a bridged XLS 2500 and a pair of yammies DSR12's

Although the Yorkies run out of gas before the yammies I balance the system so it sound even.

Anyway ...
I followed the place subs on center of the floor

Yeah normal sound
Coming from the system config.

So it occured to me recalling many reads that says if you face the subs to a corner it will "amplify" the sound ....

I just tested that that I have seen many years ... Never tried , I don't know , is just something that you think will not do much.

OMG!!
It sounded like 4 or even 8 ...cabs
(Never tested multiples next to each other)
So can't tell really but for sure sounded double or triple the volume.

Same cab count ..same single amp
The owner jAW dropped , he thought I bring more stuff and he can't locate the subs as they were not in the center !! Lol

I know that can't be done on some places
But if you can do it..DO it!!!

I will post some pics with some length measurements.
The ceiling is pretty high.

The bass covered the whole dance floor and even another section were another DJ was playing lol.

It was effin incredible....really
How it never occurred to me to actually do it
It was like ....nahh not noticeable or very little gains....nope...totally wrong
It was hugeeee.. improvement !!

And the clients were amazed
They told me ..wow new system sound incredible. Lol!!

Women never have nuff shuz...
Men never have nuff bazz!!

Hello,

I’ve build a clone Blowtorch line amp. See picture of build and circuit below.

As I’m busy with the actual assembly of the PCB’s on a bottom plate from the enclosure, I was testing with different wiring layouts and wires itself (twisted solid core and microphone coax).

During this testing I noticed that one of the 2 balanced line amps generates a Hiss / white noise on the ‘+Out = OUT_P’ output. Taking the signal from ‘-Out = OUT_M‘ gives no noise. Noise sounds like: Analoge ruis - Ambient Reel to Reel Bandrecorder Hiss & Hum - YouTube


I don’t have the knowledge to tell whether a Jfet or a mosfet can generate this as a failure mode due to a static decharge for example. And before I start to remove and change parts, I would like to ask here for experience.

Which part is highest change to resolve this noise issue?



Thanks in advance

Found - thanks.

Looking to convert my Holy Grail to balanced and need a matched set of FQA28N15 / FQA36P15 to complete - I can find FQA28N15 at a few vendors but no FQA36P15.

If someone picked up a board set from the GB and won't be building or happen to have a spare set, PM me as I would be interested.

thank you
..dB