This amazing little amp has been discussed as part of the DB thread but I think it deserves its own thread given how it has quite different objectives from the Pocket Diamond Buffer (PDB) HPA.

This desktop variant uses TO-126 output BJTs with dedicated heatsinks and does not have the separate MOSFET rail switching needed for the pocket one. The setpoints have been optimized for a moderate 40mA nominal bias current that will let you run in full Class A operation for most headphone uses. It will leave Class A around 250mW into 32 ohms. Of course, you can adjust the emitter resistors to run higher bias currents. I have used it with +/-9v, +/-12v, +/-15v and +/-18v supplies and they all work great. It was designed for 15v rails but I don’t think performance suffers much from using lower voltage. For your particular use case you may want more voltage. High impedance 300ohm cans like more voltage, for example. There is lots of room for input cap rolling and I have used huge 4.7uF 400v MKPs and also smaller 0.47uF film to bypass 10uF Elna Silmic electrolytic caps.

Here’s a photo of a built up amp board in operation:

With larger MKP caps only:


I initially struggled with an oscillation but chased it down to a feedback compensation cap that was too large. That’s all fixed now rock solid. A stability analysis on LtSpice pointed me in the right direction. So the schematic shows a 470pF cap needing to be replaced with 47pF. It’s an easy swap and I’ll do this on the SMT pre populated boards that I will be offering in my shop. I will also have bare PCBs for those wanting to assemble it themselves.

Here is schematic for the overall amp, very simple. Power supply in, audio in, audio out and volume pot with MicroMatch IDC cable to an RK09 pot helper PCB (included).


Here is schematic of the Diamond output stage:


Here is a SMT prepopulated PCB so all you need to do is to install all the through hole parts:

Both sides:

Here is the board assembled with big 4.7uF MKP caps:


Standard BOM calls for ECB pinout TO-126 BJTs (Toshiba TTA004 and TTC004) but if you have some classic Toshiba 2SA1837 and 2SC4795 (BCE pinout) you can mount them in the bottom like I did here - leaves a cleaner look on top giving you lots of access to the parts:


Testing on the bench:

Nice low distortion but second harmonic dominant distortion profile for 1Vrms into 33ohms:

You will find that this headphone amp sounds very natural and clean, but has an incredible power reserve to deliver bass slam while maintaining great control authority of the driver transducer cone. It’s a wonderful headphone amp to listen to for hours. For normal headphone use (under 500mW power) basic AC/DC modules designed for delivering circa 350mA into 15v can be used with a 7812/7913 voltage regulator and it will work fine. You can also make really fancy PSU with linear trafos, cap multipliers, CLC, and low noise LDO’s etc. with a proper PSU capable of 750mA and +/-15v you can drive up to 1.5W into 32 ohms (for those fans of the HiFiMan HE-6).

I’ll be offering the SMT preassembled PCBs for $59. Bare PCBs for $22. Boards are all 1oz copper and ENIG finish.

https://xrkaudio.etsy.com/listing/1659118057

BOM is here. Note that BOM calls for OPA1642 but the pre-populated PCB has NE5532 installed. This was due to availability issues but I can assure you that the NE5532 sounds fantastic and measures well too. If you want to swap it out later, you are welcome to do so.

If you want to boost the output BJT bias current (80mA) for operation to a higher power under Class A, use this 2.2ohm metal thin film emitter resistor (R213/213/312/313):
https://www.mouser.com/ProductDetail/Vishay-Dale/TNPW12062R20DEEA?qs=vHuUswq2%2BswIE18TmZVkaQ==

Note that the heatsinks will get significantly hotter and the PSU also needs to provide more quiescent power.

For the 8pin Wurth IDC cable with matching connector for the potentiometer, get this cable (or its equivalent).

If you have trouble understanding this test, or you have questions, please read:
TEST FAQ
Update July 3rd 2012:
Although the title of the thread contains the word "test" this is better thought of a s a survey or poll. The word "test" has caused a good bit of confusion and misunderstanding.
The test tone provided is not used to set you system levels, it is used to measure your system levels. You will set the levels by ear, or by pink noise if you wish.
What you will be doing is measuring your actual listening voltages and reporting them here. The test tone allows you to do that with very good accuracy using only digital playback and a voltmeter.

---

100Also have a look at Archimago's blog for a very clear explanation of this test, along with good photos and link to more info.

---

100
Here is a simple test to determine what voltage your speakers need to play their loudest. From that you can determine how much power you need.
Knowledge is good and this test will tell you just how much voltage (or power) you really need. Efficient speakers in a small room may need only a volt or two. Less efficient speakers in a big room may need a dozen volts or more.
I've included a poll so that we can see what's typical. If you want to also post your speaker details and room details, that would be nice.

For the moment, I'll leave this as a voltage test, we will talk about power later. Those of you who know Ohm's law can figure it out for yourselves. Amp power tends to be an emotional issue, so we'll post the results in volts for now. Since most amps are voltage devices anyway, it's not a bad way to measure. And a voltage measurement removes the speaker impedance from the equation.

All you need for this test is some sort of digital playback (CD, iPod, Squeezebox, computer, etc) and a volt meter with a low AC scale. Any scale 20V or lower should be OK. Meters with only 200V AC scale won't be accurate enough for this test.

Next post has the test tones and instructions.

There are SMD parts, electronic relays, for audio signal. What kind of relays are these ?

Hi

Long story, but I’ll make it short.

Picked up a pair of used Lowther EX4’s in really good condition.

While testing them, in Martin Kings MLTL enclosure, and swapping out amps, they were both turned into smoke machines. Not my best moment…

They are in their way to Lowther for repair, and I will have the option of having either silver or aluminium voice coils installed.

As I didn’t get to finish getting the filter networks installed, parts had not arrived by the time I ruined them, I didn’t get to hear them “properly” with their original aluminium coils

Any thoughts/experiences with the differences between the coils would be greatly appreciated!

Thanks

P

I'm searching for the JBL URIE 813B studio monitor model 840 crossover circuit diagram. I checked overload bulbs and fuses good, so must look deeper. Any info would be appreciated.

Built some single ended planers recently and noticed in distortion measurements the 3rd harmonic was higher than the 2nd harmonic. Is this typical for single ended magnet planers?? None of the free swinging ribbons I've made do this. Has anyone done a compare of distortion measurements between single sided magnet and push pull arraignments? Note the purple line above all the rest in the attached pic

I have been restoring this old amp for awhile now, I was so close to finish it but then I saw smoke coming from R30 in zobel network, it happened twice, at first I thought some short caused this because it was momentary and amp worked fine after reseting the power, but now R30 is constantly heating, what could cause this? That resistor used to be always cold.
Transistors are new, I measured all of them no sign of short, caps are also new. Resistors are old but they are all in range of +/- %5.

My multimeter just stopped working properly—it can't measure voltage correctly, although the other functions still works fine. How can I identify the issue? ChatGPT suggested that if the resistor is still heating up even with no speaker connected, it could mean either C13 is faulty or there's a 100kHz oscillation present.

Any suggestions? Meanwhile I'll look for new cap and replace the old resistor with 10r 5W one, too bad can't measure speaker's output DC voltage right now.

Mosfet F6 Illustrated Build Guide

The Firstwatt F6 is a power amplifier from the brilliant mind of our technical, spiritual, and menu advisor, Nelson Pass.

The F6 is unique in that it uses a transformer for phase inversion, and both phases (The ‘push’ and the ‘pull’) of the amplifier are powered by an N-channel device.

This guide uses this PCB set — F-6 clone boards (2 PCBs, makes 2 channels; Rev 2.0) - Circuit Boards
Or here, the PCB with transformer bundled - F-6 Board + Transformer Kit – diyAudio Store



SCHEMATIC

The Schematic can be found here -



CHANGES TO THE SCHEMATIC - Change Z1, Z2 to a value higher than 5.1V if you need more bias. Increase in ~.5v steps until you can bias properly. The 1N4xxx series diodes are used here, 1N4733 is 5.1v, 1N4734 is 5.6v, 1N4735 is 6.3V, etc...

Suggested change, make R11, R12 110ohm. More information can be found in the BOM.

And an example PSU is here -




diyAudio PSU v3 build guide — diyAudio Power Supply Circuit Board v3 illustrated build guide



CHASSIS

The example amp is build into a diyAudio Deluxe 4U chassis — Deluxe 4U "Jack of all Chassis" (All Aluminum) V2 - Full width with 40mm Heatsinks - Chassis

Also utilized was the diyAudio Premium back panel parts kit — Error 404 - Page Missing

ASSEMBLY



Here are the PCB. The production boards are blue, and have some very slight changes to the silkscreen, but are otherwise identical.

Note that the PCB are a matched pair, not 2 of the same PCB. This is to facilitate keeping the input transformer as close to the back wall (and away from the power transformer) as possible.


Stuff resistors first.


After the resistors, stuff the larger components. Jfets, pots, capacitors, etc…


A stuffed PCB, sans transformer and LEDs (Didn’t have any blue ones on hand. 🙂 )


Input transformer. Note that on pair of pins is slightly narrower than the others. There are corresponding pads on the PCB for the narrow pins.


Transformer mounts here.


Both transformer mounted.


Remember, for best heatsink efficency, the mosfet should mount towards the bottom. This heatsink will make the left wall of the case.






You can see all the wiring of the PCB in this photo. (Red Black Green) is from the PSU, (Black White) is to the speaker posts, and the thin coax connects RCA to input.




Black is PSU GND and Speaker negative
White is Speaker positive
Red is V+
Green is V-



Input wiring from RCA




Completed, working amp. Need only to finish assembling the chassis around it.
(The black box on the transformer lead is a clamp-on ferrite clamp. Necessary? No. Might it help? Yes.)







POWER SUPPLY


IEC socket. You must wire the fuse holder to the switch as shown.


The long wires go to the terminal block with CL-60s, line cap and then the transformer primaries. I needed a bit huskier wire for the safety earth, and added the yellow for clarity.


PLEASE NOTE - this shows 120VAC wiring. Please consult your transformer for proper wiring for other voltages.
Into the block you can see the thermistors and cap, the AC mains from the IEC switch (Red Black) and the transformer primaries (Black Brown White Orange)


Safety earth and the CL-60 used to elevate PSU ground.


Two things to note - 1) I didn’t know if the power transformer was going to magnetically couple with the input transformers, so I wanted it as far away as possible. This will be almost touching the front plate. The L bracket was something I had from a salvage broken amp, so I can’t suggest a part number. Sorry. 2) Although I didn’t use the current PSU PCB, you can see that there is sufficient room (barely) for the board you can buy from the store.


I used an older diyAudio PCB that I had on hand, mainly because I really wanted to use it up.


Bridges mounted.


Wiring up the PCB.


More wiring.

For those of you wondering why I choose to have 2 big caps on one side and 4 on the other, it’s simply because I had those caps in my box. There’s no gain nor detriment to doing it in this manner.




The secret to clean wiring? Simple - start with your leads overly long and trim as you need to, and use lots of zipties. 😀


You can see the connections from transformer secondary to bridges and from bridges to PSU board.


PSU lit up.



TESTING AND POWERUP

Test PSU first for proper voltages BEFORE connecting amplifier circuit boards.

Then, connect only one amp PCB at a time.

Power Up

I suggest starting with a meter across the 0.47ohm source resistor, and watching it as it turns on, you want to have less than .5V across it to begin, turn it down with P2. If you start with the pot in it’s default position, it will most likely have too much bias initially. Turn off the power, turn down P2, and try again.



BIAS


P2 is marked BIAS on both PCB. Adjust this pot as necessary to set bias. Please note that as the pots and source resistors have the 1000uF capacitor in-between, the adjustments will happen in slow-motion and take a while to stabilize. Make small adjustments and wait as necessary. Patience is a virtue.


With DC voltmeter across the 0.47 source resistor, start by setting a bias reading of .5V (500mV) This will give a current of 1.05A, (.5V / 0.47ohm = 1.05A) which with a 24V rail gives about 25W of heat. Then zero your DC offset. Once you are satisfied that everything is stable and happy, you may increase the bias if you choose.

Remember the 3 rules of maximum bias… stop when you reach any of these -

Heatsink of 55C and/or Transistor pin 2 65C
Total bias , both channels, (in watts) of no more than 1/2 the power transformer’s VA
1/2 the maximum dissipation (in watts) of the output device. In the case of the IRFP240, it’s a 150W device, so no more than 75W. (Which is really, really hot…)

Generally, you will reach the 65C limit of the transistor before anything else.



OFFSET

P1is marked OFFSET on both PCB. Adjust as necessary to set 0V DC across the speaker terminals. Again, because the capacitor needs to charge and discharge as you make changes to the pot, any input to the pot will take some time to display on the meter, and it will seem as if it’s reacting in slow motion. This is normal.

Lightbulb Mains lead notes -



Please note that it must be an incandescent light bulb, not an LED or Florescent.
If the bulb ever turns on and stays bright, you probably have a short.

Normal operation when turning on a cold amp will have the bulb glow bright for a second or two, then dim, perhaps off. (this is the capacitors charging, then full)

As you increase the bias of the amp the bulb will glow brighter, and this is linear with the bias amount. A fully biased amp can get the bulb to glow very bright.

You cannot set full bias with the bulb in place - as it increases the bulb will glow more, limiting the voltage to the amp and all the readings will be wrong compared to when the bulb lead is out.

You can, however, set the initial bias with the bulb in place. Start the procedure, turn the pots until something happens, and set, at maximum, 0.1V across the source resistors and zero offset. Getting the pots started this way is a good idea. Remember, this is with a reduced voltage, and the pots will make the circuit draw MUCH more bias when the normal mains lead in used. Expect to measure 0.2V or more with a normal cord.

I've been given a bunch of Caps that the giver was using for experimenting crossovers.
Unfortunately they appear to have been liberally coated with some sort of bitumen type compound and a strange sticky grey blu-tak like goo. (possibly to damp vibration ?)
Alcohol based solvents dissolve the goo but so far only Turpentine will remove the "bitumen"

Is it ok to soak the caps in Turps followed by Meths to clean ? or have you any other suggestions ?

So I upgraded the Nord Rev D buffer board in my Hypex NC 500 based amp with Sparkos opamps and 15v Sparkos voltage regulators. I decided to measure the DC offset at the binding posts with no load and nothing connected to the inputs. My meter started at 20 mV and over the next 10-15 minutes settled out around 350-400 mV. The range would fluctuate by some 30 mV consistency after 15 minutes and never stabilize. Obviously this seems like a very high number and it concerns me. I never checked the amp’s DC offset before the upgrade.

Should the amp be under load for this measurement? I can pick up an appropriate resistor if needed. Also, should I have the amp’s inputs connected? Thanks for any help guys. I don’t want to hook up my speakers until I know the amp is safe. The voltages check out fine and the regulators are holding a steady 15vdc +/- on the buffer board.

Have been gifted a pair of Toa Horns. I've no PA experience and looking at current Toa product I assumed these were 70v line drivers...but was told they were used on std home hifi as additional drivers.
Plugged them in to a basic 20watt/ch 3-in-1 home mini system and they seem to play fine. (other than the constrained range of course)
Can anyone explain what may be happening ?

Thank you for accepting me here in this forum

So the question is a new hivi dmb-a or used dynaudio D52 assuming they are in good condition? Always loved the D52/D28 combo.

Hello everyone, I'm new to the audio world. This is my first practical assignment for my courses so i want to improve it more. Since English is not my mother language and this is my first question so maybe there are some mistakes.
The requirement of the circuit:

• The bandwidth: 15-50Khz
• Output power on load should > 30W
• The efficiency η > 45%
• Distortion < 1% at 1Khz
• Input signal is in range 50mV - 100mV.
• The input resistance: Rin > 20k

This circuit i adapted from Audio power amplifier design handbook of Douglas. However, when i tried to simulate this, i met some problem:

• The quiescence current is too high, about 130mA through Q15 when a load is attached (if no load, floating, the ide current is much lower). I changed the bias voltage with Vbe Multiplier (Q10) but it didn't help.
• The gain to fulfill input 100mV and output 30W which make it become tremendous ~ R8/R9. So i think i need a preamplifier before feed the signal into power amplifier.

Could you suggest me some improvement to the circuit?






thank you verry muchh.

This thread is a place for links to other threads that are related to the freeDSP project. Feel free to post with these links and a brief comment on what the thread discusses. Occasionally the moderators will consolidate them into fewer posts.

Please create individual threads (and link them from here) to connect with other people working with the freeDSP for discussion and to support each other. Please keep in mind that freeDSP is a spare-time project and not a commercial product. If you want to get a freeDSP you need to build it yourself (manufacture board, order parts, …) or organize centralized buying with other DIYers.

The freeDSP is a low-budget open-source digital signal processor family, which is published under a creative commons license. It allows the unrestricted use and modification of the modules. The applications range from active loudspeaker concepts and room equalization over advanced musical effect processors to car audio signal processing. We would be happy if you join us and improve or extend the project.

GitHub is used for file exchange. If you want to join the development team, just send us a private message with your ideas and your GitHub user name. Most freeDSP PCBs will be designed using KiCad. Some guidelines were defined to make future freeDSP development and extensions as compatible as possible. These layout guidelines can be found in the freeDSP-Wiki.

In the following you’ll find a summary of the current freeDSP plans:

green = sources tested and available,
black = work in progress,
gray = on the wish list

freeDSP motherboards:
freeDSP CLASSIC (ADAU1701 / 2 x In & 4 x Out Analog via RCA) freeDSP thread, SigmaStudio AutoEQ
freeDSP CLASSIC SMD (ADAU1701 / 2 x In & 4 x Out Analog via RCA)
freeDSP INSANITY (ADAU1452 / 4 x In & 4 Out Bal. Analog via Jack, alt. 8 In x 8 Out Unbal. / 1 x In & 1 x Out SPDIF via RCA & Toslink)

freeDSP compatible motherboards:
PiDSP (ADAU1450 / RasPi In + Out / 3 x I2S In + Out ) PiDSP thread

freeDSP programmer:
freeUSBi + EZ-USB

freeDSP IO expansions:
freeDSPx AES/SPDIF IN (1 x In AES/EBU via XLR / 1 x In SPDIF via RCA)
freeDSPx SPDIF IO (1 x In & 1 x Out SPDIF via RCA & Toslink)
freeDSPx BAL OUT x16 (16 x Bal. Out Analog via SUB-D)
freeDSPx ADAT IO x3 (3 x In & 3 x Out ADAT via Toslink - maybe even 4 IOs)
freeDSPx BAL IO x4 (4 x In & 4 x Out Balanced Analog via Jack, alt. 8 In x 8 Out Unbal.)
freeDSPx UNBAL IO x2 (2 x In & 2 x Out Analog via RCA)
freeDSPx PHONES AMP
freeDSPx AMP
freeDSPx HDMI IO
freeDSPx DOLBY/DTS/AC3 IO

freeUSBi kits and freeDSP classic kits are almost always available - please use the contact fomular on our website 🙂

Which wood to use? Which fill? What does bracing do? Is CLD best? I'm starting this project to start quantifying some answers to these questions. A lot of work has been done before in fits in starts by others (BBC, Kef, etc.) but they were often narrow investigations and you couldn't compare results because measurement conditions were different among the different studies. Some have cited manufacturer's damping specs, but I have no idea how that translates to the real world. Ditto on accelerometer data. So here I will be doing SPL measurements, that is after all, what we actually hear. The cabinet is ordinary, large enough for the panels to have some radiating area and magnify differences between changes.



I will be using a Peerless 830970 firing into the cabinet using MLS signals. I have to test mic placement, but it will probably end up near-ish field, probably at a side panel. When I do the fill/lining I will mount the SB15 as a dummy woofer and measure what radiates through it. I intend to measure SPL, CSD, and THD. Any other recommendations appreciated.

For the material test I currently have: high and low quality plywood in 1/2" and 3/4", and MDF in 1/2" and 3/4".

For the fill and lining I have: SDS CLD tiles, AcoustX coating, denim insulation, polyfill, 1" thick dacron blanket (what the TL guys use) and Sonic Barrier.

For bracing I will do: simple oak crossbrace versus oak crossbrace using CLD construction with Sikaflex 292i adhesive. I also have some microspherules to combine with the adhesive to see how that changes things.

I would appreciate all suggestions on any other additions to the above. A priority will be that the materials must be widely available and reasonably priced, and the methods must be accessible to an ordinary DIYer.

Hi,

Alex Torres suggested that I should tell my story so I might be able to help others. I had a certain conflict between my laptop and his USB isolator. I will be sharing on digital line level how it was resolved.

I hope this will be useful to others!

Hello, I have 8x DAC PCBs for sale. DAC chip AK4490EQ is soldered on the board. 768kHz/32Bit and I2S/DSD input. All are unused/new bought originally from DIYINHK. Intended for a DSP crossover project but never got to work on it.

I am thinking of 15 EURO per piece plus shipping from Germany. All 8 PCBs together 100 EUROs.

Please let me know if you have any questions.

Best regards,
Klaus

A very simple question! Does anyone know/remember if the mains transformers on these amps can be reconnected to take 230-240vAC mains electricity? If so I shall be delighted to hear - either way - as I have an opportunity to acquire one which appears to be in good condition!

I would also very much welcome a copy of the circuit schematic....please.

[Are you out there Zen MOD????🙂]

Many thanks.

Hi everybody! So I have a Revox B 226-s

I was recommended by a user to change these opamps. It was supposed to be 2 different Opamps NE5532 and LM833.
and the replacement was OPA1642 and OPA1656. But when I opened the machine.. (Wich I should done inte first place I know) there different Opamps. 4 RC5532N
I have tried to contact this user although he doesn’t answer. Is there someone here with Experience in this? Or do I miss understand him? I’m attacking a picture on the Inside of the Revox.

Thanks.

Hi all,

I have a set of LEM Tour Master 12.2 (along with subwoofers), where the one full range speaker acts a bit weird. The system is from 1993, I believe, but they should be of decent quality (Crown amplifiers and RCF woofers).

The speaker plays just fine with no hum until a certain volume. After crossing this volume, the woofer starts distorting a lot. I suspected a blown woofer to begin with, but I am thinking the amplifier is at fault now.
After reaching this volume, when I turn down the volume again, the speaker hums loudly, and it pops loudly as well, when it is turned off. It does this for some minutes even after turned off and on again.
Then it works normally again after a bit of time.

Could this be bad capacitors or something? Has anybody tried something similar or have any ideas? 🙂

Thank you in advance!

I desperately need the schematic for PS Audio PS IV pre amp.

I already called PS Audio tech support and was told that they no longer have any info on discontinued products since the company was sold. The owner's manual is available on line.

The regulator is blown. I need to know the rail voltages.

Any help would be greatly appreciated.

Art

Orders now are open for Quad PCM58 Iout boards, ready built and tested or as a bare-board kit of parts. This board is not a complete DAC, it requires an external I/V stage. PM me while not forgetting to include your chosen payment method and location so I can quote you inclusive of fees and shipping.

When developing the 'Dark LED' I/V stage I prototyped a number of DAC chips to feed it and it was the PCM58 that won out in our listening. According to its datasheet, the PCM58 is the lowest noise multibit DAC chip available - later generations (including PCM63, PCM1702 and PCM1704) haven't been able to improve on its idle channel noise.

The quad PCM58 current out design has discrete regulators fed from IR LEDs for the DAC chip power rails and accepts I2S input. To create a complete DAC you'll need an interface board on the input (S/PDIF, Toslink or USB) and for generating the output, an I/V converter. The intended use case is with the 'Dark LED' module - the power rails are the same for the two PCBs, +/-18V. Using 0R resistor options the PCM58s may be configured in single-ended mode or alternatively, balanced with an on-board phase splitter. Multiturn pots are included so you can trim the 4 most-significant bit weights of the DACs assuming you have suitable measurement kit. If you don't have that kit, I'll leave those trimmers off the board by default so as not to disturb the factory trimming of the DACs.

Physical dimensions: 100mm * 100mm, max height 20mm. Fixing centres : 95mm * 95mm, M3 holes.




Price for a quad PCM58 built and tested unit is : 780RMB (~USD108) to include 4 recycled PCM58 DAC chips which are fitted in turned-pin IC sockets.
for a bare-board kit, including 4 DAC chips the price is 640rmb. We'll be putting up the stuffing guide shortly.

If you already have your own PCM58s the price for the board without DAC chips is 500RMB and I'll supply the sockets unsoldered unless requested otherwise. This is because many recycled PCM58s have disfigured legs which are rather tricky to fit into sockets when they're soldered down due to the nearby components.

Our preferred payment method is via Wise which typically adds a 2% fee. Our receiving currency is CNY, alternatively USD or Euro. PayPal may also be used, in USD but will attract higher fees, 5.5%.

Shipping is in addition and depends on your location and speed of service. Courier (FedEx, TNT, DHL) typically takes 8 - 10 days and e-packet four to eight weeks. Not all locations can be serviced by e-packet though.



What else is needed to turn the Quad PCM58 into a fully operational DAC?

You'll need an input interface board to feed an I2S input to the Quad PCM58 over 3 wires (no MCLK is required). The popular gold-standard USB interface chip CM6631A though is typically not compatible with this board as in stock form it outputs a BCLK of 128fs at 44k1 sample rate whereas we require a 64fs BCLK. If you buy the CM6631A card through us we can perform the firmware upgrade needed to make it compatible. Your output stage (I/V stage) could be as simple as a couple of opamps, configured in transimpedance mode. A low-noise, high speed one is preferred. The ideal I/V from the SQ perspective would be a discrete one like 'Dark LED' customized slightly to run from a unipolar input current. Seeing as the PCM58's DAC internals are unipolar, it generates a bipolar output current through the provision of a secondary fixed current source which adds half the full-scale peak-to-peak output current (1mA). This is the function of the BPO pin on the DAC (pin5). Seeing as this additional current source adds noise it is best to avoid using it if the I/V stage can handle a unipolar input current, which the Dark LED can, given a change to two resistors (R1 & R2). When the BPO pin isn't being used the 0R resistors R7,R30,31 & 32 are not fitted, nor are C7,C14,C20 & C27 which decouple the BPO current source.

You'll also need a well regulated low noise power supply of dual rail +/-18V rated at at least 150mA An LM317 and LM337-based board will be sufficient as the PCM58s have their own regulators. If you already have a single winding AC supply (an AC wall-wart) then 18VAC at 10-20VA would be in the right ballpark. Use a half-wave rectifier so you get both rails from a single winding or alternatively, a centre-tapped trafo. I don't recommend using switching supplies due to issues with common-mode noise, its very hard to filter out unless you're using balanced connections to your amp or pre.

Speaking of balanced mode, the board may be configured into being a 4 channel (L+,L-,R+ & R-) system at 1mA peak per channel or a 2 channel, 2mA peak per channel solution with the four DACs configured in two paralleled pairs. In balanced mode you'll need 4 channels of I/V. The configuration is handled by 0R resistor on the jumper 0R resistor locations U16-U18 and R33,R34.

Hello everyone! I'm new to this, so apologies if my questions are a bit basic.

I'm working on a small project to "re-factor" an old Samsung HW-D570 soundbar into a compact 2.1 audio system. My plan is to move the amplifier section into a standalone box, connect two wired speakers to it, and keep the wireless subwoofer as it is.

According to the specs, the original soundbar uses two speakers described as "2-way, 80 Watt, 4 Ohm, wired". Each speaker actually contains two small woofers and one tweeter. The woofers are labeled "HUSNF-00900, 8 ohms." When I measure the impedance from the wires going to each speaker (which includes both woofers and the tweeter) with a multimeter, I get about 4.5 ohms.



I have a couple of questions:

• Is this a 2-way or 2.5-way speaker system? And are the two woofers in each speaker likely wired in series or parallel? I can't see the internal wiring clearly.
• Main question: Can I replace the current speakers with some spare Monitor Audio Radius 45 speakers I have? These are 8-ohm, compact satellite speakers rated for 15–50W, with a 3" woofer and a rear-firing tweeter. (Or should I rather use 4-ohm speakers? Or only keep the original speakers?)

My main goal isn't necessarily to improve the sound quality, but to change the format and case of the system to better suit my needs.

Thanks in advance for any advice or insights!

I'm thinking about upgrading my surround system to Atmos so have a need for in-ceiling speakers. In my search I came across these:

https://www.crutchfield.com/S-bYgFC...c2IpAbYCHpZzlSha3oc8dEza696tvhZYaAhdoEALw_wcB

This got me to thinking... these use a single full range driver shoved in a can which seems doable with a 3d printer and a few hours in Fusion 360. Can anyone think why this wouldn't be a good idea? If this isn't a completely dumb idea what drivers should make my short list? I think my starting qualifier would be that they have to get loud enough to be used at reference volume without notable distortion.

Hi
the base of this amp is the huge thread about :

100W Ultimate Fidelity Amplifier​

these is the thread for all fans of the Vertical brother of APEX FX8 done by XRK and prasi:
the amp was modified by XRK and others and use no Lateral MOSFET --> therefore well known IRFP240 and IRF9240 are used.

here are some examples and comments:
here
here:

I have for sale 2 sets of components for building the Miro AD1862 DAC. Each set consists of 1 x AD1862 SMD DAC board, 1 x Miro PSU1 board, and 1 pair of AD1862 IC's. AD1862 IC are genuine (from Paddy & zoom777) Each set is priced at 40 usd plus shipping. Please see the attached pic.

Regards,
MM

It has been a Long time after the DDDAC1543 saw the light and the world moved on, especially if we look at available sound track material. So much great soundtracks with high Resolution. Red book is not dead of course, but if I spend money on sound tracks, I look for 96/24 or even better 192/24, or the SACD equivalents 88.2 and 176.4...

In the past year I have been working on my new project: designing a DAC which will play up to 192kHz / 24bit material from a PC (or Mac...) There are many DACs who can do this, but I wanted to design a NOS version, as so far, every listening test I did, this led to a more open, natural and detailed sound.

I found this to be possible by using the well-known PCM1794 and avoiding the digital filtering built into the chip. This proved to be quite tricky, but I managed to find the key to do it. It has been a great adventure with much to learn and a fantastic result at the end....

To feed the DAC, I kind of integrated the WaveIO USB Receiver from Lucian. Great design and better sounding than all others I tried (see my website for this as well); I give a very detailed report on www.dddac.com

You can download the circuits here: dddac1794_nos_ver30.pdf



The Images below show:
The Mainboard with I/O and I2S to right justified conversion
The actual dual mono DAC Module (can be stacked parallel)
Built up with 4 modules and two power supplies (5 and 12 Volt)
Wired up in Chassis (Mundorf Gold silver oil output and optional Sowter 1:2
The DAC in my Audio rack
Another view inside
Block diagram


I opened this thread to share my design and experience and give a platform for questions, feedback, remarks and/or discussion...

Hi all,

Talk me out of this? I'm just enamored with the idea of a small foot print "tower" that is taller with lots of 8" drivers in it. Instead of a big box, a tall skinny tower. The small foot print is just easier to place in various rooms. Visually it's fun to see. It allows the use of lots of inexpensive drivers. And generally doesn't need much power to get loud. But would it need to be a sealed enclosure? Or ported? All the subs sharing one volume? Or make groups of 2 in their own cubbies in the enclosure? 4 to 8 drivers so that wiring loads is easier?

I tried modeling some options in WinISD but I'm not sure if it handles real world what happens with multiple drivers in one volume or not? Any other options that are more realistic?

Overall bad idea? Better to just build two 10's and stack them instead?

Purpose would be music mostly.

Very best,

Some pics of a prototype general purpose DAC Clock board, for which I plan to post the Kicad project. First pic is driving a Andrea Mori FIFO Buffer board; second pic is driving a @MarcelvdG RTZ FIRDAC with asynchronous USB, and reclocking before the dac board (clock board prototype is outlined in red).





The original premise was that the clock board would use 45/49MHz clock modules (or external clocks, including sine wave oscillators using suitable squaring adapters). The 45/49MHz frequencies can be used to externally clock an I2SoverUSB board. The clock frequencies are then divided by two to produce 22/24MHz clocks for driving dacs. Both continuous and switched 22/24MHz outputs are available (USB board usually controls the switching). All outputs are buffered with high performance buffers. Very clean +3.3 power distribution and bypassing is also used. That's the short story.

The board design evolved to also support 22/24MHz clocks (or even 11/12MHz), then use a frequency multiplier to drive the USB board (as an option).


So far with the prototype and some pretty good clocks, it does raise the DACs in the pics into what I would consider to be a higher sound quality performance league as compared to what they would sound like with off the shelf NDK SDA, Accusilicon, and or Crystek (although the clock board can make those clocks sound better too), and or especially better if compared to having the master clocks internal to the USB board.

Of course, all the above having been said, there are commercial solutions out there too. Some may be essentially SOA, but they may be more costly and possibly less adaptable than the clock board of this project.

----------------------------------------------------------------------------------------
Clock board v2.1 files attached below.

Interconnect Board open source project for connecting I2SoverUSB and PCM2DSD to the clock board and reclocker board is at (#68):
https://www.diyaudio.com/community/threads/general-purpose-dac-clock-board.413001/post-7702293

Reclocker Board open source project files available at (#157):
https://www.diyaudio.com/community/threads/general-purpose-dac-clock-board.413001/post-7729422

Some comments on using the clock board with very high performance dacs in #144:
https://www.diyaudio.com/community/threads/general-purpose-dac-clock-board.413001/post-7729060
----------------------------------------------------------------------------------------


NOTE: Clock board schematic was found to have wrong part number (now removed) for the Omron relays. Please see the clock board notes document for more information on part numbers.

8/11/24 - Clock Board Notes document updated to v1.1 (some changes to recommendations for regulator load resistors)

I originally made a similar post in the Room Acoustics forum; however, there appears to be a lot more discussion of CLD in this forum. (Pretend the room is a big speaker enclosure – just kidding.)

I am remodeling my family room and want to make it a better “listening” environment for stereo and future multichannel. About ten years ago I read Earl Geddes excellent book “Premium Home Theater” and one of my big takeaways was his recommendation to use CLD in a listening room to absorb bass while preserving midrange and treble liveliness. Although Geddes also recommends CLD for sound isolation - that is not my objective, since one corner of the room has an open ½ flight of stairs up to the middle level of our home with no good way to enclose it.

Tentative plan is to incorporate CLD into the ceiling, the front wall and one sidewall of the 8 x 14 x 24ft room. Sound isolation clips would be attached to studs/ joists with 7/8” steel hat channel attached a maximum 4’oc and spaced a maximum 2’ apart. Two layers of lightweight ½” drywall separated by a suitable damping compound (Green Glue?) would be attached to the hat channel.

Is this a worthwhile endeavor to hopefully preclude the need for standalone bass absorbers, multiple subwoofers, room correction software, trial & error measurements, etc?
Is this a good design? Am I on the right track with the following design assumptions?

Increase the flexing of the constraining layers (and thus energy absorption via CLD) by decreasing the stiffness and mass of the constraining layers.

Reduce the strength of reflections into the “listening” room by decreasing the stiffness and mass of the constraining layers.

The damping compound that is used between the constraining layers should have very good adhesion and be more viscous than elastic so that energy is absorbed rather than returned to layers.

Minimize the Resonance Frequency of constraining layers by increasing mass and decreasing stiffness.

I'm really looking forward to this -

Designing Audio Power Amplifiers: Amazon.co.uk: Bob Cordell: Books

Hello, I'm a new member from the USA. I'm currently building a 2 way speaker. I have interest in building a power amp.

Hey Guys, I have found this forum invaluable in helping me evaluate which sub I want to build to start my mini rig. In particular the Xoc1 TH18 thread has me sold thats the way i want to go. Thanks for having me!

Hi guys, new to the "business" and trying to build a wifi speaker 😀 will be fun

I've been trying to find a reliable formula to calculate the total group sensitivity of a relatively odd configuration and number of LF drivers. They would be arranged as close as possible to each other.

In my own personal scenario ->

- 3 identical 12" LF drivers, wired in parallel, individual base sensitivity of 87 dB/2.8V/1M per driver, arranged in a triangle.

- what would the total sensitivity be of all 3 drivers playing together at frequencies lower than the equivalent of 1/2 WL?

With an even number of 4 drivers connected 2 in series, then the 2 groups paralleled, we'd theoretically observe a 3dB gain from doubling the driver count and another 3dB from doubling the cone area, but lose 3dB from cutting the drive voltage to each woofer of the series connections. That leaves us with a total net gain of only 3 dB if we only count the frequency range that falls in the 1/2 WL CTC driver spacing.

With an abnormal quantity of drivers, CTC spacing and adding in series.resistance from a thinner awg, it becomes a bit more complicated to figure all this out. Short of putting it in a software sim, which isn't always accurate, there has to be a better way to get a closer answer.

Do i need to worry about baffel step correction if i measure all my drivers in the finished speaker cabinet or will that measurement already account for it? Thanks! Travis

The TDA7293 series are underrated gems in the DIY world. This chipamp, with its modular structure, clip detector, mute-stby functions, low noise and low distortion, produced with DMOS technology, actually has the potential to compete perfectly with National/TI's LM3886. On the other hand, it is common practice to use chipamps with a buffer due to low source impedance demands and relatively high input bias currents. Likewise, with composite feedback applications, both the noise and distortion efficiency of chipamps can increase significantly. In our implementation, two matched N-channel JFETs, together with the PNP transistor, operate as a complementary feedback pair as a high-impedance, high-CMRR differential input buffer for the TDA7293. The JFE2140 / LSK389 LSK489 dual JFET combination used provides significant improvement in THD and noise compared to the use of the chipamp alone, thanks to its extremely low noise. The composite feedback rate is 26dB (20x). In this way, ~1.25v RMS input signal is sufficient for full power. On the other hand, thanks to the DC servo circuit established with TI's low-noise, low-offset opamp OPA277, the DC offset remains below 1mv.

Our inspiration for our design was Bob Cordell's LSK489 application notes and JFE2145 datasheet.
https://www.cordellaudio.com/JFETs/LSK489appnote.pdf
https://www.ti.com/lit/ds/symlink/jfe2140.pdf?ts=1717672324923&ref_url=https%3A%2F%2Fwww.ti.com%2Fproduct%2FJFE2140
https://www.st.com/resource/en/datasheet/tda7293.pdf
https://www.ti.com/lit/ds/symlink/tps7a49.pdf?ts=1717660447218&ref_url=https%3A%2F%2Fwww.google.com%2F
https://www.ti.com/lit/ds/symlink/tps7a3001-ep.pdf?ts=1717738722969&ref_url=https%3A%2F%2Fwww.google.com%2F


The ancient 78xx 79xx series regulators, which are well known in the electronics community, have been used in audio circuits for years thanks to their stable structure, high PSRR rates and relatively low noise(?).
Measurements of 78xx-79xx ICs (even though noise values such as "0.003% Vo" are given in their datasheets) have been found to reach noise values of 500uV - 1mV in the audio band, which may disturb sensitive ears, regardless of the input noise. Of course, such a supply noise is meaningless for modern audio opamps whose PSRR is around 120dB.
However, especially in discrete applications (since PSRR efficiencies are quite low), this level of noise will be disturbing.
Although complex, discrete, parallel-hybrid low-noise solutions have been produced to solve this problem, these applications remain a troublesome alternative due to application difficulties, additional noise due to the area covered by the regulator, and the necessity of matching components in some applications.
Our application provides a good and affordable alternative to all these difficulties. TI's low-noise regulators TPS7A4901 for + rail and TPS7A3001 for – rail form the heart of our application. The effective noise amounts of these regulators in the audio band are below 15uV (RMS) level. In addition, while PSRR rates are 60dB in the 78/79 series for 100-120Hz, they are around 70dB in the TPS7As. On the other hand, load and line regulation rates and transition responses are significantly higher than the 78/79 series.
Although the input voltage of these regulators is limited to a maximum of +/-36v, input up to +/-60v can be applied thanks to the input voltage limiter used in our application.
Since the pinout of the 78/79 series is used directly, they can be used as a one-to-one replacement in applications where these regulators are used.

EPIC S

EPIC S


Positive Regulator Schematic


Negative Regulator Schematic


Final prototype measurement of EPIC S.
1K-1NF input filter
47K-4.7K NFB Resistors
4.7mA Input bias
Power: 78W 8ohm (2x42V)
Input sensivity: 1.25V
Slew Rate: 8V/uS 8ohm/1nf (JFE2140)
Slew Rate: 9V/uS 8ohm/1nf (LSK489)
Bandwidth: 5-75khz -3db
DC Offset: <500uV



EPIC S Output Noise

Hello,
I'm building a P3A amp that will be paired with the ESP tone control preamp. The pre already has an output cap so I'm wondering if there is any reason to keep the input cap on the P3A, and if not, could I also eliminate R1, R2, & C2?

Earlier in the Le Monstre Preamp thread, we posted schematics and spice files for 3 different line level buffers.
www.diyaudio.com/community/threads/preamp-for-hiraga-le-monstr-2024.421562/

They all have JFET inputs and use 4 complementary transistors in total.
We have now built and measured them all, as shown in the pdf attached.


Patrick

.

Hi

The big thread of "100W Ultimate Fidelity Amplifier" is a chaos.
i need time to search through all the page and mark all i can find.
pcb is done in yellow by jlcpcb.

Gerber and all other documents are in post 13908: here
input BJT i use the recommended KSC1845, driver are 1381 and 3503, power mosfets are the renesas 1058/J162
as C4 it is written 220µF, and XRK use a polymer elco with 330µF. I use a orange 1000/16VµF by epcos
for C9,C16, C12, C15 i use Panasonic 680µF/50V
input cap is a red polymer panasonic 2,2µF/400V instead of 10µF
i use the lower gain setting R11, R12--> 550R and 20k
i did my first output coil...its not nice done but it has about 0,9µH the resistor is under the pcb

bias setting:
post 6686 bias at 100ma-170mA, wrote about goldmund amps--> bias 200-300mA
my bias is set according to the hint of post 7753 by bimo to 400mA. the amp is working fine.
post 6971 test by XRK as an Class A amp with 1,3A works fine...big heat sink is needed!

i used for the first test a 0,65K heat sink.
i plan to use a 4U/300 case by modushop with a 300VA 2x24V sec. transformer

i will just check the other channel.
psu voltage is 31 per rail and bias is after 2 hours 420mA stable. no heat problems, no smoke..

Gain is about 31dB, at 8ohm load about 29WATT.

if you have any hints , let me know.

edit:
Apex directory - done by XRK ..thx:
here:

edit: 26.8.2024: set bias current back to 400mA #post 104
edit: 25.03.2025: post 123 latest PCB Gerber by Peter(Kleinhorn) - thank you!
post 123

kr
chris

I have a distortion in one of my tweeters on my 3-way. The distortion occurs only at certain frequencies and above a certain volume. The distortion is pretty deterministic, I can reproduce it. They are old 3 ways energy speakers from the 2000. I managed to isolate the distortion problem : it is only one side, and because I can bi-wire it, I can confirm that the distortion is coming from the tweeter and not from the mid or bass (because mid and bass share the connector).

I also swapped sides of the amplifier, so I know it is the tweeter on the speaker and not the amplifier that is problematic.

The screw to remove the drivers are not accessible (see: https://rvb-img.reverb.com/image/up...,t_large/v1639605839/olidljagmyhg8zmbkcx4.jpg), so I would like do the debugging and part sourcing before taking them apart.

Is there something I can do to pinpoint where the problems, i.e. crossover, dome, magnet, voice coil? And the potential solution?

Thanks

Few years ago I build Seas based speakers. (T25-001 and W17E-002). Speakers was my own design, even filter.

I was quite satisfied speakers sound to the moment when I get chance to listen original Jamo Concert 8 speakers. After that I was not satisfied anymore


So my question is that, does anybody have Jamo Concert 8 (Jamo D830) crossover schematic?

I have seen some pictures taken from that crossover board, but component values are not clear to me.

Picture shows how components are placed to board.
Red are capacitors, white are resistor and black circles are coils.

I got the proto of the s.e. trafo with multi tap
5 kohm, 43% and 20% UL
5 ohm single secondary
around 20H
primary wire 0,28 mm diam - Rdc 140 ohm
max Ia around 200mA
Pmax around 10 watt, column 40x45,
M3 - 0,3 mm. o.g.
4,4 kg




This is not available.

To play in the real world not in virtual
Testing some circuit.
Unfortunately I have to service now my Sofia ( some hard problem) then I can test the tube that will play in the circuit
proposed


Walter

Good morning

I am building a clone of the SUGDEN A21 amp (class A) and want to associate a very good preamp with RIAA, tone control.
That of ELEKTOR 2012 caught my attention. To validate the quality of the PCBs manufactured by JLCPCB, I ordered the tone corrector part from them. Superb quality.

Now that I am sure I can have PCBs, I would like to have feedback from those built or use/used the Elekto Preamp 2012 or want to go in the same adventure than me!
So any feedback from this preampli would be appreciated, going to build it alone with no support is a bit hard.

Please let me/us know

Georges

Morning all. I joined in hopes of finding drivers to refresh a Fischer & Fischer SL 1000 speaker built in 2000. I purchased the speakers used. I'm looking for Seas W17EX002 bass woofer preferably in black. Unfortunately, this particular Seas woofer is long out of production. The original speakers are in white, but I'm trying to redo the speakers in all black. If you happen to know the crossover points for these speakers, please let me know. Thanks.

Hi Everyone,
I`m an EE student, loved music since i was born. I`ve been on this forum for a while to gather info and ideas since there`s plenty. Have a great time.
Marco

Another, more powerful Woofer. Also - could I use a compression driver? I have these small speakers. For homework.

Hi all, I signed up here years ago, so I suppose that at some point it's time to say hello.

Hello.

This might be a silly question, but I've been wondering if there are any tips for listening to parts of a multi-way speaker individually?

I've been working on some multi-entry horn ideas, but want to test out various parts individually to get a feel for how they sound before commiting to a complete build around certain drivers.

It got me thinking of using an existing speaker and just EQing out the range of interest and then using it to "accompany" the driver in question.

Please let me know if anyone has ideas about this.

Hi
in attach the first proto of 300B OT with nano
It is 3k primary
5 ohm secondary (single)
Around 40H -100Hz but some adj with gap will be possible.
At the end it will be potted.
The first test are almost fine.






The Freq. answer; there is a little step at 50kHz, but it will be fixed
Zs are 700 ohm (blu) and 1 kohm (red)
At thr moment out of the power stage that will be ready soon



The phase

cancelled due an error

I saw a simple amplifier but thought maybe I can do it better without to many components.
Here is the result. Its not very temperature stable but in class A it gets less hot the louder you play.
So it is just to measure the current in the output transistors = voltage over R8 or R9 and adjust it with P1 to
a little more than Feed voltage / 4 * R speaker. But take half an Hour for the heat to be stable after the last adjustment.
It is easily adapted to other voltages and speaker impedances. Distortion is about 0,1% which maybe some people don't hear.
Noise is extremely low under 5 uv 20 - 20k. Common mode rejection is determined by C2. Frequency response is good 25 Hz to 95kHz -3dB.
View attachment 1418335


But looking at the results it is possible to do it much better. Faster and much lower distortion but a little more noise.
The problem now is that the P1 is very sensitive in adjusting the current.
Now the anti thump + radio interference filter on the input is included. But still pretty simple.
I changed the R12 to 0,82 ohm because I believe the IRF 540 has about 5S at about 1A.

I want to buy 2 prasis crc pcb with lt4320.

Glad to finally be a part of this community, appreciate the wealth of knowledge on this site!

Hey I'm having trouble understanding exactly how I can control volume from my TV remote. I really wish TV's still had a volume controlled RCA analog output but here we are. I only have an optical out and HDMI ARC port. Is it possible to use one of those HDMI arc to analog audio converters so I can use the RCA output to my amplifier and control the volume with the TV's remote?

Which one of these caps is for the delay of the protection circuit? and what the other cap is for?

I'm looking for a schematic and any info for an 8 Channel Delco amplifier from an older Peterbuilt heavy truck. I think it may be the same as the older GM Monsoon systems used. It's an 8 channel amplifier with built in crossovers. It uses eight Delco 82452 chip amps. This one won't power up and the head unit is displaying "Check Amp Fuse".

I'm confused because I see microphones placed one meter away, and others only 20cm from the speaker.
And while we're at it, any advice on what to take to measure with quality.
the latest generation sound card, which should also be purchased, but I already have clearer ideas, will have the best features.
I just installed REW

I think Ronnie O'Sullivan has done the 147 snooker thing about 15 times. But this one was a particular goodie:

Login to view embedded media
Obviously raising his game with Reanne Evans. Like you would. But he did give her a fair chance! 🙂

The comments are achingly funny too. People who know how hard a 147 is. 😎

VRDN: bipolar regulator PCB for line level ckts: ±11V to ±20V @ 1.5A with "De-Noiser"

A couple of diyAudio members and I perceived the need for a power supply board aimed at line level applications, such as for example Wayne's BA 2018 linestage. We felt it should include the following features:

• Rectifier diodes on board
• Transformer snubbers on board
• Usable with both dual-secondary and single-secondary transformers (AC-to-AC wall warts)
• True 1.5 ampere output capability, therefore: real heatsinks & big capacitors
• Output voltages field adjustable via multi-turn trimpots

The result is a PCB called "VRDN" - Voltage Regulator with De-Noiser technology. The DN technology was originally invented and published by Charles Wenzel, whose original article called it "Finesse" (link) . DN technology was greatly improved and expanded by diyAudio member Elvee, who named the improved circuit his "De-Noiser" (link) This regulator PCB implements De-Noiser circuitry, as shown in the schematic of Figure 1 below.

Capacitor C13, and everything to the right of C13, are the De-Noiser. Jumper block J3 allows you to easily disable the De-Noiser (jumper installed) or enable it (jumper removed). This gives the pleasant option of performing a before-and-after pair of tests, either with measuring equipment OR with human listeners. Try it one way, then try it the other. Which do YOU prefer?

The board includes two transformer secondary snubbers, namely (MOV1, C1, R1) and (MOV2, C2, R2). Metal Oxide Varistors perform double duty in these snubbers, serving as the across-the-secondary capacitance "Cx" and also acting as voltage clamps, if an AC mains spike or surge occurs. The values of snubbing resistors R1 and R2 were chosen after performing several Quasimodo tests on several different transformers, and I expect these resistor values will work well IN THIS PARTICULAR CIRCUIT, with a wide variety of different transformers. However, if you wish to perform your own Quasimodo tests on your own transformer, and derive your own optimum values for R1 and R2, please go right ahead with my blessings and good wishes. Just remember to insert your actual MOV1 component into your Quasimodo's "Cx" socket.

Figure 1 includes two bridge rectifier components, BR1 and BR2. Each of these implements a "Full Wave" rectifier, meaning it transfers energy from the AC mains to the DC filter capacitors, two times per cycle: once at the peak of the mains sine wave, and once at the trough. The circuit contains two Full Wave rectifiers, one per transformer secondary. One of them produces the positive DC voltage output, and the other produces the negative DC voltage output.

However, when conceiving this board, we envisioned that some hobbyists might wish to build a bipolar regulated supply but feed it from an AC-to-AC wall wart instead. This has the great advantage that it keeps the mains voltage completely away from the DIY chassis; instead of 115VAC or 230VAC, the wires which enter the chassis carry only 16VAC or 18VAC, which is noticeably less dangerous. An AC-to-AC wall wart also allows a smaller DIY chassis with less interior volume, since the transformer is now outside rather than inside the chassis. But an AC wall wart is just a transformer with a single secondary.

VRDN implements the conventional, textbook method for creating a bipolar DC supply from a single secondary transformer. It's called a "Half Wave" rectifier, and as the name suggests, it transfers energy from the AC mains to the DC filter capacitors, only one time per cycle: either at the peak, or the trough, of the mains sine wave. One HWR generates the positive DC voltage output, and the other HWR generates the negative DC voltage output. Exactly as is done in the Objective-2 headphone amplifier, among MANY other examples.

To use the VRDN board with a single secondary, AC-to-AC wall wart, we need to make a few very minor adjustments. They are tabulated in Figure 2 below. As you can see, the only changes are to the list of stuff-and-solder actions. No PCB tracks need to be cut, no soldermask needs to be scraped off. You simply do-not-stuff six components, add three jumper wires (each 2.5mm long) into pre-labeled locations, and stuff two discrete diodes into existing & labeled thru-holes. It's easy.

Figure 3 shows two VRDN boards. The top PCB in the photo is configured for a dual secondary transformer -- notice it has two green Euroblok connectors at the left edge of the board. One connector for each secondary. The bottom board is configured for a single secondary AC-to-AC wall wart. It has only one Euroblok connector and only one snubber at the left edge of the board. Everything else on the two boards is identical.

Figure 4 is a detailed picture of the input circuitry, on a PCB configured for two secondaries. Again notice the two Euroblok connectors.

Figure 5 is a detailed picture of the same region, this time on a PCB configured for a single secondary AC-to-AC wall wart. You can see that the bridge rectifiers are not stuffed, the (MOV2, C2, R2) snubber is not stuffed, and the Euroblok connector is not stuffed. You can also see that the three jumper wires "JOR1, JOR2, JOR3" have been stuffed -- look for pink arrows on the photo. Discrete Schottky diodes (5 amperes, 100 volts rated) have been stuffed -- look for gold stars. One diode is stuffed in holes "A" and "C", the other diode is stuffed in holes "L" and "M". Exactly as required by the options table shown in Figure 2.

The Jumper Options are shown on the schematic as 22 megohm resistors. This is a limitation of the CAD system and, specifically, a limitation of the CAD system USER, namely me. In truth, in real life, on your real PCB, JOR1 is either a zero ohm resistor (i.e. a piece of 22AWG hookup wire with white insulation) or else it is an INFINITY ohm resistor (i.e. an open circuit; nothing stuffed). Same goes for JOR2 and JOR3. Completely disregard "22 megohms" on the schematic. Ignore it. Pay it no attention. Simply remember that JOR1-3 are either opens or shorts. Opens or shorts. Repeat after me: opens or shorts. Look at the photos in Figures 4 and 5. Those options are either not-stuffed-at-all (open) or else stuffed with hookup wire (shorts). You can see them right in the photographs. Repeat after me: opens or shorts.

Figure 6 is a photograph of the output side of the PCB. Individual wire-to-board connections are provided for supply and ground, for each of two stereo channels. Also visible in this photograph are the vertically mounted resistors. This PCB uses the vertical orientation exclusively. Finally if you look very closely, you can see the amber colored Kapton tape on the back of the heatsinks.

TIPS FOR BUILDERS

Please carefully study the top silkscreen text layer when stuffing and soldering the 1N4004 diodes and the 1N4148 diodes. There is a little "A" next to each hole where an Anode is soldered. You'll find that on every 1N4004 and 1N4148, the cathode (band end) of the diode is the short wire which is stuffed straight thru the PCB with no bends; the cathode is the "butt end" of the vertical diode. The anode wire is the one which is bent into a 180 degree U-Turn, and is exposed for easy probing. The anode wire goes into the PCB hole with the little letter "A" on silkscreen.

First stuff and solder all resistors and axial diodes. Second, stuff and solder the MOVs and the bridge rectifiers (if using!!). Third, stuff and solder the Euroblok connectors and the two pin headers J3 and J4. Fourth, stuff and solder the small (non 2200uF) capacitors, the trimmer pots, and the transistors. Fifth, stuff the regulators-on-heatsinks (next paragraph).

After you bolt the LM317 and LM337 to their heatsinks, including the Sil-Pad insulator and the insulated shoulder washer / bushing, I recommend that you test-fit one of them into the PCB. You will discover that your unmodified assemby doesn't quite fit! You'll need to modify it. Bend the leads of the TO-220, backwards (towards the heatsink) about 1-2 millimeters. Try this. Aha! Now it fits. Put a piece of Kapton insulating tape on the back of each heatsink, over the TO-220 mounting bolt. The two heatsinks are positioned back-to-back and we don't want them to come into electrical contact. Kapton tape prevents that.

Stuff the regulators-on-heatsinks so the heatsink edges are flush with the PCB, but only solder the IC legs for now. Do not solder the heatsink mounting pins yet. Instead, turn the board right side up and look at the heatsinks placed back to back. You want to see a nice wide air gap between the two heatsinks, about 3-6 millimeters wide. When building my boards, I folded a sheet of paper in half, and half again, and again, etc., until I had a bundle 5mm thick. Then I jammed this paper bundle down between the two heatsinks to create a definite gap between them. With the paper in place, enforcing a nice gap, THEN I soldered the four heatsink mounting pins in place. After the solder joints cooled, I removed the paper bundle and voila! A nice, permanent, air gap between the two heatsinks.

Finally, stuff and solder the twelve 2200 microfarad capacitors.

SUMMARY OF FEATURES AND MIS-FEATURES

• De-Noiser circuit with jumper enable/disable
• Diodes on PCB
• Transformer Snubbers on PCB
• Big capacitors on PCB allowing true 1.5A output
• No bleeder resistors on PCB
• Voltage regulator ICs mounted on heatsinks for true 1.5A output
• Heatsink thermal resistance is 8.6 deg C per watt
• This is a four layer PCB, three of which are ground planes
• Gerber CAD files freely available, send to any fab you wish
• No "pilot light" LEDs on DC outputs
• Can use with either dual secondary trafo or AC-to-AC wall wart
• No SMD components; 100% thru-hole
• PCB unique identifier is TZPTR7 ; suffix -A means "revision A"
• PCB mounting holes compatible with Modushop chassis baseplate
• Board size 62 x 142 mm (edges) ; 50 x 130 mm (holes)

NEW INFORMATION

• 2023-12-21: a Forum member has created a new PCB layout of VRDN with only two layers. He has made the Gerber files freely downloadable in (this thread). Be sure to read his Disclaimer and non-Guarantee at the bottom of post #1.

I have a Quad99 system, with AE speakers, which yields a sound akin to analogue and I love it. Unfortunately, the CDs have started to skip after a minute: everything is fine, then it stops and the display goes ape with numbers randomly flashing on the screen. I suspect the laser lens is playing up. I use a CD lens cleaner but it obviously is not man enough to do the necessary. So, do I buy a new CD lens cleaner, or open the CD player up and try to give it a good clean (compressed air? vacuum?). Or, as it seems to work fine for a minute, could the laser lens transport be at fault? If so, how can I confirm?
Has anyone had a similar experience? How easy is it to get inside a Quad99 CD player? Any suggestions???
Cheers

Hello everyone,

I’m working on designing a Class H audio amplifier with a power output of 2000W RMS at 4Ω load, and I need some guidance from experienced designers.

I've previously designed a a 800W RMS Class AB amplifier for outdoor applications. Now, I want to develop a high-efficiency Class H amplifier for professional audio use.

Key Requirements:​

✅ Output Power: 2000W RMS @ 4Ω
✅ Topology: Class H with multiple rail switching
✅ High Efficiency & Low Distortion
✅ Stable Power Supply Design
✅ Thermal Management & Protection Circuits

I’d appreciate any insights on:
🔹 Proper rail voltage selection for efficient power switching
🔹 Best MOSFETs/BJTs for high power handling
🔹 Reliable protection circuits (Overcurrent, DC, thermal, SOA protection)
🔹 PCB layout guidelines to minimize noise and EMI
🔹 Any reference schematics, simulations, or design resources

If you have experience in designing high-power Class H amplifiers, I’d love to hear your advice! Feel free to drop suggestions, references, or even share your past projects.

Thanks in advance! 😊🎛️🔊

The Mission 778X is based around the design of the Quad Vena II and Leak Stereo 130 which both use the LM3886 as far as I know. Looking at internal pics I can see 6 forward pins on those two but the 778X has 8 forward pins so maybe a TDA7293? Specifications are similar for all the amps and are made in the same IAG factory. I assume when the 778X was designed, the LM3886 supply issues came into play.

I can't see enough through the vent holes and as it's new I do not want to remove the cover as well I do not have a Torx bit small enough.

Looking for a NOS pair, ideally in the UK or Europe.

Hi Folks,

I'm not an expert on SPDIF nor transmission lines, so bear with me please!

Background:
My good old well working Audio DSP has a digital output port with CS8420 feeding a AES3 (with TXP & TXN) and a Cinch (75Ohm) Output (with TXP) in parallel at Pin26.
The picture "original" shows the output schematic as far as I could analyse the (4-layer) board (pretty obviously with supply and GND layers). GND on the board is split between the output connector area and the circuit area (for unknown reasons) bridged with 0Ohm parts.

My Plan and request for confirmation or advice what to do:
The DSP is planned to drive 5m/15ft long koax stereo SPDIF BNC connections to each of my stereo speakers sporting digital 4-way "mono" x-overs. I.e. whether the speaker is "left" or "right" is selectable in the speaker. So, basically I need two parallel BNC Outs (would remove the cinch connector and disconnecting AES3 circuitry).

Please kindly have a look at the schematics and options I have come up with. I am sure my stuff needs changes to work properly, so I would be very grateful for recommendations "how to do this right".



Thanks a lot for reading all this and I hope you care to answer with any helpful recommendations.
Regards,
Winfried

Looking for 1pcs 2sk60 rank 5 or 6.
Can also trade for ...i have 100pcs 2sk170, some Black Gates, good European triodes, new in box ecc82 Telefunken, e88cc, ecc83, 3pcs rda1541
I have also 1 set of D3 tda1541 boards with 50% ,parts.
Maybe chinese board heavy modified es9038q2m with first generation Bisesik nanocrystalinev output transformers hi-fi tunimg silver gold 10A fuse new in box.
Thank you.

First, forgive me that I have not posted here in ages. I'm looking for help with my much loved Threshold FET ten/hl preamp. In my system the outputs of the FET ten/hl are connected to a homebrew converter using Jensen JT-6110K-B 4:1 stepdown line input transformers. The balanced outputs from my transformer box feed a pair of Schiit Tyr monoblocks.

The problem is I have been measuring about a 5 dB drop off from 40 Hz to 20,000 Hz at the outputs of the Tyrs.. My first thought was the problem might be caused by excess capacitance in the cables running from the transformer box. I swapped out cables. I removed the transformer box. I swapped out a lot of things. The only thing that fixed the drop off problem was disconnecting the FET ten/hl and putting in a different preamp. Nothing else made any difference.

Would anyone have thoughts or suggestions?

I've recently came across a new cheap little component tester that does way more than just the usual few basic tests.



This little beauty also can do basic DSO functions and is also reasonably accurate. There's also a signal generator and a full set of electronic diagnostic testing, including cap leakage/ESR, full inductor parameters and decoding IR remote waveforms, automatic pinout recognition of just about any device, including IGBTs, SCRs, diacs, triacs, all sorts of diode types, etc etc etc.

It run off an internal battery rechargeable through USBC and includes cables,, probes and connectors. The menus are very easy to navigate through and most tests are done very quickly. I've been trying to find specific devices it doesn't recognize or can't test. So far I haven't come across any (excluding ICs, but it will detect some lower voltage 3 pin Vregs!)

The best part is this tester is that its quite accurate and only costs $43!!!

Hi everyone, I'm a new member so let me tell you why I'm here! Since last year I started collecting vinyls and eventually found myself getting into audio equipment but I'm new to this, never really made any repairs myself. I know the basic of electronic but it's all text book, never really tried it out in real life.

For now I want to work on amplifiers mainly for turntables that I found in thrift stores or on marketplace.

That's it, lets see where this goes! 🙂

Hi everyone

I've been working professionally with solid state amplifiers for over 3 decades , and have been surprised by the sudden surge of devices going EOL. I believe that traditional class AB amplifiers are not going away, but are transitioning away from the mainstream to a boutique market. Valve amplifiers saw this happen a long time ago. With that said, there are a lot of high end solid state amps being built, and specialized parts sources are out there to supply them. I would like to list a number of sought after device type easily available, in some cases with a different part number.
They are available from Profusion plc in the UK. We have verified samples of everything listed here in pro audio gear made by LAR audio systems.
LAR audio is reason I had to locate some new sources.

KEC DEVICES (Korea Electronics Corp.):

KTA1381 / KTC3503 300V FAST VAS STAGE TO126 Replacement for the EOL ON devices really fast high voltage VAS
They should be in stock by Jan.

KTA1837 / KTC4793 230V VAS / DRIVER TO220 Replacement for toshiba 2SA1837 / 2SC4793 , good VAS and driver , in stock

KTA1700 / KTC2800 160V VAS / DRIVER TO126 Replacement for 2SA1220A / 2SC2690A , GOOD VAS and driver in TO-126
They should be in stock in Jan

KTB2640G / KTD1640G 150V 130W FAST DARLINGTON OUTPUT TO3P N Replacement for Sanken 2SB1647 / 2SD2560 improved SOA, in stock
This is a really nice output device for simple hifi amps

KTA1943A / KTC5200A 230V 150W WIDE SOA FAST OUTPUT TO3P L (TO264) This is an enhanced version of Toshiba devices, almost identical to 2SA1302 / 2SC3281 in every way , but higher voltage rating.

Unisonic This is not the most serious sounding name for semiconductors, but it is an established OEM supplier in Taiwan.

2SD669/A / 2SB649/A The TO126 versions are a good substitue for 2SA1220A / 2SC2690A There is some confusion about power ratings on the data sheets because they come various packages, including SMD, and the rated power is in free air We have verified that the SOA curves for the 2SD669A apply to all TO126 devices when on a heatsink.

2SA1837 / 2SC4793 The TO220 versions are also like the old favorite, but they are available in various case styles.

I may have post this in the wrong area, but searching threads, most appeared to discuss a specific need for a specific project. I hope I have been helpful to continue on with the construction of fine Transistor amplifiers.

How about this to start it off ?

The Class-A Amplifier Site - JLH Headphone Amplifiers

I'm a 2 year student of Ho Chi Minh university of Technology, Electrical & Electronic Faculty. I have passion on making amplifier so that is the reason why i choose that faculty.
It's nice to meet youuu.

I've had them for ages and it's time to move on. $600 plus $20 shipping in US . Probably over $70 abroad. Only transformers are for sale.

i have acquired a dynaco 150, she in a bit of a sorry state due to the previous owners tinkering, it runs but has intermittent noise in one channel, so instead of the usual recap and rebuild i was wondering if anyone had used the chassis and transformer and built something completely different inside, maybe a leach? not really interested in the update my dynaco path. so fire away, who has done what and what are some ideas. cheers.

So I have an unfinished over-the-top preamp project that I never finished using kit that is now un-obtanium. What is needed left to do is mount the Raleigh boards to the Bent chassis and wiring it all up. I am not sure what is a reasonable price to ask for this and would appreciate feedback regarding pricing, hence this post. Not sure if that is onside or not. Of course, I could bust out my soldering iron and wire it all up and sell it complete, but I don't seem to have the time and I am a bit unclear the best way to handle grounds when merging the two kits.


Bent Audio full balanced TAP-X with Slagleformers
This is loaded version of the Bent TAP-X, which has four Dave Slagle autoformers inside, fully balanced input to output. It has 4 balanced/2 single ended inputs, 2 balanced/2 single ended outputs and operates seamlessly switching among them. This one also has the Larger and Deeper Case, which permits a DIYer to add to it**. The TAP-X provides 1db steps and +7db gain, though passive. It comes with the very classy and heavy aluminum remote that switches all front panel functions.
Paid $2500 in 2014

**K&K Audio/Raleigh Audio Extreme Linestage Premium Version Kit with amorphous core iron, 6H30Ps, upgrades caps and resistors.
Kit includes Premium Line Stage assembled and tested by factory. Lundahl LL1674 amorphous core Output transformers, Lundahl LL1676 amorphous core input transformers (assembled on their own board), Lundahl LL1683 power transformer.

My kit does NOT include Analog Input Board and Select Switch, 12V Relay Power Supply, the Stepped Shunt Attenuator, nor the Raleigh Chassis from the Raleigh Kit as these tasks are all provided by the TAP-X above.​

​

Paid $2000-2500 back in 2014.
I still have the instruction manuals for each, if I can locate them. I have the Schematics of the Linestage and PSU as well from Dave Davenport. What is needed left to do is mount the Raleigh boards to the Bent chassis and wiring it all up. Also, the top plate of the bent chassis will need some holes to allow the Raleigh kit to breathe if you want to keep the lid on. I bought this before a major move and then never finished the project before moving permanently into the full active speaker setup (JBL M2s) which doesn’t require a preamp.

http://www.bentaudio.com/index2.html

http://www.kandkaudio.com/line-stage/

I can take photos of all the above, but this is more a pricing thread.

Got this PT below, as you can see it has 2 235v @ 60ma winding. Can I connect terminals 3 and 4 to create a center tap for a 235-0-235v @ 160ma?

Also the input is 115V @ 50Hz how will this be effected by a 60 Hz input.

Wanting to use it for a preamp.

BL

If you had a preference in a waveguide equipped dome tweeter, which one would it be?

It could be a tweeter that already comes with a WG or a combination of tweeter and WG you came up with.

There are a few which IMO are an afterthought just to be trendy, but tweeters which are engineered with the WG as a complete design are actually not that abundant.

My favorite soft dome in a WG is the Morel CAT378. It just does everything really well and can be crossed sucessfully 1st order with just one 3.3 - 5.6 uf cap. There is enough dampening at Fs to not need an LCR, but I still prefer one.

Mr Ohashi's new book will be available in early August 2025 from Amazon.co.jp

***** 03/03/2024 Update of project scope => move from UAC1 to UAC2 ******

Hello,

I'm putting on the bench someting in my head since a long time, that should not be too difficult to achieve for ones with know how... but could not be as easy for me ;-)

I want to program a "simple" USB to I2S, 8 channel device, UAC2, single sampling rate (48kHz or 44.1kHz), with Asynchronous mode.

Why UAC1: because ST proposes some UAC1 libraries, and no UAC2 ones (at least officially). UAC1 works without drivers on Windows and Linux.

SB Full Speed (FS) allows for 8x 48kHz x 1- bits (or 4x 48kHz x 24 bits). Great Monitor studios like Neumann KH150 have internal single 48kHz sampling rate. Many people consider that CD quality is "sufficient" and don't race for more bits, more Hertz. It is easy to find stm32 boards with USB FS, and more difficult with USB HS.

This would aim at providing the connectivity part to a 8ch DAC like ES9080 or AK4458. Overall target is 100% function and 99.5% performance of top products. for that niche need of multichannel. It could pair with RPi / Linux DSP offers like CamillaDSP for active speakers (or speakers / subwoofers combos...).

Plaform is a STM32F4 dicovery board to start with, then could be a STM32 black pill (seems to fit the purpose), which pave the way to dedicated board if needed. Those platforms unfortunatly have SAI, but if done well, the different I2S peripherals can be synchonized as slaves from a master, or all slaves from an external clock. A blackpill could be the USB "module" of a DAC board.

I would be happy if the the code could rely "as much as possible" on HAL libraries, code generated by STM32CubeIDE, and ST USB middleware. But, why did they made that so complex, with so much abstractions, while still needing to dig in all "layers" to fit/finalize code, and not really robust... I reallydon't like it so much, and it does not looks nice to me. But it should ease future portability (at least try to).

Intention is Open Source for the application part (not an expert about ST "mixed" licence for the USB stack but should be OK for DIY community).

Oh, if it already exists on Github or elsewhere, and I have not found it: let me know. I will be super happy ;-)

Current understanding is that, based on existing stereo code, I only need to:

• find all locations where the number of channels impacts the code
• change the nuber of channels from 2 to 8,
• ensure the consistency of the buffer size,
• on periodic basis, slit the 4x2 channels in sequence from the inputs to 4 distinct buffers that will each drive an I2S (through DMA)

Help of people knolageable about USB UAC and STM32 will accept to help... and I may have stupid questions.

I have an almost working stereo code based on ST examples. I have a first version that declares 6 channels, which is not working. I will come tomorrow with some questions...

Go 🙂

JMF