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.

I last repaired one of my 8 track tapes 🙂

I got volume 2 of "1950s rock and roll collection" and the pad was no good so I went out to find a good pad...... I found one and last night I went about fixing it....

When I opened the cartridge the tape came off the spool a little and I thought I lost the tape but I managed to get it wound back on the spool and into the cartridge and she plays fine 🙂

Now to get volume 3 of this amazing set.... Its a 4 record set,3 8 track set (Im not sure how many cassettes or CDs in the set)

Today I bought 🙂 (pics are not mine)

Harman Kardon Citation 11


Yamaha B-2


Nakamichi 480


Philips CD-104


Yamaha CT-610


Philips CDR-765


CEC DD-8200 with Shure V15 type IV


Grundig Fine Arts T-903 MKII


Next 😀

Oops! Almost forgot yesterdays goodie 🙂

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 recently rebuilt a pair of ESL-63s, and printed some 5 degree tilt back feet, but I don't like having the sound beamed up at my ears from close to the floor, so I designed and built some stands using 1"x1" and 1"x2" t-slot. The stands lift the speaker about 350 mm off the floor, and they are sturdy and center the weight so they aren't easy to knock over.



Details and link to CAD file here. Extra bonus: they are high enough that my cat will probably stop using them as a scratching post.

Looking at the listed data sheets for D3a by , Philips, Telefunken and Siemens, all give short data for triode operation with grid voltage of +10Vdc. Yet they don't show that area on the curves. Just wondering if anyone knows what the original idea for application was there.

Thanks

Based on PMs I've received there seems to be some interest in DIY STM32 USB-I2S boards. Earlier I have published designs and software for I2S input board mainly for ADCs (https://www.diyaudio.com/community/threads/diy-adcs.419922/). So let's up the ante and do the same for I2S output board for DACs. Over the last few years I've made several board designs with STM32F7 and STM32H7 having both I2S output and input or I2S output/I2S input only. I believe the STM32H7 based design presented here is quite versatile and should work well with most DACs.




HW specs:

• STM32H723 running at 550MHz
• USB3343 USB HS PHY. USB connector is mini-B.
• 16-pin header for output
  • I2S with 4x SD (digitally isolated). All I2S signals are buffered and routed through same number of gates/buffers to maintain timings. Digital isolators are lowest jitter 6-channel isolators currently available.
  • I2C bus (digitally isolated)
  • 3 output GPIOs (digitally isolated). In addition unused SD lines can be used as output GPIOs as well.
  • Isolator power (3.3V)
• MMCX connectors for I2S and external clock.
• 6 GPIO pin header for jumper configuration or I/O.
• 2x audio clocks (size 2520/3225) with independent enable control. Clocks can be either 22M/24M or 45M/49M. I use NZ2520SDAs.
• External clock input (MMCX). Digitally isolated clock selector GPIO for external clock.
• 20-pin header with 2x SPI (for touch screens) and encoder interface for display/control.
• I2C bus for displays etc. (independent from output I2C)
• Battery for backup memory and RTC.
• 6-pin header for SWD
• 4 status LEDs
• Powered either from USB or external 5V supply
• Board size 75x50mm (without outer mounting holes)

Currently supported protocols:

• I2S
  • 2 channels up to 768kHz/32bits (with 45M/49M clocks)
  • 4 channels up to 384k/32
  • 8 channels up to 192k/32
• DSD up to DSD512 or DoP256. Native DSD only in Linux as no ASIO driver yet.
• TDM
  • TDM128 (4 channels) up to 384k/32
  • TDM256 (8 channels) up to 192k/32
  • TDM256x2 (16 channels) up to 96k/32
  • TDM256x4 (32 channels) up to 48k/32 (not tested).
• PCM
  • 2 channel single ended or differential up to 768k/32
  • 4 channels up to 384k/32
  • Format: 2's complement, offset binary, simultaneous (also 16fs)
  • Bit depths: 16, 18, 20, 24, 32.

Other:

• HID interface for configuration
• DFU interface for firmware upload.

What is missing:

• ASIO driver. I've read that a 3rd party "universal" open source ASIO driver may be available later this year.
• GUI for HID interface. I have used simpleHIDwrite (linkhttps://github.com/benbaker76/SimpleHIDWrite).

The board works "out-of-the-box" with stand-alone DACs. For DACs requiring I2C or GPIOs a "driver" can be implemented. Currently I have driver implementations for ES9038Q2M, ES9039Q2M, AK4490, AK4493, AK4499+AK4191.

Connection to DAC can be made either through the 16-pin header or using MMCX for I2S with separate wiring for I2C, GPIOs and isolator power. In my DACs I have "sandwiched" the USB-I2S board with DAC board as all my DAC boards have same format and connector placement.




With display/control board, power supply (e.g. SilentSwitcher) and headphone amp board it is possible to build a compact DIY USB DAC having versatile functionality and good performance. Even bettering commercial offerings.



Attached are the gerbers, schematic and BOM. BOM has Mouser part numbers except for the NZ2520SDAs which are available from Digikey. Board is quite challenging for hand soldering. Smallest resistors are 0402 size and some ICs are leadless. I have assembled lots of these boards but I have a reflow oven. If there is larger interest it would probably be easiest (and most economical) to organize a group buy and have the boards manufactured professionally.

For sofware development I recommend STM tools which are excellent and license-free. For uploading software to an empty MCU a ST-Link programmer is needed. Easiest is to use a STM32 Nucleo development board which are available for about 15 EUR in Mouser. After initial programming DFU interface can be used for software updates. I will not publish the software source code here but it is available directly from me. If interested send me PM. Sorry, not for vendors.

This thread is about all ways to use OpAmp as input stage in power amplifiers.
Let us see your way to do this.
There are many ways to make an opamp control a power amplifier.
There are good ways, I am sure - and less good ways.

I have one way up my sleeve now, but I also have Choctaw in another thread.

I made this grasshopper script a while back that would dynamically make a multi entry horn from the output of ATH.

This is the sort of output you get



There is a lot of options down the side to adjust the ports and other things



The input it takes is the slices output from ath, you can select it using the Ath Slices File input (Right click and select one existing file)



I didn't end up adding the cone plug which I had planned to do, you can easily change all the dimensions currently then export it to some other software to slice up and 3d print.

I have uploaded the grasshopper project file if anyone wants to have a go with it.

As the beloved SSLV1.1 BiB shunt reg was getting long in the tooth, especially for NOS JFETS, I had in mind for some time now to design its successor. The goals were: 1. In production parts 2. Much simpler to set up. 3. Better technical and subjective performance.

After many breadboard experiments and two prototype PCB iterations I feel that my goals were finally met. So here comes the UltraBiB V1.3 😀

-Uses no NOS parts.
-Can do 5V to 40V output without changing a thing in its parts configuration.
-Nothing to choose and match. No tolerances in predicting its CCS limit setting.
-Has 45dB more open loop gain and many times less output impedance than 1.1
-Sounds easily better.
-Its an electrically and mechanically drop in replacement for an upgrade from 1.1

I have already given spare earlier proto boards with now deleted experimental features to few local beta testers. They are all happy by now as far as I know. The pictured board in green is the final layout in just cheapo proto that's a pain to rework and its pads vaporize in the end. It will come in proper grade black solder mask and immersion gold three sections board just like the original.

Attached: Zout plot for 150ma spare current and noise plots for 100,150,220,330,470,1000uF C2 (red 220uF). Also the rails probed on the scope for positive and negative sections. Here are typical values schematics also. I will write a PDF with instructions. Not that there is something truly special to consider when building it with the suggested parts but to describe it better as an item and to clarify details and precautions.

Updates:
31/5/18 R9's value update
1/6/18 OL sim at 100mA spare
2/6/18 Build Guide PDF added (0V0b)
24/5/19 Triplet board dimensions picture
21/6/19 The PF5102 JFET became EOL. But GB long term supply is secure. For substituting with J113 see info in post #1559
15/5/20 BC560C became EOL. But Group Buy long term supply is secure. BC559 can be used up to 30V output. BC556 for 31V-45V output.
2/7/23 LNA instrumentation measured a very low 2nVrtHz spectral noise density from the UltraBiB under real working conditions. post#3,688

Dear Community,

I recently watched a video about the B&C coaxial compression drivers. After looking them up i was shocked how pricey they are. Then i saw the diagram below and thought: can't we make our own coaxial compression driver with conventional cheap drivers/parts?
When thinking about the concept i became more and more excited to try.
Directivity control provided by a horn + point source behaviour like a full range/coaxial driver = why not?

Let's discuss the possibilities and try to succeed!




So here is my fist take on a design concept of a DIY coaxial compression driver (will name it CCDIY):

• HF: 1 inch dome tweeter like the Dayton Audio ND25FN-4 or Peerless by Tymphany OC25SC65-04 etc.
• MF: full range/ midrange driver. about 2.5 or 3 inches, like pdf_dayton audio_PC68 or a pro driver from Faital etc.
• using B&C phase plug design, more info in the link of the video.
• all in a tight package:

In light blue: HF dome tweeter;
In orange: basic shape of the midrange driver with cone;
in dark blue: phase plug for both drivers (is not finished, need to implement the B&C patented method for blending the HF with the MF);
in grey: front and inner channels in a housing.
Thinking about 3D printing the parts after first design prototype is ready.
Also, the dome tweeter has to be able to dissipate some heat. still thinking about solutions for that.

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.

TU-8850SE KT170 SEP AMP 16W X 2​

Available in September 15, 2023

$1,185 with stock OPT​

$1,750 with Lundahl LL3727/90ma​

​

Audio Note Tantalum Resistor set is available 26 x 2 pcs = $286 singnal path only ​

When I was young, we spent hours on end playing what we called word game. Same rules but it was always 4 letter words.
I was introduced recently to this new sensation and tried it today. Now I know it's beginners luck, but first time out I got it in 3 words.
I think I'll try again and let it bring me back to earth.
Who else is doing this?

Use this link for more details:
https://electrodac.blogspot.com/p/dac-ad1862-almost-tht-i2s-input-nos-r.html
Note: 8th pin on the I2S input header is missing connection with the GND (on some of my PCBs this pin is unconnected, if you are going to use it, solder it with the nearest GND pin on the header)


People who are selling brand new unused AD1862 or AD1865 chips:

• Paddy Garcia new group buy
• https://www.diyaudio.com/community/threads/ad1862n-j-dac-chips-for-sale.405922/
• https://www.diyaudio.com/community/threads/ad1862n-d.417897/

I want to thank @Paddy Garcia for the whole happy DIY community. He did very honorable work: Group Buy #92



Special tested DACs:
DAC702BH/CH (Glue logic - I2S data input) (result: audible digital noise on the output, not good for audio use)
DAC8811 (CPLD data input) (result: audible digital noise on the output, sound far away from very good audio, don't waste time and money on such DACs)
...

Edit Nov 5, 2022: nice analysis and summary of DML materials and response with python script by @homeswinghome

https://www.diyaudio.com/community/...s-as-a-full-range-speaker.272576/post-7159983

Edit oct 27, 2021: a nice Google drive library of DML publications by vdljev:

DML - Google Drive

Edit April 1, 2020: nice summary of DML patents by Burntcoil

A Study of DML's as a Full Range Speaker

Edit Aug 12, 2019: Nice summary here by BurntCoil on how to maximize performance.



I had some cheap exciters that I got from PE a while back and tested them out a while ago with a full 20x30in FC panel here:

Foam Core Board Speaker Enclosures? - Page 225 - diyAudio

I found that a large 20in x 30in panel can sound quite good with nice bass extension and a snappy transient response:



Impedance:



Frequency Response & Harmonic Distortion:



Impulse Response:



The results were interesting in that it sounded nice - with surprising bass and good midrange. I thought nothing more about it until recently prodded by the master of DML, CLS. He has worked on this extensively and is a treasure trove of good info. I since have discovered that you can really do some cool things with them by playing with driver placement, cutouts, adding mass, adding felt, adding ribs, making them huge, making them multi-way FAST, etc. the options seemed almost limitless. Take for example, a large center channel and a super 40Hz capable multiway that CLS built here: PIEZO NXT type panel - Page 60 - diyAudio and PIEZO NXT type panel - Page 61 - diyAudio

I feel like this technology just isn't getting enough attention. There are several very large threads on this on the web. It might be tough for a newbie to comb through. I am a newbie at this so thought I would document my journey for the Full Range forum to follow. I think it really could be a great full range speaker with some careful experiments. This will be really useful with some modeling using CAD and FEA modal analysis - such as available in many CAD packages like SolidWorks. One can play with shapes, cutouts, mass loading, variation in thickness and materials, boundary clamping conditions, etc. The idea is to spread the modes evenly with not any one mode dominating and causing a spike.

First thing is to play with it to get a feel of what we are dealing with. What is nice is that it is relatively cheap to play with. Exciters cost $3 to $20 ea and can be made by removing or cutting out large holes in the cone from an old driver and leaving the spider and VC to attach to the panel. Foam core or corrugated cardboard doped with shellac or PVA seems to be the materials of choice.

Some interesting facts:

1. Although it has no baffle, it is not an open baffle (OB) dipole in behavior - that is, there is no huge bass rolloff and it hits surprisingly low (40 to 50Hz is easy) for a zero baffle driver.

2. It is not a dipole but behaves more like a bipole or an omni.

3. It has very quick snappy transient response - nice drum sounds.

4. It is sensitive to how you mount it or frame it or hold it.

5. You are building a driver in reality - a driver and zero enclosure.

6. You want to avoid symmetric shapes and symmetric exciter placement in order to reduce the effects of the main symmetric transverse drum head eignmodes. Think reflection anti-symmetric shapes like uneven trapezoids, pentagons, blobs, etc.

7. It operates more by having high velocities and large areas for good efficiency vs large displacements - thus small drivers and large panels can be surprisingly loud.

8. The impedance is essentially "flat" relative to normal drivers in that there are modal peaks (many of them) but they range from nominal Re value (say 6 ohms to maybe 8 or 9 ohms throughout the 40Hz to 20kHz range - this presents a very flat load to an amplifier.

Here is a photo of a basic panel I was testing (right before I cut off some edges to form a trapezoid) - panels is about 1 square ft in size and made of standard dollar store foam core board. I suspended it with two pieces of twine from the top corners between a ladder to reduce the effects of edge boundary clamping:



Then I started to play with mass loading by adding blobs of modeling clay (8, 5, 4, 2, 1, 0) so you can see the effect on the resonance modes and the harmonic distortion. What is neat with this test is you can do it live while playing music and immediately hear what sounds more pleasing to the ear. Note the 50Hz bass extension. It is also surprisingly efficient with 85dB and a puny little voice coil. These are 0.5m and 0.71v for equivalent SPL at 2.83v and 1m.

Frequency Response and Harmonic Distortion for...

No added mass:



8 pieces of added mass:



5 pieces of added mass:



4 pieces of added mass:



2 pieces of added mass:



1 piece of added mass:



Impulse Response of 1 piece of added mass, note the sharp transient capability with relatively low after pulse or ringing:



I liked the sound of the 5 and 1 mass the best and for a FAST with a 200Hz XO, the 5 mass might actually do fairly well. Of course, sound clips to follow once I have more time to work on this. Just to demonstrate how quick and easy this is, I did all these experiments in less than an hour including making the DML.

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 ?

I'm at a point where i am confident in this board design, at least for the SRD-7. I have minimal interest in commerce and i know there's not much money in this, but i did order a stack of 30 boards.

It's largely a distillation of the official schematic for the SRD-7 MkII and modifications recommended by a few other tinkerers, so i don't really take credit for the circuit itself, just the board design. I'm using the same kind of cockroft-walton voltage multiplier as Stax used for the bias supply. Ballast resistors are 4.7M. Optional bias reserve caps moved to *before the ballast resistor. Noisy PTH devices replaced by regular power resistors. Room enough to reuse vintage resistors or to use whatever boutique power resistors you want.

It can also be used to convert a self-biased energizer to wall-powered bias. It *may be able to give you a self-pro-bias energizer out of an old SRD-4 but I haven't tested this yet and i may need to revise the design for that. I also haven't tested fitment in the SRD-6 yet and at a minimum it needs another hole drilled, or some foam mounting tape, as i hadn't realized that the mounting brackets in the SRD-6 are 100mm center to center.

I've also tested it in an SRD-5.

In an SRD-7, you basically desolder wires from the original board and solder them to the same position on the new board. The exception is that I've provided a position for a fuse when adding wall power (or upgrading an SRD-5).

The BOM is less than $10 from mouser for one board.

$10 plus postage, will come in a padded envelope with a short length of 2.85mm black 3d printer filament that you can use to plug the center hole on the socket you upgrade to high bias.

If i have enough parts left over when i am done testing and upgrading the energizers i have on hand, i may offer one or two fully populated boards. I Really don't want to upgrade your energizer for you. The SRD-7 pictured below is actually an earlier prototype layout, for full disclosure.

It can also be used to build a fully DIY energizer with high-quality transformers from Lundahl or similar, or whatever push-pull output transformers you got (must have a center tap on the HV side).

I'm not tryin'a be in business, I'm just proud to have produced it. You don't have to spend hundreds of dollars on a direct-drive amp for any modern staxen - you can modify a good old SRD-7 and use it with whatever amp you got. Dip a toe in and see what 'stats are like. Even the ancient ones are good, and yes the modern ones are mostly better.

Standard disclaimers apply - uses wall power and generates high voltages so exercise due care. Applying too high of a bias voltage to your electrostatic headphones may destroy them.

I asked AI. If there's something better please comment below. TIA

Question: Can you recommend good quality brands of wood polish mentioned in the list below for vintage KEF speakers from 1974?
(I live in Portugal, so please only European products)
• 1st. Furniture polish
• 2nd. Scratch remover
• 3rd. Wax stick
• 4th. Wood stain or varnish (for exposed wood)
Important considerations:
• Do not use polishes that contain silicone or excessive oil.
• Although it is most likely a varnish, try to confirm the original finish if possible. This will help you choose the right restoration products.

Answer: Absolutely! Finding quality products in Europe for the restoration of your vintage KEF speakers is perfectly possible. Since you live in Portugal, I will recommend some European brands that generally give good results for the types of products you are looking for.

It is important to note that the availability of specific brands and products may vary by store and region in Portugal. I recommend checking out DIY stores, furniture restoration stores, and some larger drug stores.
Here are some suggestions for European brands in each category:

1. Furniture Polish:

• Borma Wachs (Italy): This Italian brand is well-known for high-quality wood treatment and finishing products. They offer a wide range of polishes for varnished furniture that are silicone-free. Look for products like “Borma Wachs Polish” or liquid waxes specifically for varnishes.
• Liberon (UK/France): Although this brand originated in the UK, its products are widely distributed in Europe. Liberon offers excellent wood polishes, including silicone-free options like “Liberon Beeswax Polish” (pure beeswax, which can give a soft shine to varnishes, but test first) or varnish-specific polishes (“Liberon Furniture Polish”).
• Blanchon (France): Specializing in products for wooden floors and furniture, Blanchon offers good quality polishes to maintain and enhance the shine of varnished surfaces. Check out their range of "Huiles d'entretien" (maintenance oils) or liquid waxes.

2. Scratch Removal Product:

• Borma Wachs (Italy): Again, Borma Wachs offers products specifically for repairing scratches on varnished wood, such as touch-up crayons ("Retouch Crayons") or fine polishing compounds ("Abrasive Paste for Wood").
• Liberon (UK/France): Liberon has products such as "Liberon Scratch Remover" which is formulated to disguise superficial scratches on wood finishes, including varnish. * Möbel-Reparatur-Set by Akemi (Germany): Although best known for stone products, Akemi also offers furniture repair kits that include materials for removing scratches from wood and varnish.

3. Wax Filler Sticks:

• Borma Wachs (Italy): Borma Wachs has a wide selection of “Wax Fillers” in various colors to repair deeper damage and scratches in wood and varnish.
• Liberon (UK/France): Liberon also offers “Wax Repair Sticks” in various shades to fill chips and scratches in wood furniture.
• Picard (Germany): Picard is a German brand known for tools and materials for craftsmen, and they offer wood repair wax sticks in various colors.

4. Wood Stain or Dye (for exposed wood):

• Borma Wachs (Italy): Borma Wachs offers a wide range of wood stains and wood dyes in a variety of colors and shades, ideal for touching up areas where the varnish has been damaged and the wood is exposed.
• Clou (Germany): Clou is a German brand specializing in varnishes, paints and wood care products, with a good selection of wood stains and paints.
• Sayerlack (Italy): Although more focused on industrial finishes, Sayerlack also offers high-quality wood stains and paints that can be found in specialty stores.

Important Considerations (to Remember):

• Confirm the Finish: If possible, try to confirm that the original finish is indeed varnish (which is highly likely for these loudspeakers). You can do this by checking whether the surface is hard, smooth and water-resistant. Varnish usually forms a thicker, shinier layer than wax.
• Avoid Silicones and Excessive Oils: The brands mentioned above usually offer silicone-free options. When choosing a product, read the description and ingredients carefully to ensure that it does not contain silicone or excessive amounts of oil, as these can interfere with future finishes or attract dust.

I have just picked up a Marantz MR2020 for my daughter.

It has a buzz on one channel:

• fault is in the amplifier, not the cables or speakers
• buzz remains on all sources
• buzz is a constant volume
• the buzz goes at volume 0 plus when other muting operations are in progress

I was going to open it up and have a poke around with my DSO. I am expecting that it is faulty capacitors. I do have a copy of the service manual with the schematics.

Any other suggestions of what could be wrong and where I be starting starting?

Many thanks, George

We have just picked up a rather special Marantz Layla MR2020 for my daughter. The CD on occasion skips. It plays discs and advances tracks fine.

I was going to open it up and lubricate and replace the laser mechanism. Not sure which one it is, so will find out when I have it open.

Is there anything else that could be causing this issue?

Thanks, George

I looked around and can't find a good place to file this thread, so if I missed something, Mods, please feel free to relocate.

I am about to purchase an Alesis BRC which will (hopefully) control my HD24 [24-track hard disc recorder]. It was originally designed to control many ADAT machines, daisy chained together. ADAT uses SVHS tape for audio recording. Anyway the HD24 is 24 tracks od digital recording onto hot-swappable hard drives. But since access is orders of magnitude faster on the HD than spooling tape, there can be sync issues... (it CAN allegedly daisy chain HD24s to provide 48+ tracks, up to 128 max or something silly like that)

I have the Alesis service manual, and it indicates an appendix for the schematics, but they are not there. It must have been a separate file or insert added at the repair facility. I do have the "confidential" service tech final testing procedures at the end of the copy of the manual I have so it seems to be "complete".

Please have a look around and see if you have one or can point me to an online source. I can't find it anywhere and I was digging pretty deep...

Thanks in advance for any help!

Folding a new bass horn design or adjusting the dimensions of an existing design to meet a specific requirement can be somewhat of a challenge and prone to error. To make the process easier and to avoid errors as much as possible, I've created a number of different Excel workbooks that can assist with the folding process. All of them should be considered as "works in progress" as I'm continually adding new features and of course fixing the odd bugs that I come across, so always make sure to check here for the latest version before using any of these workbooks. Some of them I've actually used to design and build my own subwoofers (the best way to confirm that the workbook in question works), and in those cases I've included a link to the project sin question. If you have any questions about the spreadsheets, please post them to this thread, thanks!

The latest versions of the workbooks are available at this location:

The Subwoofer DIY Page

Thank you for accepting me here in this forum

USSA-5 build and review:

Amplifier Design/layout: Fab (The Great and Powerful Ape)

Amplifer PCB Layout: Alex

PSU Design/Layout: Project16

PSU PCB Production: Prasi


I couldn't stop thinking about this amplifier. The obsession began with a few fleeting traces of it's existence. The few members that have actually completed it were mostly silenced by its greatness. When I had enquired about it, people simply said "Oh, the USSA-5? It's the best amp I've ever heard."

I had to build it. But, the great mind that designed the amp wasn't going to just give me the boards. I offered to build a children's wing at his nearest hospital in his name. He said no. I said I would make huge donations to wildlife preserves around the world. He still said no. He was like talking to Willy Wonka and I was Charlie Bucket. I needed a golden ticket....

In the end, what worked was honesty and a promise to do two things: to build the amplifier and write a review.

So, here it is.

For sale 2PCS Sylphaudio D400M PFFB TPA3255 PBTL mono
one of the two has a slight problem: the output level is slight lower then the other.

Price 120 € + shipping (2 connex SMPS 600RS available)

Hello all,

I am working on my DIY Preamplifier new project based on the Elektor Preamplifier 2012 (Douglas Self) schematic.

I don't need the TONE and BALANCE control sections in my design, painted in Red in the picture below. I will directly connect the output of IC2A pin 1 into the input of IC9B pin 5 (Green traces).

Still I have this question regarding the filters in the input of IC2A pin 2 and 3, painted in Yellow. Do I need them or they are part of the BALANCE section only?

And if needed, should I replicated the R11-R12-C6 filter with R8-R9-C5 connected to ground?

Thank in advance 🙂

Hello, I am planning to build a small mobile wifi speaker for my daughter. As main board I'll use the orange pi zero 2w (because i already have it, it has usb c and 5ghz wifi and it's small) and i have a small 3W speaker (not tested sound quality yet, but I guess it should be good enough).

For in between i was thinking a mono i2s amp, since that doesn't require a separate dac or psu. So i got the MAX98357A

Turns out, the orange pi doesn't have i2s. It's there any other way to use the amp or do i need to get a different one? And if so: which one do you recommend?

Would be grateful for any information or guidance.

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.

This thread is a continuation of woody1's thread (flat to 40Hz) which can be found here: http://www.diyaudio.com/forums/subwoofers/170749-15-tapped-horn-rcf.html

Epa, Xoc1 and others have been contributing their time and effort in attempt to help me reach my goal: Flat to 30Hz with as much clean output as possible - XTRA points for breaking physics and keeping it small. Heh.

We have found 15" drivers are limited by xmax far before maximum power is reached.

Here's where we left off:

Xoc1 said:

Post 73 (Epa) @ 2.83V
30Hz = 95.28 dB
40Hz = 98.58 dB
3.3 dB difference

Click to expand...

Xoc1 said:

Post 98 (Xoc1) @ 2.83V
30Hz = 96.17 dB
40Hz = 100.74dB
4.57 dB difference, so step from 30 to 40 Hz is 1.27 dB more...

Click to expand...

My REAL 6 Fold mod (I meant to post) @ 2.83V
30Hz = 98.29
40Hz = 98.48
0.19 dB difference, so step from 30 to 40Hz is 4.38 dB less AND 2.12 dB more output at 30Hz






What's your take on this version?

Xoc1 said:

Post 98 is more efficient at both frequencies as would be expected as it is bigger. External is 600 litres compared to 397 for post 73.
Biggest misgiving I have about the post 73 design is the small S3 dimension.
At 1236 this is about the same size as SD. You can see how tight the mouth is positioned to the rear of the driver. This would require care and adjustment to make it work due to the driver seriously restricting the mouth. I suppose an access panel would be needed to actually get the driver in the box!😱

Click to expand...

epa said:

restriction is not to bad because of the neodinium magnet.

Click to expand...

epa said:

i designed the post 73 th for the use in a cardroid sub aray.
this is why its symetric.
i like xco1 model 2,its a easier bild but for me a little to big.

i wil work this weekend on it to give you al the details😉
meanwile here are some designs,most of them are posted here.to make your choice more diffecult😛

Click to expand...

Must they (post 73) be used in a cardioid sub array? Will I/we loose any performance if they're not?

epa said:

yes the idea was to make the front parts removeble 🙂

Click to expand...

(Referring to post 73)

epa said:

ok lets make it 15mm.i think 18 is a bit over the top.
horizontal there are only small spaces ,and vertical wil do some bracing in the middle.
front wil be removeble in one piece.
it might be a good idee to send you the dwg,dxf file when im finnished.
then you can make a 1 on 1 print on A0 paper(it fits)in the copy store.
cost about 4 euro here on this side of the pond🙂

Click to expand...

epa said:

justin ,we need to make a new topic,because this has no longer anything to do with woodys design ,

Click to expand...

Welcome 🙂


Any others: feel free to post your design if you believe it betters any of these and fits the goal posted above. Many thanks!

Justin

Folks,

Over in the TDA1541 Meta Thread we had a lot of input to analyse the TDA1541, what makes it tick and how.

https://www.diyaudio.com/community/threads/building-the-ultimate-nos-dac-using-tda1541a.79452/

What I would like to do here is extract the practical part and propose a complete DAC design that may be realised as open source group project.

I would first propose a basic structure.

There will be a main board for the TDA1541 with a super capacitor based "stacked" 20V supply, optimised layout for the various decoupling capacitors needed as discussed in the other thread.

There will be attenuator/slew rate limiter circuits and a set of Flip-Flip's for reclocking and / or division of BCK & LE/WCK as well as a DEM Clock divider from a 512X MCK. The aim is to run a continuous BCK with the lowest frequency possible, so the default option will be I2S with 32X FS BCK, as the lowest sensible without FPGA/CPLD/MCU etc.

There will be embedded clock generation with 22.5792 & 24.576MHz clocks and dividers that allow full slave operation of a frontend at either 32X FS or 16X FS. It will also accept two external 512X MCK from (for example) an SPDIF receiver or USB Board with embedded MCK out that cannot be slave clocked from our on board clocks. Super capacitor based 5V supply.

Space for an analogue stage "plug in" to keep the choice open. My own intention is to use a SuperSziklai Pair based design detailed in the original thread.

Space for a SAA7220 socket that could be filled with an SAA7220 or any PCB holding any FPGA, CPLD, ASRC or whatever. Space will be be left around for a larger PCB to fit this socket. Again, this leaves the choice what goes into this socket (including nothing) to the builder. For my own setup an SRC4190/AD1896 will be fitted to operate as 64X FS to 32X FS I2C converter and as optional dejiiterer for jittery SPDIF Receivers and sources that otherwise sound worse than ASRC engaged.

Anything else will go. Pick your poison. If the 74ACT2226/28 returns from "active but zero stock deads" it may become the FIFO based I2S 2 SIM converter discussed here, otherwise SRC4190 in bypass makes a swell FIFO & format converter with a bonus ASRC):

https://www.diyaudio.com/community/...g-discrete-logic-collaborators-wanted.424459/

Input choices will include standard (Amanero) footprint for USB Boards (mostly XMOS) with isolation via SPDIF (no I2S digital isolators allowed, why will be discussed in good time) and a clock feedback from our crystal clocks with 44.1/48 base selection via opt-isolated pin, probably at least two selectable options.

SPDIF Inputs with SE BNC, XLR AES and Optical plus USB. The SPDIF Receiver position gets a CS8412 DIP socket that accommodate any number of SPDIF RX Solutions, all the way to "roll your own" with FPGA/CPLD/CPU.

I'd draw a block diagram if needed... is it?

So, if anyone is interested in making this a reality as an open source project, join into the discussion.

This is not a channel to debate fundamentals, redebate stuff from other threads or "can you make it into this thing?". If you choose to actively contribute time, effort or money to the project you can of raise suggestions.

I expect to hear crickets, cicadas and the occasional lizard chitering on this thread, but G*d love's a trier.

Thor

PS. As an additional note, all real PCB work will be in Altium, not Kicad or similar educational toys. All simulations and initial schematics capture in TINA-TI. This is non-negotiable.

If you need help with Altium Designer, we can talk in private how to use the free download with full functionality for evaluation and educational purposes beyond the eval time limit. TINA-TI, download at TI.

Have no idea in which forum this question belongs, but for the time being the subwoofer forum seems most appropriate.
Is there a DIY technique to measure the BL curve of a woofer?

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.

There is a recently applied patent that takes another very different look at making an electrostatic loudspeaker. Particularly the way the stators are made.
I contacted the patent holder and asked if it was OK to give a DIY approach on his design a go. As long as I was not promoting it commercially he was OK with it.

So here is a short video of the very open design as a proof of concept. Yes it works and it works fine.
Sensitivity is very acceptable, distortion is low, but what is most remarkable and I don't know of any way of measuring that, is how OPEN is sounds. That was also most likely the intention of the design in the first place.
The designer claims a complete break on the relation between open area and capacitance, but that is, sorry to say, not true. There is a fixed relation between the two, but obstructions area is much smaller than with wire or PCB stators.

Took several days of 3D printing the 500+ parts that are in there and several weeks of manual labour to apply and assemble the whole contraption.

Per panel the square space is like a Quad ESL-63 panel. Eventually we want to try and build a complete speaker with 4 of these panels each. The ultimate goal would be to apply the delay rings as well.

Panel size Mylar surface 58x17 cm. That is 98.600 mm2 of surface area.
Obstruction surface is only 18.300 mm2, only 18,5%.
Open surface remaining is a whopping .... well you can see it in the short video.
Stator to stator capacitance is 90 pF.
Diaphragm movement space twice 2,5 mm (also like the Quad-63 panel)

Please note that this is a patented design and done here just for the fun of proving that it can be made with simple means and to have a listen at the characteristics without having to travel half the world to hear the original concept. There are still a lot of thoughts on how to change/improve upon what is demonstrated here.

Full meaning the whole thing.....

I last listenend to my BEST OF MOUNTAIN record 🙂

As per the title really.

I work up far more designs than I usually publish on the forum or ever normally see the light of day elsewhere. Some were done at request, some are simply quick design exercises or checks / investigations, some I can't actually remember the reason for doing. I don't always have time to systematically catalogue things, and they can end up lying around on one of my HDDs for months until I sort through them. I'm currently in the process of going through one of my old external USB HDDs before it's recycled & found quite a few. Most weren't worth bothering with, but there were some more practical / interesting ones so rather than scrapping them, I thought I'd start a thread that can essentially be a bit of a dumping ground for misc. designs that I've done that either haven't found a home elsewhere, or that have previously been posted but got buried on other threads, so at least they're concentrated together.

Advance warning:

-There is little consistency in the layout / sketches. I'm lousy at CAD, and there's no way I can inflict more than a small portion on the long-suffering Dave to draw. In these cases, it's probably not worth a high quality drawing anyway. I've quickly re-saved most into png format to save space & done a little tidying if needed. The rough date is sometimes mentioned, sometimes not, depending if I remembered to add it at the time. Sometimes damping is shown in the sketch (all done in Windows Paint so they are very basic), sometimes it isn't, but I always refer to it in any accompanying notes. Same for bracing or vents / ducts; I often don't show the dimensions but they're in the notes. There may be some errors; I don't think there is anything major though. Normally I work in Imperial & assume a baseline 3/4in build / sheet material thickness. Some are metric, in which case the assumption is 18mm. Change as desired, providing the internal dimensions are not altered.

-Most of these are physically quite simple; vented box variations, some MLTLs or ML-Voigts etc.

-I'll add periodically as / when I find things. All are free for personal use.

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.

I am new to the world of personal audio. I come from a background of music appreciation/collecting/archiving and a bit of hobby digital DJing (yes, I am not a "real" DJ lol). I decided to introduce my wife to DJing and my first step was to purchase two V-Moda M100 Master headphones (for shared listening) and Pioneer DJ VM-80 speakers. In order to connect these to my laptop, I bought a Zen DAC 3 and an iFi Audio 4.4mm to XLR cable one month and a Zen CAN 3 the next month. I use an iFi Audio 4.4mm Pentaconn analog cable to connect them. After I did that, I bought two iFi iPower X power adapters. Each step in this process brought even better audio to the setup.

I joined a message board and began to look for ways to improve this setup even more and came across information about what is the best USB cable to purchase. I was informed that isolators/regenerators/reclockers were good to have as part of the chain between laptop and DAC, and I was recommended to purchase an Altor T1 from Alex Torres.

The small .070 gram USBISO arrived after a month and both Alex and I were challenged to figure out exactly what was wrong with it. In the end, I realized that nothing was wrong with it! This is how we figured out together what I had to do to get it running.

When I first plugged the USBISO to the DAC, there was no connection, but its red LED light was lit (even before connecting it to the first iPower adapter that came with the Zen CAN 3). I used a Gothic Audio Semper Fi The Outsider .3m cable for both before and after the isolator. So, that was strange. Later, Alex suggested that I should connect either a removable storage or mouse to the isolator, which I did, and there was "energy" but no "data" running between these.

Alex didn't know why and suggested that I should return the device, but I decided to run a diagnostic because I didn't want to give it up so easily to return it for a refund or exchange. I chose to run this particular diagnostic by connecting the isolator to my wife's laptop and I noticed that immediately the cursor worked. So, that was interesting.

Here's the key part of this story:

My laptop is relatively new, purchased in January of this year. When I got it, I did something kind of geeky and tried to remove everything that I thought was nonessential, however, in the process I most certainly removed something that communicates data between laptop and devices. I am not sure, then, why the DAC worked, but I decided to save all my data and reinstall_Windows. That allowed my isolator to work. Problem solved! I am not sure how the Isolator sounds without the Gothic Audio cables but the entire setup sounds great. I am very glad for this tiny device, but most of all for Alex's patience and guidance because I tend to overreact and send multiple panicky messages before he gets a chance to write back.

There is so much more that I can do with the setup, but given a limited budget, I am going slowly and step by step to try to have the ideal setup for our little jam parties at home.

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!

After finishing some electronics projects I'm now into speakers again. Call me crazy but I've really like to build something big this time. Currently my main system is build around a Limmer 630BC-flare with B&C DE360 and 6NDL38 - supported by a 10" Wavecor for the lows. Really nice whats coming out there - but you all know how life is going and we all are never finished with our projects.

So this time it will be something big and extravagant. I've bought the 3D drawing of the ES-290 by @Joseph Crowe yesterday and they are prepared for 3D printing already. Means: The big chonky horn is divided into eight pieces per half to get them on the bed of my printer. I'm using Fusion 360 for this. I've also prepared the single parts to be put together again using holes and dowels as you can see in the figure below.




It's looking neat and tiny in Fusion... But my printer bed is measuring 220x220 - know do your maths 😵. My CAD workstation got into some sweat during preparation.

Currently I'm doing some evaluation about printer settings, layer height and printing speed for the best quality with the least rework as possible (well the latter will always be a dream in reality - but let me stay dreaming for now). The evaluation models are printing using nearly no and the simplest infill for saving time (10% grid). It's just about how smooth it can get. The first try was using 0,5mm layer height with a 0,8mm nozzle (taking 14h for the bigger part in the picture below).





Not satisfying - so currently I'm doing another print using 0,3mm layers with a 0,6mm nozzle and slower speed - the example above was printed with 120mm/s. I'm at 60mm/s now and it's looking a lot more promising by now - but not as perfect as such an expensive and time consuming project could be in the end.



So I decided to use 0,16mm layers with a 0,6mm nozzle for doing the calculations for "final manufacturing" with a much bigger wall strength (set to 17x0,6mm -> 10,2mm walls) and stronger solid bottom- and top layers (10mm). I also calculated with much more and multidimensional infill - 50% gyroid for the direct exposed parts and equivalent less dense for the outer and rear parts should be enough.





Everything is prepared for using a single 1000g respectively 3000g roll filament per part as it can be seen above (green/yellow highlighted weight). I'll print using PLA filament because it's tought when using the correct settings and good for glueing without trouble. It's good behaving during post-processing (grinding, fill-ups), too. Last but not least it's easy to get in 3000g rolls too. PCTG would have been my favorite but it's not that easy to handle for glueing and there are not that many offers for 3000g rolls. I've been using PETG-CF in the past for adapters and baseplates in other projects but that filiament is not that good for doing smooth things like the inside of the horn.

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

Hi Everyone,

Just letting you know that Hornresp Version 18.00 has now been released.

The Tapped Horn Wizard tool has been significantly enhanced. For a given overall horn length, flare rate and driver specification, the optimum driver position can be determined more easily than before.

Note that there are three ‘Interaction’ options. The operation of the first two should be pretty obvious from the titles. The third ‘L12 & L34 Linked’ setting enables the user to “slide” the driver along the mounting baffle / separating panel towards or away from the 180 degree bend (similar to changing the acoustic path length of a slide trombone).

Hopefully the Tapped Horn Wizard is now a far more useful tool. Enjoy 🙂.

Kind regards,

David

Here is my latest Tokin SIT amplifier. It is a DEFISIT amp (Depletion Enhancement Follower SIT) output stage based on Nelson's First Watt SIT-5 output stage. Diyers more knowledgeable than me picked apart what information Nelson provided in his SIT-5 owners' manual (https://www.diyaudio.com/community/threads/first-watt-sit5.418023/), and I rode on their coattails and came up with this version.

My build is only the output stage, and it incorporates the duo output capacitor arrangement which can be tweaked to adjust the amount of current amplification that the mosfet part of the DEFISIT push-pull output contributes to the total current amplification. This is a departure from the regular DEFISIT amplifier, where the SIT and mosfet both fully contribute to the total current amplification.

Here are a couple of LTSpice simulations. I did many simulations, but these two are what I based my design on. I chose -47VDC for one sim since I intended to build it using one of my existing amplifier chassis and power supply. I chose the -63VDC for the other since that is a reasonable voltage that will take advantage of the power capability of the THF-51S and still keep the power supply voltage reasonable for capacitor voltage and heat sink heat dissipation.

As shown in the simulation results (47VDC power supply and 2.0A Iq), the measured current through C4 connected to the SIT drain was 361mA peak, the measured current through C7 connected to the mosfet drain was 150mA, for a total of 511mA. As a check the current measured through the speaker, R8, was 509mA - close enough. So the contribution of the mosfet was 150mA/511mA = 29%.

I have not included the LTSpice screen shots of power output but the 8 Ohm power output was nearly 23W at 1% THD and 4 Ohm power output was nearly 40W at 1% THD.


63VDC PS 2A 8R 1Watt:



63VDC PS 2A AC signal current through C4, C7, and 8R Speaker:



47VDC PS 2A 8R 1Watt:



47VDC PS 2A AC signal current through C4, C7, and 8R Speaker:





As mentioned I decided to build the 47VDC version. The chassis and power supply is from my BAF2015 Amplifier - A SIT Mu Follower Revisited With Feedback (https://www.diyaudio.com/community/...t-mu-follower-revisited-with-feedback.408193/). The power supply is CLC, comprising 2x22mF - Hammond 156B - 2x22mF, and an Antek AS-3218 transformer per channel for full dual mono construction.

I put together a PCB design and sent it off to JLCPCB for fabrication. In addition to stuffing the PCB once I received it, I also changed the CLC filter from V+ supply to V- supply.



Initial power-up and subsequent Iq and SIT Vds adjustments went smoothly, and Iq and SIT Vds were stable.

Next up was testing with an 1kHz signal and distortion measurements. Unfortunately when powered up with an AC line powered 1kHz oscillator connected, the amp immediately blew the powerline fuse. So it was trouble shooting time. After checking the schematic and pcb, checking that the input capacitor was not faulty, and not finding any visible ground shorts with the meter, I was baffled.

I knew that the issue was mostly likely a grounding problem and that it only manifested itelf when an input device was connected to the amplifier. Luckily a thought came to me fairly quickly - when I changed the CLC filter from V+ to V- supply, I forgot to move the power supply connection to safety ground on the CLC filter board. The correction was made and success. Only one fuse was permanently damaged.

The dim bulb tester came in handy here as once the fuse blew, the dim bulb tester went in and I was able to probe around with power and not blow anything. In addition to using it during trouble shooting, I always use it for first power-up testing of power supplies during various stages of construction and first power-up testing of audio circuits.

As for the amplifier supply not shorting with no input device connected but shorting with an input device connected, I gave it some thought. The power supply was for V-, with V- from the supply connected as power to the amplifier circuit. V+ from the supply was connected to the amplifier circuit board as ground. The V- from the supply was ground in the previous amplifier but I forgot to change it so it was now incorrectly connected to chassis safety ground. With no input device connected to the amplifier circuit, the power to the amplifier board was floating. Although the V- was also connected to the chassis safety ground and to the powerline/IEC ground, there was no other direct connection to V+, so no short. The live from the powerline/IEC was connected to the transformer primary, which was isolated from the transformer secondary. Also the speaker and input jacks were isolated from the chassis. So with no complete direct connection of V- to V+, the current can only flow through the audio circuit. The chassis was connected to V- but there was no path for the current to flow from the chassis to power supply V+.

When the oscillator was connected to the amplifier input, the circuit was completed, current flowed, and the fuse blew. That was because the oscillator was AC powered and had a safety ground connection. So the oscillator line safety ground connected to the oscillator power supply ground, then connected to the oscillator signal ground, then connected to the amplifier signal ground, then connected to the amplifier V+. The oscillator safety ground is also connected to the amplifier safety ground through the line ground, and the amplifier safety ground is connected to amplifier V-. The net result was amplifier V+ and amplifier V- were connected together - a short circuit. There was a CL60 thermistor at the amplifier power supply ground connection. That would have limited the short circuit current: 47V/10R = 4.7A. Fuse was 2.5A slow blow.

I have swapped power supply polarity before but had always remembered to switch the safety ground connected, until this time.

So always be careful. Electricity can kill.

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 everyone,


I'd like to share my latest project — a fully custom digital audio processing platform built around an FPGA. The system accepts multiple digital audio inputs (I2S/SPDIF), applies jitter reduction and fixed-point filtering, and outputs a balanced signal through a discrete R-2R DAC.


The goal was to achieve low-jitter, low-distortion output using hardware-domain techniques, while maintaining maximum flexibility in digital audio processing.

---

🔧 Hardware Specs:​

• FPGA: Altera Cyclone IV EP4CE115
• Inputs: I2S (from XMOS, HDMI, STM32), SPDIF (optical & coax)
• Output: Balanced R-2R DAC (via 74LVC595 shift registers, isolated with SI8660)
• Clocks: External 22.5792 MHz & 24.576 MHz oscillators, dual PLLs with dynamic reconfiguration

---

🧠 Signal Processing Chain:​

• ✅ Sample rate detection: 44.1 kHz to 384 kHz (auto-detect)
• ✅ Bit depth detection: 16-bit / 24-bit
• ✅ I2S Reclocking: FIFO-based low-jitter BCLK generation
• ✅ Fixed-point FIR Filter: Linear phase, group-delay-aware, designed for steep roll-off and minimal ripple
• ✅ ASRC (Asynchronous Sample Rate Converter): Linear interpolation based, upsampling to fixed 192 kHz / 24-bit output

---

🔊 Analog Output Path:​

• DAC type: Discrete R-2R, balanced
• Output filter: Passive RC (887 Ω + 3.3 nF → fc ≈ 54 kHz), optimized via REW tests
• Preamp: OPA627, gain x2, input RC filter (1k + 470 pF)

---

🧪 Measurement Results (REW):​

• SNR: 81 dB
• ENOB: 12.6–12.8 bits
• THD+N: ~0.018% @ 1 kHz, -1 dBFS
• Tested Scenarios:
  
  • FIR only
  • Reclock only
  • FIR + Reclock
  • Raw (flat path)
• Interesting Findings:
  
  • THD increases at 44.1 kHz compared to 192 kHz due to increased jitter sensitivity
  • RCA measurements underreport performance due to lack of CMRR; balanced output performs better

---

🚧 Current & Future Work:​

• ASRC output working on (fixed 192 kHz)
• Next step: porting design to Xilinx Artix-7 (100T)
• Native DSD passthrough & PCM-to-DSD conversion under development
• DSD dac by upper PCB

---

🎶 Subjectively, the system sounds clean, fast, and airy — with a notably improved low-end and clarity after FIR+ASRC stages. THD harmonics are drastically reduced when reclocking and passive output filters are tuned properly.


📎 I'm later to share code, VHDL modules, schematics, and REW test plots if anyone’s interested. Feedback, suggestions, and collaboration are very welcome!


Best regards,
Erdal

Hello diyers,
I've been playing sub-less for few months but with time I realised I was really missing a clean first octave.
My boxes are not able to go very low, it's 2x 15OB350 per side in around 200 L.
They can go low with EQ of course, but they just sound better crossed around 35/30hz, lower is just noise.
Don't know if it's the drivers high fs/qts, the "still" too small boxes or maybe not strong enough, or the room placement, probably all combined.
I see them more as big mid-bass in fact.

I had in mind to upgrade the drivers at one point, for the nero-15sw800 that should better fit the needs, but even if they're not that expensive it's still four of them that I'd need. And more important I'm not 100% convinced it will be a big improvement due to the mains placement.

So before going there I have to try spread bass again, within following constrains:
1 - nice look and integration with furnitures
2 - high sensitivity
3 - critical damping if possible

#2 means big drivers, I still have my four 18sw450 from my previous dipole, why not using them?
#2 and 3 kind of go together, but against #1 for the size implied.
#3 is a way to finally decide for myself if an L/T box can really sound the same as a big natural one (and I believe not, but lets' test).

I started with what I had around and within the new room layout constrains, with a sealed um12-22 in maybe 100-120 liters (much bigger in real but with sand walls) right behind me.
Not bad at all but not enough juice, plus it's not easy to nicely integrate with the furnitures.
I couldn't try my old concrete ported one as I got rid of it few months ago (and for that I had to destroy it in place, too heavy to get out 😛). I don't regret, it stayed for years unused in the same spot, it was fun to build, but also just too bulky, and too hard to move around to test different spots.

So, I had to find a way for 2-3 subs, placed behind me, light enough to move around a bit, but big enough to not require an L/T, easy?
Thought about it few months ans was about to buy woods when… again my favorite/hated shop just launched a new serie: the rådmansö.
Beautiful pieces in my mind, I would have loved to start with these when I've built my mains, a lot of cool stuff could be built around them.
Went to see them in real, and realised even the big one could do it, this one, 159x64x46cm, 468L. :




Got one, some mdf, some valchromat and let's go.
Like previous stuff I've just built a box within a box, 18mm glued all around inside, 24mm for the back, half the baffle doubled in valchromat, bracing with scraps, kind of light. The good surprise is that the radmanso is full particle boards, no honeycomb core on this like on many others, so nearly 4cm thick particles + mdf at the end.
The baffle is removable so I could add more bracing later if needed (I already found a spot that may need it).
And I could try 1 or 2 drivers, even clamshell isobaric if the box is still not big enough 😛
It's around 330-350 L. I'm guessing. Yes I did some sims but, they're just sims…

I of course did few mistakes again, my very first move was one! While sanding the inside of each panels for better glueing, I messed up and sanded the wrong spot of the panels, the edge that would have been visible outside. So I couldn't get the recessed baffle like I initially wanted.
Damn' what an idiot. So the baffle is now flushed, I'm sure it would have looked much better with the recess (like the mains).
Maybe I'll add a fabric grill, I'll see, it's fun to see the big driver too.
Then the valchromat was finished with tung oil, for a nice dark contrast.
Some details are left for later like the right cable, neutrik connector, better driver screws, replace temp wheels once in place to decouple a bit etc.
Not sure I’ll use the original steel stand that comes with it, it’s sturdy but will add 20cm in height.




I quickly plugged one little fosi on it yesterday and it seems promising, efficient for sure.
I hope I can move it in the right place soon to measure it.
Meaning right below the diffusors behind me, where I now have a bunch of besta used for storage.

In all this I kind of try to find the same effect I had in my car years ago. I might repeat myself but it was so fun with 2x15'' at few cm right behind the head. Breathtaking was the right word, literally, and I'm not talking SPL, just air movement in your lungs even at low levels, like someone with a pump trying to empty your lungs at every beat!

Of course home environment is a total animal but if I can get 10% of this I'd be happy 😛
Will come back with measurements soon, if a success I will build another one, and maybe even 3 in total.
Cheers!

Hey Folks!

I want to build a small and lightweight speaker that I can take with me while traveling.
Yes, I know there are tons of products out there already, but I really enjoy building my own gear instead of just buying something off the shelf.
I'm also hoping to get better performance than a typical JBL Xtreme-style speaker.

I'm already quite experienced with electronics, so that's not a problem. However, I’d really appreciate some feedback from the diyAudio hive mind on my current plan.

My goals:​

• Small – max. 3 L internal volume
• Loud – ~100 dB SPL
• Lightweight – <2 kg
• Low-end extension – 50 Hz @ -3 dB

How do I plan to achieve this?​

• 3× Tang Band W3-2052S – 3" woofers
• 2× Dayton ND20F-4 – ¾" tweeters
• 1× Dayton DSA175-PR – 6.5" passive radiator
• Custom PCBwith:
  • ADAU1701 DSP
  • 3× TPA3126D2 (6 channels @ ~35 W each via boost converter)
  • Qualcomm Bluetooth module
  • USB-C PD input/output
• Custom 4s 21700 battery pack

The enclosure will be around 18×18×8–10 cm and made from carbon fiber to keep it lightweight. All components were chosen based on size and weight.
So far, WinISD suggests I should be able to get around 99 dB SPL @ 30 W without exceeding Xmax (for both the woofers and PR), and around 49 Hz @ -3 dB.

The real issue: Woofer/tweeter crossover​

The W3-2052S shows some signs of ??cone breakup?? around 750 Hz, according to Voice Coil Magazine measurements (available on Tang Band’s site).
However, distortion looks decent up to 2 kHz, so I’m wondering if I can push the woofer that high and just tame the response peak with the DSP?

Even so, the tweeter would still need to go quite deep. I'm also unsure whether the ND20F-4 is the right choice from a distortion standpoint.
HiFiCompass did some measurements: LINK
Looking at the HD response at 315 mm, -30 dB H2 at 2 kHz doesn’t inspire much confidence.

Mounting constraints and alternatives​

I’m planning on an 18×18 cm front baffle, with the three woofers arranged in a triangle. That leaves me with only around 50 mm of space for a tweeter.
I’m wondering if I should go for a 1" tweeter instead – but which one could still fit, offer lower distortion, and cross low enough?
I’d really prefer a neodymium-magnet tweeter to keep weight down. Any recommendations?
I've also considered using a compact full-range driver as a mid+tweeter, but couldn’t find anything that fits the size constraints, has enough sensitivity, and uses a neodymium motor.

Some preliminary renderings of the planed enclosure:

So… what do you think?​

• Will such a low crossover work?
• Will my ears bleed from the tweeter distortion?
• Is this whole setup a bad idea?
• Any better tweeter or full-range suggestions?
• Any issues with the three woofers? (Cancelation?)

I planned to drive the left and right Woofer and Tweeter with the corosponding input signal and L+R sum for the center woofer, I could and maybe should use another DAC to drive it seperatly?


Thanks Folks!
Have a nice day
Farbe

EDIT:
I just found the SB Acoustics SB14ST-C000-4, its smaller with just 0,6" but seems to show better distortion (again HifiCompass). However sensitivity is also 3db down, not sure about this one.

EDIT2:
I might also be able to fit a Dayton Audio ND25FA-4 1" by milling two screw holes off. Not sure how this would change the response. I also found no distortion measurments.

Yeah so i bought a sub on a whim. Its a skar vxf 15 d4.
T/S Parameters
Bl 39.1 n/a
Fs 35.4 Hz
Re Dual 4 Ω
Sd 779.3 cm2
Cms 0.05 mm/N
Mms 384.3 gr
Qes 0.43
Qms 4.78
Qts 0.40
Vas 47.7 L
Nref 0.47%
Xmax 30 mm (one-way)
Sensitvity
88.7 dB
I plan on attempting to build an hroar box for it. So far i have modeled it in a roar box with very nice response from 26hz-80hz. I havent figured out how to translate bp8s function in hornresp to hroar, but i have a good idea what will work without modeling it. Ill post a hornresp response later today. I really have no particular need for this, i just want to build it and see what it does. Ill probably try it in my ht but inevitably it needs to be in an suv or something, so ill probably sell it haha. What would you do with it?

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!

In the last few days off, I have been somewhat confused and have tried to collect enough reasonable data for a possible replacement circuit using the usual home remedies. I don't know whether I succeeded or whether I was able to develop a common thread; the more diagrams one plot, the more confused the scenario becomes. I deliberately left out the interesting mechanical side, because how can one approach it reasonably well with home remedies? Of course one can think about the elastic side between the cantilever and tip and the plastic disk, but on average we always come to a conclusion, reso fo is always less than 35kHz and the cutoff fc moves from 61kHz to 32kHz ... outside to inside.

Perhaps we can develop a test procedure, a common thread, together at this point.

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

I have a pair of brand new in the box B&C DCX50 2" coax horn drivers for sale. Boxes were only opened to inspect them, otherwise brand new. Need to free up some space for new projects. Price is $850 shipped to lower 48 US states

This Group Buy relates to this thread:

This is an interesting list, and I’ll keep updating it whenever someone expresses interest. Once we have 10 people, I’ll produce a batch.
1, Ricardxgf,
2, Extreme_Boky,

Here is the info for the 45 amplifer from Sunvalley - SV-S1645D​

The composition of the vacuum tubes:

12AT7 x1 or 12AX7​

12AU7 X 2 ​

45 X 2 ​

274B/5U4G (tube rectifier only) ​

Output transformer Hashimoto H20-7U.​

Choke and Power Transformer 115/230V custom made​

Made and Assembled by Sunvalley Japan with 4 x VCAP ODAM (0.1 X 2 + 0.22 X 2)​

Available in February 2024 - Only 25 units for the 1st production​

Price : $2,725 ( included development fee $250 per unit charged by Sunvalley)​

[


[URL='https://flic.kr/p/2oPzp7X']



[/URL]

WE expect it will take 8-9 months to complete the following processes. Please stay tuned....

1. Circuit approval - DONE
2. Prototype production - DONE
3. Prototype Evaluation - DONE
4. Prototype approval - DONE
5. Quote Approval - DONE
6. Mass production - WIP
7. Assembly _ Sept 2023
8. Characteristics Verification
9. Shipping jAN 2024
10. SCHEDULE ARRIVED Vancouver Feb 24, 2024

WITH THIS >

Male / Female
Up / Down
Left / Right
White / Black
On / Off
Yes / No
In / Out
High /Low
Day / Night

I can't help but think about Polarity.
Another thing about polarity is that it gave us ONES & ZEROS. That is the 'language' of a Computer.
A Silicon switch on or off.

Perhaps the miracle is that if it wasn't for the grains of sand we see on the beach, that is Si = Silicon, todays computers wouldn't exist.

so WHITH THAT ... what do you think?

This thread is intended to discuss the preparation and construction phase of building an F5 Turbo. A thread already exists for the circuit boards, so I'm guessing we will see a lot of activity here!

AB100 Amplifier - Micro Article by Nelson Pass

Since villagers armed with pitchforks and torches have begun camping
out my front door demanding a Class AB power amplifier, and

Whereas, I am inclined to demonstrate my goodwill in this manner and
do in fact have a 13 year old piece laying around unpublished,

I Hereby Present the AB100 Class AB Bipolar power amplifier.

Salient Specifications:

Rails: +/-50V or so

Gain: 36 dB (this was the request of the people who commissioned
this but did not want to pay for it)

Bandwidth: 5 Hz to 100 KHz (-3 dB)

Damping Factor: about 500

Maximum power: 100 Watts 8 ohms, 200 Watts 4 ohms

Slew Rate: 30 V/uS

Bias Current: about 0.08 A per channel

Following are several graphics. (Unfortunately I scanned them in the
distant past, so the quality is not up to my usual standards)

Let It Not Be Said: That I never did anything for you....

😎

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 hope you’re all doing well! I’m reaching out to this wonderful community because I’m in a bit of a pickle with my beloved Lavry DA11 DAC.
The headphone output has stopped working, and upon inspection, I found several burnt components (transistors, resistors, and capacitors) in the headphone section. Unfortunately, the markings on these components are now completely unreadable.
I’ve tried contacting Lavry support several times for component identification, but haven’t had any luck getting a response. As a former Lavry dealer in France (2007-2010), I really love this DAC and would hate to see it end up in a landfill over something potentially fixable.
What I’m looking for:
• If anyone owns a working Lavry DA11 and wouldn’t mind taking clear photos of the headphone circuit components
• Any repair technician who has worked on these units and might recognize the components
• Someone who might have schematics or component list for this section
I’ve attached photos of my unit showing the damaged parts. Any help in identifying these components would be greatly appreciated!
This community has always been amazingly helpful, and I’m grateful for any assistance you can provide.
Thank you in advance,

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.

By popular demand, here is a new thread devoted to PETTaLS flat panel speaker simulation software, developed by Dr. David Anderson. Flat panel speakers may also be referred to as bending wave speakers, or DML speakers.
Let's use this thread to discuss topics specific to the use of and improvement of this great new tool. Topics could include, but are not limited to:

• new release announcements
• questions about how to use it
• features you'd like to see
• possible bugs
• interpretation of results
• other....

The free version of the software is available here:

PETTaLSFree

Some videos about the software are here:

PETTaLS Videos

Eric

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:

Hi guys, as tittle said "Building BA-3 as analog output stage of noDac", well the noDac outputs 0,19Vrms so I was looking for something with a gain x10 to get near the standard 2Vrms, helping me to take the BA-3 option because I own original SK170BL, SJ74BL, K2013 & J313 from when I used Erno Borbely stuff, also it's not complicated to do it and adjust (very similar to Borbely), don't take to much room inside noDac box and is a lot of more easy and a lot of less money than make a DHT 4P1L with the same gain of course I don't expect the same SQ.

I read with attention NP BA-3 article, it seems with standard resistors values BA-3 fits nicely the x10 gain I'm looking for.

Attached NP BA-3 article to know if it's the latest or there is other new and also B.O.M. for a balanced unit that I will not do because all my system is SE.

I read Jfets have to be "matched if possible" at 10mA Idss, I have on hand:

SK170BL
9.48mA x 2

SJ74BL
9.02mA x 1
9.03mA x 1
or
10.11mA x 1
10.24mA x 1
What's best pair to use with the SK170BL?

I guess with these values P1 & P2 will remain 500R, right?

With the input voltage of +-25V can reach the x10 gain?

I will post pics of my work in progress.

TIA
Felipe

N.B. Big thanks to Jim aka 6L6 to give me a couple of PCB.

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.

Greetings.

I present the DIY FRONT END 2022.




EDIT: For current boards marked "V0R1" please find the changes and additional information in post #686 of this thread
https://www.diyaudio.com/community/threads/diy-front-end-2022.394339/page-35#post-7323986

I'm really looking forward to this -

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

Roederstein draloric 0.60w resistor

It uses the the bias network to do error correction. It is equivalent to CFP with MOSFET output, but it can output from rail to rail, with help of the bootstrapping.
The example below is +-14Vp into 8 Ohm with only +-15V supply voltage. You can see the distortion is low. Note that you could put opamp at the front and wrap NFB around it to reduce distortion farther.

Just got them delivered from the UK. Doing up a Lowther Acousta Enclosure for them pretty soon

Premium Drivers information sheet​

Standard specifications: Many misleading terms have been used to describe the paper preparation of the Lowther diaphragms, doping, age-conditioning, curing, varnishing, etc., leading to discussions & myths around old preparations, similar to the legend of Stradivari’s varnish. There is no magic formula. The standard process, used for many years, is to apply freshly prepared 'special pale' French polish to the made-up cone/voice coil sub-assembly prior to fitting to the motor. This soaks through the paper leaving pure shellac distributed through the paper fibres as the spirits dry. Sugar-like crystals can be seen on the surface of a freshly treated cone. A few months sees a uniform yellowing of the diaphragm due to the reaction of light & air with the shellac. The paper fibres are locked with the shellac to be stiff yet flexible; It is generally accepted that the driver performs optimally after this point, though the foam suspensions co-incidentally loosen up over this period, and this has more of an influence at the early 'break-in' period. Older drivers (pre-Thornton) mostly exhibit a dirty, patchy yellow appearance due to poorly prepared lower grade shellac, which does not cure well over time. Controlling the quality & preparation of the shellac is an all-round long-term improvement, but due to the cleaner appearance, some still think we do not shellac the cones. Standard spec. drivers will continue to be available as usual.

Premium Specifications: The main difference over the standard range is the treatment of the paper. In this process the paper is impregnated with our own in-house formular before cutting, forming & assembling. The 'curing' process is mostly complete by the time the unit is tested & packed, though improvements will still be noticed over the 'break-in' period. The new treatment reacts with the paper fibres differently to the shellac process, resulting in a more stable, consistent & dimensionally accurate cone. The impregnation formular, however, can only be made in small batches in our own workshop and will not scale up to industrial preparation. This limits production to the ‘premium-hand-made' scale,
with the inevitable expense involved. The premium diaphragms are also fitted with the new back centre, replacing the foam standard. The entire set of components is laser-cut, enabling greater control of consistency & accuracy over the die-cut standards. Laser cutting also allows us to make small detail modifications like rounding the points of the 'pips' that hold the voice coil to the inner cone, cut elliptical 'pressure-relief' holes to control the dual cone phase errors, adapt the turned edge of the inner cone to a more suitable paper weight & composition and re-design the assembly relationship of the inner / outer cone / back centre / voice coil - most of which
would have been adopted before, but not possible because of the limitations of the old die cutting. The result of all these adaptations is a diaphragm which excels in all areas, including appearance, with the Lowther logos, radial embossing, and consistency of paper surface & colour. The legendary hand-crafted double-wound speech coil and overall <>10.5g moving mass of the assembly are retained. Foam surround suspension is the same as standard.

Sound: Initially developed for use in the Edilia, testing in all driver models & cabinets showed a marked improvement in all areas. Diaphragms prepared straight-out-of-the box for our most critical 'test bed' - the Lowther-Hegeman with field coils - were received by everyone with amazement; described by various judges as 'game-changer', 'most significant positive step for Lowther in years', 'improvement in all areas of presentation of natural sound'. We are confident that the premium 'upgrade' easily justifies the enhanced price, and all customers are encouraged to hear the difference at our studio showroom. All of our top range handmade cabinets (TP2, Almira/Edilia, LHSR, Audiovector) will be fitted premium regardless of driver type. The premium diaphragms can be fitted to almost all existing Lowthers as an enhanced Re-cone, or LFL trade in.

Premium diaphragms: Key points - Changes / Improvements

Paper gsm weight & fibre type
Laser cutting & CAD design.
Lowther's own paper treatment impregnation process
Inner Cone improved, so:
Enhanced choice of phase plug /centre pole treatment
Aluminum or Silver voice coil
Radial embossing
Logos printed onto cones.
New back centre & assembly
Improvements in quality consistency, appearance, longevity.
Significant improvement in all aspects of sound presentation.​

Click to expand...