I'm curious if anyone who's built an Aikido has done any distortion measurements, especially harmonic distribution. Some circuit particulars, too, please.

I am interested in this driver but can't find the Thiel Small parameters for it. Has anyone here who has used it measured them .

I am looking for replacement TO220 transistors. Checked Mouser and DigiKey: NLA. Also checked alternatives 2SC2907/3163/3186: NLA. Both transistors are used in a Sony Audio power supply regulator circuit +/- 25V DC input, +/-15V DC output. 2SC2275 has transient failure. I guess I should replace the pair. Any recommendation? Many thanks!

I'm playing around with the Grok AI and so far its an absolute beast - I really like it and compared to Gemini and ChatGPT the answers are much more coherent. A little bit frightening maybe, but absolutely helpful.

Hi folks, it's been a while since I last posted. I have dabbled with horns and high efficiency for the best part of the last 20 years, bit of an addict you could say, but following a recent house move I have had to 'make do' with a pair of Castle Winchesters... Whilst I miss the air and openness of good compression drivers in the upper frequencies, I must say I am rather taken by how the Winchesters do bass, especially considering their very domestically acceptable size. With the right amp there is a lovely sense of articulation down below (and they go quite low). This has given me ideas for a DIY system using a similar driver layout, i.e. one front firing, one upward firing (as illustrated by the cross section below), but using bigger and more efficient drivers, say 12" or even 15". I have a pair of of 400Hz 2" Stereolab horns waiting in the wings, which I could use for the HF. Any thoughts on which drivers could be made to work in this context would be welcome, or for that matter any feedback from those who have embarked on similar projects 🙂

Audio Analyzer with AK5397/AK5394A and AK4490

For more than 3 years I have been working on and off (mostly off) designing a high-end Audio Analyzer. Parts of the design have been implemented as separate units. I have finally put the pieces together on one PCB (+ some small existing interface PCB's).

Some of the design goals were:

1. The Audio Analyzer should work as a sound card, enabling use with a wide selection of Audio Analyzer SW packages.
2. The performance should be somewhere between a good high-end sound card and an AP, R&S, SR etc.
3. The performance should approach the limits possible with the ADC and DAC used.
4. It should be possible to use it with any relatively modern PC. This is the reason for choosing a USB interface to the PC. SPDIF/Toslink was considered, but this makes it difficult to use portable PC's.
5. The PC should be electrically isolated from the Analyzer to avoid ground loops and noise.
6. No fan for cooling!

I have selected to use high-end converters from AKM. I have previously made a design based on AK5394A and AK4399. This design shows a very good performance, but it was not ideal in terms of flexibility, due to limitations in the input and output stages. The input stage was a relatively straight-forward design based on OPA1612 and OPA1632. The performance is fairly good, but there is no attenuator, so the input range is limited.
The output stage has a limited drive capability, since it is based on the OPA1632.

For this design I originally chose the newer AKM converters AK5397 for the ADC and AK4490 for the DAC. After testing the AK5397 I decided to go back to the AK5394A, since the distortion is much lower.
The input stages of the new design are based on a JFET/op-amp combo, with LSK389 and LME49990. This part of the design has been verified on a separate PCB, with attenuator and gain options.
The output stages are based on LME49990 buffered by LME49600.

Main features:
1. AKM AK5394A 24-bit Super High Performance Stereo ADC, 192 kHz sampling, 123 dB SNR (changed from AK5397)
2. AKM AK4490 32-bit Premium Stereo DAC, 192 kHz sampling, 123 dB SNR
3. Dual fully differential inputs and outputs on XLR connectors
4. Dual single ended outputs on BNC
5. AC and DC input switchable
6. 44.1, 48, 88.2, 96, 176.4 and 192 kHz sampling, selected by the Analyzer SW. Potentially 384 and 768 kHz sampling later on (TBD).
7. Full scale inputs from 100 mV (-20 dBV) to 100 Vrms (+40 dBV) in 10 dB steps.
8. Overvoltage protection on the inputs.
9. Full scale outputs up to 10 Vrms differentially (5 Vrms single ended). Can be attenuated to 1 Vrms and 100 mVrms full scale differential output.
10. Single ended output impedance of 50 ohm (changed from 25 ohm)
11. Single ended input impedance of 100 kohm
12. On-board low jitter crystal oscillators (NDK NZ2520SD).
13. Parallel control lines for relays, controlling the attenuators, gains and AC/DC settings.
14. Power supply is integrated on the board (except for the transformer).
15. USB interface is implemented using a completely isolated XMOS based design (changed from USBStreamer).
16. Expansion connector for future digital I/O
17. 20x25 cm PCB

The parallel control lines may be controlled from either manual switches, e.g. rotary switches for the level settings. Other possibilities are to control it with a microcontroller or from the PC via a USB to serial converter. Initially I plan to use manual switches. Update 161222: Settings can be controlled from the PC via the USB interface.

For the power supply a 2 x 15 V and 1 x 9 V transformer is needed. I plan to use an R-core transformer with 2x15 V and 2x9 V, 30 VA, which I have also used for the AK5394A/AK4399 design.
The transformer may be changed to a 2x15V or 2x18V ring core transformer with a slightly modified power supply design. Update 161222: A ring core transformer is used. Primary voltages selectable between 100 to 120V and 220V to 240V.

The isolation between the Audio Analyzer and PC is done with a Master Clock and Isolator, see:
Master clock and isolator for the MiniDSP USBStreamer
Update 161222: The isolation is now part of the XMOS based USB interface.

I have just received the PCB this morning. It will take some time before I will be able to start testing, since I plan to mount the PCB by hand. And I still need to get some of the parts. There are around 750 parts in the design.


I would like to thank Demian Martin (1audio) for his design inputs, reviews and encouragement.

Update April 8, 2017:
GB is now open. See:
GB for RTX6001 Audio Analyzer with AK5394A and AK4490
Update:
Group buy is closed.

Hardware modifications:
To make it easier to find the descriptions of the hardware modifications, I decided to make a list here.

Disassembly and assembly + overvoltage:
GB for RTX6001 Audio Analyzer with AK5394A and AK4490

The overvoltage modification was originally described here:
GB for RTX6001 Audio Analyzer with AK5394A and AK4490

Shield and stability:
DIY Audio Analyzer with AK5397/AK5394A and AK4490

*Moderator note: Second block diagram added at Member's request.*

Has anyone direct experience with a FirstWatt clone that has gone from a CRC supply to a CLC supply? I'm talking same amp.If so are there any sonic benefits to doing so? In addition it is a standard 1 toroid,dual secondary,dual FWB rectifiers.I can get a couple of Hammond 159Zl 2.5mH 10A chokes for about $40 each.I can see from simulations quite a reduction in ripple voltage but would there be sonic benefits from this relatively cheap mod?

Hi - I’m a new member from Leadville, Colorado, living at 10,000 feet of altitude. My favorite current hobby is multichannel audio of all flavors, which I particularly enjoy over headphones using a binaural processor (Smyth Realiser A16).

I’ve been working on two projects lately. The first is using an RPI5 + DAC8x + ADC8x (HiFiBerry hats) to convert the 6-channel analog output from my Surround Master V3 to HDMI (my AVR does not have analog inputs). This has been very successful, and I’ll probably go on to explore using CamillaDSP in addition. I’m also using it via Audacity as a replacement for my old Korg D888 8-channel analog recorder.

The second project is exploring ways of extracting HBR audio streams from HDMI. I have an old Unigraf frame grabber which works, but I need to write some code for it to work better. I have a Chinese HDMI input to I2S x4 board that should work but I haven’t had a chance of hooking up my logic analyzer to the 4 I2S outputs yet. I have a Rock Pi 5b that has the necessary hardware, but I need to wait for the mainstream kernel to support multichannel audio in from the HDMI RX (6.15-RC1 is just out and may do the trick; the current kernel driver only supports 2-channel). Finally, I have an HDMI extender over 10 gigE TCP/IP with a managed 10gigE switch that will allow WireShark or similar snooping.

Selling a pair of Faital Pro 8PR200 8ohm drivers for £100, which is 50% of the retail price (£200).

• Condition: Good, with only minor blemishes on the chassis foam surrounds.
• Usage: Used only in my front room at low levels, crossed over at 100Hz to subwoofers. Never pushed hard.
• Packaging: Original boxes are stored in the attic and will be retrieved upon sale.
• Details: An excellent driver that needs little introduction - manufacturer page here: https://faitalpro.com/en/products/LF_Loudspeakers/product_details/index.php?id=101030200

Postage for drivers is £7.00 in the UK

Optional: I can also include the cabinets for the cost of extra shipping (~£10). These are 10-litre enclosures with a 50mm PVC pipe port (friction fit), Neutrik Speakon connector, and Van Damme cable. Birch ply and MDF. Not perfect nick...

I have a Philips FB297 model 3-way speaker. I wanted to change the electronic capacitors in the crossover circuit, but the capacitor shown in the picture (blue) which I thought was 1uf showed 4.5uf in the measurement. I was surprised. Old blue capacitor with 1u0 written on it. What could be the real value of this capacitor?
Sorry for my English. Thanks for any help.

I'm putting this up as a place holder for the new nx2-Amplifier (hereinafter just called the 'nx2') which has been in gestation for some 15 months now. I've had quite a few nudges from various people via email asking about progress. I can understand that some folks have fallen into believing the nx2 is 'vapourware' (if you've worked in semiconductors or software product development, you'll now what that means!).

I started work on this in September 2023, assembled the initial proto and decided it needed more refinement - details which I will cover when the documentation is released. However, things came to a grinding halt when TraxMaker (part of the CircuitMaker CAD suite) stopped working in early 2024. I made the decision to switch to KiCAD (a good move!) and had to redesign the board from scratch. After another 2 proto/dev cycles, the amplifier module PCBs are finally where I want them to be, and quick bench testing shows peak mains hum harmonic components at below -125 dBr reference 60W RMS into 8 Ohms (the power limit of my test lash-up) and <35ppm distortion.

I am currently finalising the PSU +inrush PCB and that will go off for manufacture sometime next week. In parallel with this, the mechanical layout is also being finalised.

Part of the delay in this project has also been the fact that I became a grandfather (yeah, I'm old) for the second time and that entailed putting a lot of stuff on the backburner, and the decision to move closer to family, which we will do in the next 3 weeks. Its chaos here with boxes all over the place, packing etc.

For all of the above reasons, the nx2 won't be completed and released for at least another 2 months, but hopefully you wiol get an idea of whats n the pipeline from this intro. The boards will be available on the hifisonix web shop if you want to build one (but only after I release the completed amp + documentation).

Here are the specs (note these may change once the completed, housed amplifier is re-measured)



I've attached some photos below - taken in my workshop today in amongst the packing boxes. (note the front plate is still being assembled so looks a bit manky - it still has to be cleaned, screws fitted and then polished)

"Hello, as already mentioned in my introduction, I am inexperienced but use the soldering iron for basic tasks, like soldering RCA connectors, etc. Well, I bought a used Tricord Dino phono preamp to connect it to my Rega Planar 2 turntable. As you know, Rega turntables share the arm ground with the left channel. This turntable uses a very cheap direct cable from the arm to the phono preamp. I wanted to modify this cable by cutting it and mounting a metal box under the turntable with female RCA terminals so I could choose the cables based on construction or length. Now, when I turn up the volume without music, there is a hum coming from the speakers. If I touch the box, the hum increases, and it also increases if I touch the arm. The RCA connectors do not come into contact with the metal box. I think I’ve created a ground loop; how can I fix it? Should I connect the left and right negatives to the metal box? Should I connect the metal box to the negative terminal of the phono preamp?

Building the Pass/Firstwatt F4

This is a fantastically good sounding amp - read more about it here before staring;

https://www.firstwatt.com/wp-content/uploads/2023/08/prod_f4_man.pdf

The thread at DIY audio -

http://www.diyaudio.com/forums/pass-labs/97540-f4-power-amplifier.html

And this the the corrected schematic (The schematic in the Firstwatt article has a typo, also this one agrees with the PCB)



Here you will find a build guide for the Pass / Firstwatt F4 power amplifier using PCBs and chassis from the DIYaudio store.

The 5U 'BIG Amp Chassis' is shown, because that's the one I have. It will fit comfortably in a 4U 'Jack of all chassis' and have enough heatsink as well.

~~~~~

There are plenty of places that you could start, but for the sake of illustration let's begin with the heatsink assemblies -



This is the heatsink(s) from the 5U 'BIG Amp Chassis' It has a mirror-imaged set of pre-drilled heatsinks and brackets to hole them together and make a mounting point for the rest of the enclosure. The 4U is similar, but the heatsink is a single piece.




This build will also utilize the 'DIY friendly' baseplate, here shown with the feet and hardware, and also the heatsink's brackets.


There is a hardware package available for the pre-drilled back and heatsinks, including input and output jacks, IEC module, and hardware for the PCB and heatsinks.


The contents of the hardware bag.


Using the brass PCB standoffs, install them into the PCB mount holes as shown, to get the following pattern;


Now there is a place to mount the amplifier PCB


Speaking of PCB, it's a very nice layout, plenty of room, and the ability to use many sizes of resistors and caps. This is the front.


Here is the back.


Stuffing the PCB should be done in the usual order, from smallest device to biggest - so that would be diodes and resistors first.

Of note, I got the bigger (Dale/Vishay RN60) resistors to see how they would fit on the PCB. They are the size of the PRP resistors that are quite popular amongst the fancy parts crowd. They are great everywhere except the row flanking the small transistors right in the middle. They don't fit there side by side. You could mount them soldier style, or just mix in a few smaller resistors like I did. Or just get RN55's. They are the smaller size.


Pots and transistors next. (yes, I didn't stuff the input pair when the photo was taken…)


Ah, there they are.
The ziptie is just to help their thermal tracking.


And finally the capacitors.


There is a top and bottom to the Universal Mounting Spec holes, the board mounts as shown


I find it helpful to bend the leads of the transistors first, and mount them (a little bit loose) to the heatsink.


Like this


And then mount the PCB. You can snug all the screws down and then solder and trim.

Ok, now lets move on to the Power Supply.


Here is a photo of the PSU board, I am going to use integrated bridge rectifier blocks, so you need to remove the part of the PCB that mounts the diodes. The new PCB, not quite yet available at the time of this writing, will have a similar feature with the diodes, as well as room for more/bigger capacitors.


As always, stuff the small components first - the light blue resistors are the filter resistors, the darker ones with the teflon are the bleeder resistors, and the small ones are for the LEDs.

This PSU board is the exact same DIYaudio PSU board, just without the top blue soldermask.


This shows the INPUT edge (from the diode bridges)

The capacitors are Panasonic T-UP 33,000uf 35V


Connecting the bridges to the PCB




This is the OUTPUT edge of the PSU - the colors are
Red V+
White GND
Green V-
The black connects the PSU GND to the CL-60 to the chassis.


The wiring from the PSU to the amp PCB is clearly shown.


Here you can see the bridges with the wires attached from the transformer secondary. Remember that the green attached to a bridge must have continuity with the blue attached to the same bridge. (As it's the 2 ends of the same piece of wire)



As long as were are tailing about the transformer, here is a photo of the terminal block shown wired for 120v. The Blur lead is the AC Live, and the clear the AC neutral. The reds and blacks are the transformer primaries.

Transformer is an Antek 400VA 18v+18v, part number AN-4218.



The last bit of the PSU wiring is the chassis connection, the black comes from the PSU GND, and the green is the AC safety earth.

The AC to primary wiring confuses everybody, so;

Let's look at the PSU schematic just to make sure everything is OK... Remember that I am wiring it for 120v operation, so the transformer primaries are in parallel. People wiring for 240 with a transformer like this, please ignore.



Notes in red are mine.

Look at the connections of the transformer primary, through the thermistors and line cap, to the mains.

Hot AC is connected to the "120" (which in my case is the red leads on the primaries) One red primary is connected to AC hot through a thermistor.

Neutral AC is connected to the black "0" leads, one of which is connected to the AC through a thermistor.

AC Hot and Neutral have a cap across the leads.


So, yes, the AC will be connected to the center 2 posts, which is across the cap.


(This photo lifted from my F5 thread, but it's the same PSU…)

Left to right we have post 1, 2, 3, 4

POST 1 - Transformer primary 'B 0' which will be connected to AC Neutral at post 2, through the thermistor between post 1 and 2.

POST 2 - AC Neutral in (not shown in photo), connected to Transformer primary 'A 0" , a thermistor to post 1, and a line cap to post 3

POST 3 - AC Hot in, connected to transformer primary "B 120", thermistor to post 4, and line cap to post 2

POST 4 - Transformer primary "A 120", connected to AC Hot through the thermistor to post 3

If you look at the red and black wires in the photo you will see that the Mains AC must to pass through a thermistor to connect to each of the 2 primaries. And that is the point of them, to keep inrush under control during powerup.

Ok!

Now we need to put everything together -

But first a bit more mechanical assembly.


This is the pre-cut back plate and the thick front plate.


Gather and mount the IEC plug.




The speaker posts.




And the RCA jacks.

Note that the shoulder washer goes on the inside, so the metal of the chassis doesn't touch the metal of the jack. There is a similar washer on the speaker posts.




The inside of the back panel.


And the outside. Looks good, yes?


The IEC module is wired as shown. This will switch both the Live and Neutral. The blue (live) and clear (neutral) go the the wiring block with the thermistors, cap and transformer primaries.


The amp PCB completely wired.


A bit closer.


The top connections labeled.


This happens to be the other channel, but the connections are all the same.

Remember that V- / GND / V+ is always left to right as you are looking at the format of the PCB


I'm not entirely sure what I was trying to show here, other than the screw and washer. It looks cool. I will keep the photo in the guide.





A few notes on bias -
P1 controls the bias, measured across any of the 3W source resistors. Adjust for 0.13v when it's cold, and watch that it doesn't get higher than 0.2v once it's up to temperature in about an hour. Adjust for 0.2v when hot.

P2 is used to adjust the DC offset on the output to zero.


Attach a DC voltmeter across the speaker outputs to measure offset.




If you find that the P1 doesn't have enough range, I.E., you can't turn it up enough, replace R9 with a smaller resistor, I used 4.75K and it works well.



Connect a voltmeter across any of the source resistors. The outboard ones are easier to clip across.

Adjust for about 0.13v cold, and once the amp is up to operating temperature, trim for 0.20v - It takes a long time to warm up, take your time.

Adjust P2 for zero offset, then re-trim P1

Here is a photo of it all connected and working -



I'm driving the F4 with an O2 Headphone amp sourced from an iPod; Driving 85.5db speakers. Although it is a small room, it gets louder than I want to listen. It still can't drive it to clipping, but it does get really, really loud.

One thing worth mentioning, and it speaks very highly to the quality of the amp, is that it is completely non-fatiguing, and more interestingly, very easy to listen to turned up too loud… I don't realize how loud it actually is sometimes. Complete transparency is a word used a lot when describing this amp - but I have to agree. It's fantastic!





Please comment away if you desire.

Also please feel free to ask any F4 questions here, and if you would like to post photos of your F4 completions, old or new, please do!

In the same vein as 6L6's MK2 thread, I thought we should start one for a related, formidable, Technics beast.

There's not a lot of information out there on these, even if you can read Japanese, or want to try your luck at deciphering Google Translate... Sometimes I think reading Japanese would be easier.

Should any pertinent links or information come up, I'll try to document them in this first post. Skip down to get to the good stuff. 😀

-------------------------------------------------

I thought it would be a great idea to open a thread on proper ways to mount transformers/chokes/Output transformers.

I cannot find my copy of a great book on the subject.

I was looking for the do / not do of mounting flat chokes on a chassis. I could not find anything on the subject.

1. Rule : never place metal across laminations

I am new to XSim and still learning. This is and all Dayton audio 3-way design with the following speakers.

DC28F-8 tweeter
RS180P-8 ohm paper driver for the midrange
RS180P-4 ohm for the for the woofer. (It looks like the impedance is drop to 3 ohm around 130 Hz.)??

Just a note on my amp. I am using the Outlaw seven channel A/B amp which is rated to 130 watts in 8 ohms and 350 watts into 4 ohm all channels driven.

Enclosure info.

3/4" mdf material
Midrange has it's on sealed compartment
Woofer section 1 Cu/ft. ported to 35 hz.

Can someone take a look at the attached to see if I am on the right track or totally off base. Any recommendations would be appreciated??

What is the benefit with Open Baffle?

So I've got a 3 way build about to begin... And my midrange is a JBL 2251 10" mid and the plan was to use what I have on hand... That includes either a pair of Selenium D220ti or Celestion CDX1-1745 drivers and these 2 waveguides/horns:

https://usspeaker.com/dds cfd-1-90pro-1.htm

Or

https://www.parts-express.com/B-52-PHRN-1014-10-x-14-Plastic-Horn-Waveguide-299-2303?quantity=1

I'm guessing I want to keep the width of the waveguide closest to the woofer, and the one has a lower C2C distance. As I've not figured out anything other than my midwoofer and bass woofer, I was curious if there was a set of drivers and waveguides that stood out as the obvious choice to start with.

I am starting active and want to make a passive crossover for the MT section, and will be taking measures etc. Just wondering where the best jumping off point is. Thanks

I currently am running a pair of k402MEHs as a 3 way with a pair of f20 subs. I just picked up a Yamaha sp2060 and it offers a 2 way + sub configuration and a 3 way configuration. I like having the full range for my stereo setup (my MEHs will dom20hz to 20k, but I am pushing the M3H 15"Celestion woofers too hard..hence using as a 3 way now).
What is the difference, and what would the advantages and disadvantages be. I am using a Denafrips Athena 2 channel preamp for my stereo setup and feed my Marantz av8805a into one input on the Athena for movies.

This is the module:
https://www.tianjiarun.com/post/125.html?language=2
My aim is to have a BT receiver that supports "advanced" codecs and that can output a I2S stream.
The QCC3083 is not directly available to consumers.
This module seems a reasonable solution (~ €10 on aliexpress), but the documentation is just an enumeration of pins...

Hello. I'm simulating Frank's 6sn7 preamp and have a question about output 5uf capacitor. If I leave it with 5uf than the operation will stabilize around 30 sec (please consider red and purple waves in the plot) . If I use 0.047uf instead the amplifier will work from start. What's the use of such capacity?

I built the attached amplifier circuit using OPT’s with the specs indicated in the 2nd attachment, I paid pretty good money for these. My system is located in my small office which is 11’ x 12’ x 8’. Finally got around to running test-tones and saw that I’m not getting anything below 50hz and the low end starts rolling off at 100hz with my speakers. I can fill in the bottom end with my subwoofer. Just for grins I hooked up my mobile amp, a little Dayton T-amp which cost me all of $35. Without my subwoofer the low end is not missing with this little amp. Can anyone point me in the right direction on what could be causing the low end rolloff with my 2A3 amp?

Well not match but to be closer to the mid. Since the Volti Vittora is 105dB efficient with the bass horn. I thought integrating the mid bass with the mid would be somehow better with mid bass being 103dB instead of 100dB.

Situation:

Listening area is 45m2 (floorarea)

I'm planning an open baffle mid bass with 15pr400 (Got it years ago for a good price so not looking at any other drivers) (120Hz-500Hz)

Bass will be done with a closed sub I already have with BMS 18n850.

Bass and Sub Bass will be powered and 'crossovered' with miniDSP PWR-ICE250. ( <120Hz) one per side so. 4 channels total) So I'm assuming at the levels I play the amp can handle the 4 Ohm mid bass easily. (should I choose to go 2 mid bass per channel)

Mid 500Hz-6kHz=BMS 4592 mid on JBL 2380a clones.
High 6kHz =faitalpro 10hfak on sth100.

Crossover will use Crites 3636 to bring the mid and hf levels down. So mid and high with SE tube amps and as you can see a passive crossover.
Mid Bass open baffle because I baulk at building a box with current prices of birch ply and my poor woodworking skills. And Troels raves about the bass on his OBL15.

Question: should I got for an extra pair of 8 Ohm 15pr400 paralleled for an extra 3dB? Would be another 600 euro ex shipping. So what would that extra 3dB bring to the whole setup? In terms of subjective added value.

This is an end game speaker for me.

For calculating internal cabinet volumes, how do you factor in cabinet fillers? For solid rubber matting or something dense like XPS foam, it seems reasonable to just subtract the volume used from the total volume. But what about polyfill or denim matting? Maybe for each cubic foot of polyfil, subtract something like 0.5 cf? Then a little more for denim, like -0.6 or -0.7? Some of the air in the polyfill has to contribute to the overall volume, right? Thanks for the help in advance! -Tony

I built a 12 inch 3 way system using SB34NRXL-08, AC 130F1 and Seas 27TAX/BG. The baffle is not stepped. Attached is the impulse response. First of all how bad is it? How/what can I do to improve it?

Well here's some info on my newest project. (Skhorn is pronounced like the word scorn.)

In short this's something I've been working on for years to replace my Othorn cabs. The design goals going in were pretty much the same. PA or very high level music playback in large or outdoor spaces. The main areas that I wanted to improve over the Othorn are as below.

#1 Cut down on cabinet and panel vibration near the driver.
#2 Improve the fold or reduce build complexity.
#3 Higher overall performance.
#4 More driver options and some less expensive ones.
#5 Increase the “FUN” factor even further

After simulations of what seemed like every cab type and most of the commonly available drivers, I ended up ditching the 8th order BP I had, which looked really good on paper, but was an absolute nightmare to turn from a sim into a buildable cab. Not to mention the extreme sensitivity to mistuning, etc... After that a TH was going to be the follow up and I had it all drawn up but I just wasn't satisfied with it as it seemed like an incremental step rather than something new. Finally I settled on the cab type also used on the M.A.U.L.. It was basically a parallel effort on both designs. The big one just happened to get built and tested first.

The Skhorn is a 24x32x54" cab that is comparable in size to the low tuned dual 18 vented cabs on the market. It's 24cu ft or 680L external volume. Originally I wanted to keep the size the same as the Othorn which is 18cu ft external but using 2 large drivers simply required a bit more volume and a different shape. The Skhorn is exactly 33.33% bigger than the Othorn, so 3 Skhorn are the same volume of space as 4 Othorn cabs.

I settled on a quasi 6th order BP, where the low end is vented and the upper end of the loading is a short, expanding slot/horn, with an opposed pair of 15-21" drivers, in an OD arrangement. I ended up at this while trying to work on translating a standard 6th order sim into wood and screws. I was having trouble getting the upper bass vents to fit into the cabinet and this caused me to look at how a straight slot performed instead since it is MUCH simpler to fit into the cab design. From there it was natural to model the slot with both a positive and negative taper and see what happened. A positive flare seemed to offer some improvements over a simple vent and with tweaking of the length, area and expansion, I found that this type of loading for the drivers could produce a smooth response over a wide bandwidth while also exhibiting some gains over a standard vented system using the same driver.

EDIT: 02/20/18
I have improved the accuracy of HR simulation inputs when compared to the measured ground-plane performance of the final speaker. The HR inputs below are not nearly as accurate as those developed after testing the final cab.

Here’s the cab design.

Greetings everyone I am in need of some NEC 2SB795-L & 2SD986-L These are Darlington Transistors if anyone has any for sale please let me know.

Regards

The Yaoin Tube buffer...:

Arne K

Hello!
I wonder if someone have bought this item or similar?

I just found this on AliExpress:
5 631,87kr | TransAudio D5PRO Clone Dartzeel NHB-108 Hi-End 300W HiFi Stereo 2.0 Channel D5Pro BC capacitor version Power Amplifier
https://a.aliexpress.com/_Eu2kFpq

Hi *,
I bought the LC30 two weeks ago. In my eyes mechanical and optical a very well done device but for me it has two issues:

1. It does not self power on if 5V comes up via USB. Would like to see a jumper inside where this could be configured.
2. The display is uncalibrated. Nearly no movement at low levels and the meter swings much too fast.

I decided to develop an add on prototype PCB which I mounted into the box. There is much space inside so my additional prototype PCB nicely fits in there. It uses two AD8310 devices (used it as I never worked with it before and was curious) to create a real logarithmic output. Also I added some other components to be able to exactly calibrate the meters.

As I‘m not able to solder the tiny AD8310 I bought two of these PCBs, and mounted them on top of my breadboard. I had to modify them a bit to work with audio frequencies but this was quite straight forward.

With two potentiometers I’m now able to calibrate the 0dB and the -40dB levels at 1000Hz. After calibration the levels in between also fit. Only drawback is that the scale of the meters above 0dB is not logarithmic. So the values shown above 0dB are wrong. A damping resistor across the meter establishes critical damping so I get only a very small overshoot from pulses.
I‘m very happy now with this solution 🙂

I also integrated a NE555 and a small DIL Relais that simulates pressing the power button after 5V from USB comes up.

If someone is interested I offer to draw the schematics and publish it here. I also made some fotos during my work that I’m also willing to share. Maybe someone with skills in PCB layouting and producing will kick in and build a circuit board equipped with components so others also can upgrade their unit.

I uploaded the schematics and Fotos of my implementation in the Audiosciencereview thread of the LC30. Please follow the link for more information...
https://audiosciencereview.com/foru...speaker-amp-switcher-meter.57415/post-2270108

Best DrCWO

Hello guys.
Im from Sweden and i saw an older post of someone seraching for the "Audio Generator GAG-808D", and i have one at home if he still is interested, i especially created a account to reach out to him.

So i hope to get help to reach out to him, i tried to reply to his older post but i somehow wasnt allowed to post since i havnt "reach enough credit points" or something?

Hope you guys can help me/him out.

Cheers!

Hello, I am retired and live in Green Valley, Arizona.

Since I'm plugged into the latest GB on the Wolverine amp, I'll be putting in a production pcb from the list below...

PSU Board (2oz blue)


Small heatsinks SK 75 37,5 STS - price on request


MBR40250G - When ordering 100 pcs, the price on the mouser is already acceptable 😉

SSR protect (2oz blue)


Dual Ground Lift (2oz blue)

Bal/Unbal input

Support psu for bal/unbal input
...in proggress

If anyone would like, just write me a private message. The deadline will be 4.5.2025 and the boards will be ordered.

Hi everyone, hope all is well. This is the latest design that I came up with for a Low mu pre-amp with autoformer. Very simple design with a quiet power supply. The pre-amp is intended to drive a 10K impedance. Input will be a 2Vrms digital streamer. I can't find much user experience on the BH rectifier, other than the spec sheet (attached). Would like your thoughts on this design.

klh asw10-125c
Ports are annoying create too much noise . This annoying and unwanted noise is a pain.
Plus my line level inputs are broken .😞 I'll get a pic layer of them.
Amp and driver work fine. Except auto off which never worked .

Hi guys.
Today I'm messing around with a Yamaha B100-115ii bass amp from the '80s. Actually it's only the power amp, the preamp does not exist anymore.
The heatsink on this amp gets very hot, 84 - 88 degree Celsius, you can fry eggs on it...
My question is: is this normal?

Yeah I financed this with my bank
I was blown away after the first spray

Tim Stinson here. Owner of Luminous Audio. Just hoping to share ideas and perhaps obtain some knowledge from others on designs repairs.

Just uploaded to AudioMack an album of the best mellow tunes from my Collaborations albums.

https://audiomack.com/hkurtrichter/album/mellow-collaborations

Hopefully this is in the right area.

I have a Cocktail Audio X30 that doesn’t want to power up fully.

When the power button on the front is pressed, the LED briefly flickers green and the relay can be heard to click, but then it instantly powers down.

I had hoped that it was just a fault with the relay, it’s a Handouk HE-1A-DC, but when I have removed it and tested it it seems to operate as intended.

Does anyone have any experience with these?

Any help and all suggestions appreciated.

Hi,

continuing from this thread I would like to open a new thread about interfacing a TDA1541 based DAC to a PMD100 based frontend, but the information about this topic seems to be lost. Here are the links I found:

• Nakamichi glue logic
• philpoole's information
• guido's solution
• some useful diagrams about how to interface the TDA1541A

I also found some information on hifidiy:

• SM5813 to TDA1541
• SM5813 to TDA1541 follow up
• PMD100 to TDA1541A
• PMD100 to TDA1541A follow up 1
• PMD100 to TDA1541A follow up 2

I simulated the mentioned circuits:

• Nakamichi glue logic
• SM5813 glue logic
• PMD100 glue logic

Attached screen shots of the simulations where the timing of the created LE signal is shown.

EDIT: simulation of PMD100 glue logic for stand alone mode. Post #102 gives a summary of the reclocked glue logic circuit I installed into my Arcam Delta Black Box v.1, post #228 contains the final simulation, #229 the latest mcu code to configure the PMD100 in programmed mode and the latest schematics, #230 shows the final state of my Arcam.

Hello all - I hope your projects are going well.

Today I got a CA Azur 840a for a very reasonable price. The seller claimed input one was dead on both the RCA's and XLR's. Reading up on this amp - I discovered the inputs are controlled by relays - and my suspicion was that the input one relay had gotten sticky. Not so - I can hear it activate robustly. Turns out the wee toggle switch on the rear panel that selects either the RCA or XLR inputs has been broken and is flopping about.

I found the service manual and I cannot, for the life of me, find mention of this switch anywhere in it.

In my initial listening - I like the sound of this amp - and I'd like to get it working completely. All the other inputs operate fine.

I'm happy to either replace the switch - but would love to discover the part # so I can order it before taking the amp apart - or simply rewire the circuit so that input one is hardwired to the XLR inputs. I don't often sell gear and this will likely stay in my collection for a good while. So getting it back to factory spec isn't important to me.

But I can't discover in the schematics or the BOM what's needed for either approach.

Any tips?

Link to the service manual is here: https://elektrotanya.com/cambridge-audio_azur-840a_ver.2_sm.pdf/download.html

Thanks in advance!
RK

It's nice to present this project - separate delta sigma modulator for a dsc2 dac.
Special thanks to Jussi and Pavel.

The basic idea is to convert pcm to a dsd stream without the use of a PC.

Design flow goes as follows:

i2s serial data is converted to parallel, next we have the first interpolation fir filter,
in this filter the data is oversampled x2, x4 or x8 - this depends on the incoming data rate.
(for 44.1kHz we have x8 interpolation, for 96kHz x4 and for 192kHz x2)

next, the data is going to the two stage CIC filter and is oversampled x32 to final x256

Then the data is going into the sigma delta 7 order modulator, which has classical CRFB structure,
as I said earlier all arithmetics are done in CSD arith, so no multipliers here.
all coefficients are carefully optimized using delsig package
If someone asks, yes, there are g0, g1, g2 coeffs.

At the end the dsd streams are time aligned using ODDR2 primitives

XC6SLX9 is almost full, so there is no ability to add anything 🙂

The device was designed to run with a mclk of 22/24 MHz.

To upload the attached bitstream file to the board you need a Xilinx programmer cable and Xilinx program Impact.

In the next post my friend PJotr will post gerber files, BOM and some measurements.

I hope this project will be interesting for someone.

DIY use only!

--------------------------------------------------------------------------------------------------------------------------------------------------------
Update 24.07.2023 New firmware !

First of all, I'd like to thank Markw4 for your help!

The entire software is practically completely new.

The first FIR filter now has x8 fixed interpolation (fs 44.1kHz - 192kHz), has 2048 taps, 32 bit coefficients and 100dB stopband attenuation.
Then it's a two-stage CIC filter and it's oversampled to the final x256. Oversampling depends on the rate of incoming data.
Then we have a 5-order sigma delta modulator, which has classical CRFB structure, with noise shaping and dithering.
And as before, at the end the dsd streams are time aligned using ODDR2 primitives.

We made tens of different firmware configurations and finally chose this version after listening sessions.

Latest firmware:
https://www.diyaudio.com/community/threads/simple-dsd-modulator-for-dsc2.370177/page-42#post-7407495

I'm rebuilding a Classe CA-401 after one channel blew out. I'm documenting the entire experience to keep track of everything I've done and to help out others. Enjoy!

Every power transistor in the channel was blown. Here's a picture of the vaporization leftovers of the 10N20 and 10P20 gate resistors. Also, check out the 383R VBE multiplier resistor residue in the top right corner:

Hey, my name is Mike Flathers, I'm from Lake Stevens, WA (Seattle area). I do a lot of DIY, professionally I am a creative pastor and spend a lot of time in AV, building and maintaining PA systems. I love keeping old quality gear alive!

evotronix KARPEN

Good evening everyone,
my name is Fabio and I'm writing from South of Italy - signed up aware of my ignorance in DYI as well as that there is always time to learn if you have a pinch of good will!

This request is for a friend of mine and it's for the DAC module of Twisted Pear 9028 DAC, preferably from someone in EU.
There was one sold here not long ago but we missed that and hoping that someone else also has it in his closet.

Howdy... I'm a regular on SBAF, but a newbie here. I've built some bottlehead kits in the past, have also played around with some speaker kits. I want to delve a little bit deeper into speakers and figure this is a great place to get more info.

I am a relatively new tube amp DIYer. Built two spud amps, a 6BM8, and a 6LU8.

Now, I'd like to try a single ended EL84. I understand that there are at least three output modes, pentode, triode strapped, and ultralinear. I haveno interest in triode mode.

My understanding is that pentode mode would have highest power, and highest distortion. That is mitigated through global negative feedback which reduces distortion, but also decreases output. Ultralinear, reduces distortion,likely better than gnf, but doesn't it also reduce output power?

My goal is to drive the el84s to getthe highest listenable output, basically, headroom. I normallylistento my 6lu8 at a modest volume, and likely similar volume for the el84s. My quest would be not to have the lowest distortion at max output, but low(ish) distortion at listening level, butwiththe mostheadroom I can get.

Now, I know gnf is typically defined in dB negative feedback, but I do not have the knowledge to understand the relative significance of say 15db at a 3 watt output. Especially compared to say any power sacrifice going UL.

There are ma y schematics outthere using both either UL. or gnf.

Help me understand the benefits of each approach.

Tia,
Roger

Hi all,
my NAD c326bee has serious problem. It keeps turning on and the buttons on front of the panel are irresponsive and also the standby button. The only mode to turn the amp off is to use the hardware switch on the back panel. When the amp is in this state (power on but irresponsive), it becomes quite hot. The problem was fixed in april with a quite expensive repair, now the problem returns.
Is there a way to fix this weird malfunction?

Thanks

I have for sale a new, assembled and tested module with MJ’s cap mx and active bridge rectifier.
Pcb is done by Analog Design.

Sold

Hello all,

My name is Neeraj, I am from Maharashtra, India. I am 38, but new to the world of DIY Audio and am not from a technical background (I was a commerce student) but I am very much interested in audio.
Some years back, when I came to know that the expensive audio equipments can be made at home, I started studying from the very very basics of electronics and started learning etiching pcb at home.

Till now I have eatched PCBs and assembled Amp Boards like couple of designs for TDA7294, Couple for designes for TDA2009, TDA2030. Some transistor amps based on the design of Sir Marshell Leach, Honey badger Amp, Apex A14 Amp. Also OM2.7, OM3 and ISOTOP from Russian website. Some preamps like P20, P30, P4 etc, some Subwoofer filters including sub filter from Mr. Mile, also eatched some soft start, Protection pcbs, power supply pcb, selector pcb etc etc... All are working.
Using some of the above PCBs, I assembled some Stereo Amps and some 2.1 amps using DIY pcbs and Old Dead AVReceiver cabinets (bought from scrap) . All are working without any major issue and with regular improvements.

There were some failed projects too. some noisey preamps were there, some amp designs were there like Jat501, Apex SR150, Apex SR100 etc etc etc.
I have Zero knowledge of PCB designing as of now, hence I need to rely on PCBs designed by the experts. But I will start learning it soon. My failure to success ratio is 1:1 as of now will try to improve it.

After 4 to 5 years, One thing I understood very well that "DIY IS NOT CHEAP". You will always end up spending more on DIY then on a professionally made commercially available similar powered AMP purchased. Also there are some issues like getting good quality (Original) componants at an appropriate price. I have seen vendors selling componants at 3 times 4 times the actual price. And the most important part is DIY have its limitations, It is not imposible to DIY Good Quality DAC, Preamp, Digital selector etc etc but the cost to do that all will be hudge. But, for me, I am not in a stage where I can turn back, hence I will continue doing it and will try focusing on Class AB Power Amps Only.

Since I have read about Apex A40 Amp a lot and every body was loving the quality of sound it produces, I also tried to assemble it. But kid of failed as for now and trying to find a solution on the forom.

Hello, my name is João, and I’ve been working with a few friends to develop an audio analyser called beHydra (feel free to check it out here: https://beembedded.com/behydra/).

I joined diyAudio to learn and understand everyone’s needs when developing audio related products, which is why I’m developing the audio analyser myself, given what’s in the market is a bit expensive and not necessarily developer friendly. Happy to discuss what I've been working on 🙂

Looking forward to exchanging ideas and learning from everyone here!

Hi there,

Up for sale is a large lot of power tubes, all in the octal family of EL34 / KT88 / 6550 / KT90 / 5881 / 6L6 and 7591.
All details can be found in the file attached to this post. All good and tested in my calibrated TV7-D/U. You can find the actual reading and the "good" threshold for comparison, plus a lot more information on condition, origin, etc. And if there's anything else you are missing, just ask.

Price for the entire lot (54 tubes): 650 EUR / 700 USD. Shipping (priority and registered) should be between 25 EUR (Europe) to 50 EUR (America / Asia).
Payment by bank transfer or Paypal F&F.

I am also open to reasonable offers, either for the whole lot, or for significant parts of it. Offers for smaller lots are also welcome, but please understand that I need to prioritize the bigger ones. I will try to reply swiftly to any offer or question.

After one month from this post, I will start offering whatever is left in individual lots, but prices will have to reflect the unavoidable overhead of smaller transactions.

Thank you!
Ignacio

Amazingly, 40 years on, one can still find plenty of DIY examples on the internet.
It has been on the to-do list for long.
And finally we got round to it ourselves, with 4 different version.
https://www.diyaudio.com/community/threads/hiraga-le-monstre.5462/post-7622181





Patrick

.

HI everyone.
I have spent some time designing my system and have now almost completed the preamp design. I am at the moment looking into a headphone output, and hence looking at various circuits around the bazaars. I came across a simple circuity using THAT 1646 balanced driver amps that looked promising but I am confused as to why the input is connected to the amp sense terminals rather than pin 4 of the amp. Hoping someone can explain why this has been done. I have attached the circuit here for reference

I'm not the seller, as these are somewhat rare to find I will share the ad found here : https://www.finn.no/recommerce/forsale/item/396684651

After 2 days of scouring Mouser, Digikey, image files, and various manufacturers, I can't find replacements for this 2mm pitch, 11 position, square pin receptacle/connector that include the PCB stabilizing clips at the ends. I've tried multiple keyword variations, and all I can find are right angle receptacles without the clips. I'm redesigning some burned circuit boards for a rare Audire amplifier, and would definitely prefer to replace this with the same configuration if it's even possible.

Any help much appreciated - thanks!

Buddy Holly is great music.

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My name is Prezmo and I live in Poland.
In my free time, I do audio electronics as a hobby.
I came across this forum looking for information on improving the JLH1969 headphone amplifier.
I got my hands on a ready-made Gzlozone board, but the sound doesn't satisfy me. I'm looking for solutions to improve it.

Hi. Thanks for reading. After listening to a pair of dipole speakers I am impressed by the spaciousness of the sound. Two drive units connected out of phase with each other, which destroys the direct sound reaching the listener’s ear, so the listener only hears the reflected sound from the room’s walls, causing a spacious kind of sound.

But it seems to be an expensive build, with more drive units required. Is it possible to mimic dipole speakers with a non-dipole speaker, by using the out of phase back radiation in some way. I tried it by removing the back of the speaker enclosure, hoping the read radiation would destroy the direct sound, but it did not work well, the dipole speakers sounded much more spacious

I've been using Fabfilter Pro-MB for some time now to reverse some of the effects of music track dynamic compression during mastering...but not mixing, of course. I've found some interesting tidbits of information while using it:

1) Dynamic expansion with a good multiband upwards expander (like Pro-MB) is possible if the dynamics of the music track haven't been completely crushed into oblivion. In DR Database terms, this means that if the music track has at least a 6-8 dB rating using the "TT Dynamic Range Meter" (the standalone version or as a plugin for foobar2000), you have a reasonable chance of a good listenable outcome using multiband expansion. The maximum DR ratings that can benefit from expansion seem to be around 13-14 (dB-crest factor) on the DR Database scale. Any tracks compressed to lower dynamic range values seem to be "lost causes". It's much more productive to go on a hunt for a higher dynamic range version of the recording rather than trying to use a multiband expander.

2) The plugin settings used to expand tracks are usually most successful in increasing dynamic range in the higher frequencies of a track than the lower frequencies below ~100-200 Hz. This means that the crest factor ratings of the tracks from the TT DR Meter may not change very much (due to the fact that crest factor is really measuring bass dynamic range, but not really higher frequency DR), but the tracks themselves will present a much more crisp and forward sound quality for dynamic transients--like ride cymbals, crash cymbals, bells/cowbells, glockenspiel, all handheld percussion instruments (particularly latin instruments), string attack transients, marimba/vibe strikes, drums, and human voice transients.

3) The "steeliness" of string orchestras used in lush pop string arrangements and mass string orchestral scores (classical) will largely be suppressed if using the multiband expander carefully. This was a big surprise.

4) Any human voices, particularly female voices, will begin to sound much more realistic and without typical harshness that comes with listening to recordings having compression applied during mastering. This was the biggest surprise of all in my explorations using the expander.

5) The albums that respond most strongly and easily to multiband expansion seem to include those that had analog compressors applied during mastering (e.g., early-late 1970s albums), which apparently used continuously varying nonlinear compression curves without a sharp knee breakpoint/thresholding applied during mastering.

6) The music genres that seem to respond most strongly are funk, rock, folk, jazz (including smooth jazz), progressive rock with lots of drums/percussion, dance, and related genres. Those genres that seem to respond with more difficulty include smoother music genres having lots of midrange energy (i.e., ambient, new age, classical string orchestras, classical guitar, etc.).

7) All tracks that have used multiband expansion need to be adjusted before and after expansion using parametric EQ demastering.

__________________________________________________________________________________________________________________________

It's clear to me that the use of a multiband dynamic range expander is now standard equipment for my typical demastering tasks that I perform on most of my incoming discs.

I don't recommend trying to do expansion "on the fly" (without review and adjustment of expander settings track by track) unless extremely mild expander settings are used, which leads to barely audible subjective differences in the compressed tracks. Using an expander offline and taking a little time to get the setting right to get the best and most audible results is my strong suggestion in using these type of plugins. However, the task only has to be done once for the music tracks--then saved for future listening, instead of having to set up a custom string of plugins and parametric EQ settings each time a music track is played.

(My original text above was originally posted to another audio forum, albeit an unstable one, and has been moved to this forum to ensure its longevity.)


Chris

I'm looking for a PC hardware expert who can help me diagnosing and repairing a PC motherboard.
Mombo is an Asus P6T deluxe ver.2 socket 1366.
Mombo fail in POST procedure.
Nothing ,dark screen.
I have check with another good power supply,another graphic card, pulled memory,nothing.
Power supply start,CPU cooler fan start,nothing else,no beep nothing!
Any help out there?
I have done some measurements around VRM and I didn't see any voltage across electrolytic caps.

Just got this on cd. Reminds of the Stone's Exile on Mainstreet. Good hard rocking vibe throughout. Not too much tired top 40 stuff either.

I've got a Jensen 15" woofer and three Jensen tweeters someone had pulled from an Emerson console and sold online. The woofer itself has Emerson on it which likely means it was designed specifically for Emerson by Jensen or maybe just that Emerson asked for their name to be put on it.

I used the cabinet from a cheap 1950 Silvertone console that wasn't worth restoring (except the V-M Tri-O-Matic record changer which will go in my Magnavox CR-198C console) as the cabinet for the speaker. The only good thing about the cabinet is the piece of plywood I used for the speaker baffle.

The stock driver configuration of the Emerson console had the woofer in one spot and the three tweeters arranged in a triangle shape at an angle from the woofer. I arranged them as seen here. The extra hole was when I had a pair of dome tweeters on either side of the top tweeter. Didn't like how it sounded as it wasn't very good up close to the speaker given the distance between the two domes.

So I found a ribbon tweeter I had and used that which improved the sound.






I made a 6dB/octave crossover for the speaker as that's what was originally used except the woofer didn't have any inductor in series with it. The reason for the added tweeter crossed over so high is I didn't like how the cone tweeters didn't reproduce the treble up to 16kHz that good.




The plan is to eventually have a new proper open back cabinet built and make it to where the ribbon tweeter mounts above the top cone tweeter.

Cannot decide if I want to paint it black like the baffle is or have it properly finished like the rest of the cabinet is.

Should I keep the 6dB/octave crossover or switch to a 12 dB/octave crossover as in will it make the speaker sound better or is it best to keep the stock crossover slope?

EDIT:

Here's a 12dB/octave crossover I've calculated. The .16mH, .32mH and .12mH inductors I could only get .15mH, .13mH and .33mH, however if I go with this crossover I will measure those and see how close they are to the calculated values, although if the inductors have a 5% or higher tolerance (website doesn't specify) they will be within the calculated value.

Good morning everyone,
my name is Alex and I come from Italy, and since we are on the subject of audio, I specify that I live in Sanremo the city known for the Italian song festival.
I have a PC with a b650 motherboard and a ryzen 7600 cpu, a xonar dx audio card and hivi t200c speakers.

I signed up to learn how certain mechanics work and improve my audio chain in the future.

Greetings from the Ligurian Riviera!

Hi all,

I'm trying to repair my Sony CMT-M373NT bookshelf stereo which has rendered excellent service as a USB sound device with my PC for nearly 20 years. 25W + 25W output, with 6R speakers

Symptoms:

• System starts as normal, relay clicks, sound is produced, but:
• Right channel has a buzzing/hissing sound and none of the actual signal. Constant volume, regardless of volume setting.
• Left channel is 100% normal
• Result is the same when switching speakers around
• Same can be heard through headphones
• With right speaker disconnected, the system can be used Left channel only with no issues
• With right speaker connected, the system drops to protection mode within a few seconds

Looking at the circuit diagram in the service manual and knowing that these components are well known for going bad, I’m strongly suspecting the STK403-030 amp module. Anyone think I might be barking up the wrong tree?



However I can’t probe it in situ because of the way the system is built; the STK and all of the supporting components are masked by other closely stacked and rigidly interconnected boards. However with the board out and checked on the bench there are no dead shorts on any of the transistors or capacitors and no open resistors.



I was planning to solder fly leads to the pins of the STK in situ and then reassemble the unit to allow me to probe voltage the STK pins with the whole thing powered up. Assuming that’s confirmed bad then a repair would be pretty straightforward if it weren’t for the lack of availability of these STK modules. They are listed on ebay and from China but it seems that more likely than not they will be fake, although they do look quite convincing, with the same moulded top cover.



Are ALL STKs subject to fakery? This system was bought in 2004ish so we’re not talking about parts that haven't been produced in 40+ years as with some models of STK. Does the top cap come off these easily? If so I could probably buy an ebay one, pop the lid and compare the internals with the original one, a fake will be easy to spot I assume?

Then as I’ve started reading I’ve realised that it’s probably fairly simple to directly replace the dead STK with a couple of either LM3886 or TDA7294. Could anyone offer some tips/pointers here?

As a hobbyist electronics tinkerer I’ve designed, built and coded many Arduino-based devices to do with supporting functions in car engine conversions etc so I’m no stranger to the skills required and I have a decent working understanding of the electronic principles at play, but the specifics of audio/amplifier design are beyond me at present.

Cheers!

Hi all,

Newbie here looking to get some advice and learning on repairing some old Sony minidisc units. Currently my Sony CMT-M333NT has died on me (push power error) so looking to get some help on that.

Sav

Just wanted to drop in and say hello! I’m a mechanic by trade, but I’ve always been fascinated by electronics and audio. Lately, I’ve been tinkering with some old speakers and amps, trying to learn more about DIY audio projects. Figured this would be a great place to pick up some knowledge and maybe even contribute where I can.

Hi, I have a nice cabinet of a very closely spaced MTM. Stock drivers are ok but deep bass somewhat lacking for size of cabinet. The woofers aren't really designed for it like a more modern driver. A warm but "woody" kind of sound.

Trying to decide using 2x Mark Audio Pluvia 11s per can either in 1.5 way or 2.5 way with tweeter - or both with crossover switch!

Or, using SB Acoustics 17nac woofers in 2.5 way or mtm and a low Fs tweeter (must be small neo/small flange due to existing arrangement).

How are the Pluvia with bass? Is it worth using them as more of a woofer? They would certainly be the more artistic option as the cabinet is a darkish orangey kind of wood veneer. Roughly 40L.

Drivers need to fit in a 172-3mm rebate if others are to be suggested; slightly smaller can work too

Offered for free is my childhood collection of Audio Xpress and Speaker Builder magazines. There are 90 different copies of Audio Xpress, 10 copies of Speaker Builder, one copy of Glass Audio, one of Audio Electronics, and some random copies of UHF and Inner Ear.

I was lucky enough to have these growing up as a kid / teenager, and would be happy to pass that experience on to someone else. These are by no means in collectible condition, but they are all clean, dry, and ready to be enjoyed.

Also included is a copy of Mullard Tube Circuits for Audio Amplifiers and one copy of Audio Amateur Power Amp Projects.

Free to anyone willing to pick them up in the Ottawa area (downtown).

Cheers,
Owen

Hello,
I created the board using special software based on the published schematics and PCB photos!
Shannon Park (diytube.com) Unfortunately, his website and forum have been taken offline,
but I saved everything I need (e.g., firmware for the Arduino Nano).
I used the top side of the original board as a template, and that was relatively easy.
The back side was more difficult, as there isn't a decent photo of it.
And the back side also serves as the ground plane,
so you have to look at the schematic again for each component!
To protect the tubes and fingers, these things are available at aliexpress



audio Xpress:
Our DIY section also brings a great project from Shannon Parks, with its Arduino-based real tube preamp.
This is a simple 12B4 linestage controlled by a Arduino Nano board and the tube power supply is probably
the simplest in our magazine’s history: it uses a 24V, 1A desktop supply.

I'm only selling the boards. If you're interested, please PM me.

Pre-BAF appetizer project posted. Enjoy 🙂

Build This MoFo!

EDIT: (6L6, Jan ‘23) above link is dead, here is link to the article - https://6moons.com/audioreviews2/mofo/mofo.pdf

Hope to see you at BAF! Don't forget admission is free for everyone who brings a project.

Let us know what you're bringing in in this thread:

What are you bringing to Burning Amp 2017?

IMPORTANT NOTE: A Zener diode from the MOSFET Gate to Source is needed to protect the MOSFET from inductor "kickback" I got away without this when using MOSFETs with high maximum VGS and lower supply voltages, but had not anticipate the range of FETs, inductors, and supply voltage that might be used. Even when the stars align, it's too close to not consider the Zener mandatory.

There's already a place for this diode on the PCB sold here in the store. When my stars align, I'll update the article pdf.

In the meantime, my good mentor PRR has provided some illustrative filler:
https://www.diyaudio.com/forums/pass-labs/313649-build-mofo-279.html#post6368030

Hello,
I have 2 chokes Hammond 193T (50mH 2A) for sale.
Shipping from Germany.

Hello. I come from Vietnam, my real name is Bao Huy, 32 years old. Sorry for my bad English. I joined the forum to learn more about DAC and Speaker design. Thanks everyone

Hi all

I am on a learning curve, and made my first speaker measurement today. I followed the great guide on Audiosciencereview Link on how to make Quasi-anechoic measurements with REW.

My aim is to get experienced with the needed measurement techniques, before i jump right into building my first speaker builds 🙂

I got the best measurements with the distance of 50cm, but all guidebooks recommend 1-2m. Gated window allowed ~5ms. I am sure i can improve that with better placement, and increased dampening in the room.

I would be happy with your feedback, tips & tricks etc.

Hello,

Anyone have access to the proper passive crossovers or schematics for the BM12CXA + BM-D440 compression drivers?

This is a coax 12"


thanks!

It is any other OPA equivalled with 2068DD?

ZD3 like new . in retail box. two sets of op amps, pair of xlr cables $133

CANARE 4S11 speaker cable. New. I have two lengths. one is 12 feet so you can make a pair of 6 feet speaker cables. the other is 24 feet so you can make a pair of 12 feet speaker cables. both are NEW. 4S11 comes in two colors black and gray. mine are gray. $88

Quantity = 3 Dayton Audio Designer series Paper cone woofers 6.5 inch. two are tested and are in perfect cosmetic and functional condition. one is new. all three are perfect.
https://www.parts-express.com/Dayton-Audio-DS175-8-6-1-2-Designer-Series-Woofer-295-428?quantity=1

quantity = 1 Dayton audio measurement microphone NEW.
https://www.parts-express.com/Dayton-Audio-EMM-6-Electret-Measurement-Microphone-390-801?quantity=1

Bundle of random DIY components
1- Quantity = 6 WBT RCA connectors
2 - quantity = 2 DC power connector large
3 - quantity = 2 DC power connector thin
4 - quantity = 2 DC power connector medium
5 - 20x 1uF mica capacitor for power supply
6 - power on/off switch with LED indicator
7 - power on/off switch with LED indicator
8 - brass ground connector
9 - quantity = 4 ceramic tube connector
10 - aviator connector for building DC power supply 4 pin
11 - aviator connector for building DC power supply 2 pin
12 - quantity = 2 IEC power connector with fuse and on / off button
13 - big AC power distribution with usb A and C
14 - gold plated copper sliding connector with shield around 40 quantity

everything together: $111 shipped. I only accept paypal friends and family