Hi friends! I come back with another rockford fosgate T10001bd, they tell me that they hit the speaker cables and it started to generate smoke. When checking it I found this damaged ground track, I did a general check and nothing else marked me wrong and when I turn it on the current consumption is high and the output fets make noise. I disassembled them to check them and I found no damage to the output fets and transistors mpsa56 and 06, I measured signals in the ic drivers u202 and u204, but u204 is always sending pulses, but 202 is not, when placing audio both send pulses. It is normal? and I see that the ic of u200 (lm6172) and u203 (tl072) are different greetings and I appreciate your support

Hi Everyone.
I have an amp that is 4x200w, and I am wanting to make it 100v for some distributed speakers on two of the channels.
The speakers all have transformers in them and are tapped at various wattages throughout each line on these two channels.

My questions come with regards to installing a transformer between the 8ohm Amp and the 100v line.
Will a transformer from an old speaker work? I have a transformer from an old speaker that had many taps/windings, one of which is 200w, the speaker it was installed in had an 8ohm driver. Would this work on one of the lines? Actual load on that line is likely to be around 150w at maximum, which it is likely to never run at. Would this work? I assume you would just adjust the volume of the amp channel to affect the whole line on that channel without causing any issues with the transformer?

If not, what type of transformer should I be putting in here? Any links to actual products would help a heap also.

Cheers,

I am from the Analog test and measurement equipment era , though I do have two Digital multimeters.
I am intending to buy a Computer , and want an operating system in it which will accept software for test and measurement of audio equipment.
I realise that many audio engineers use Apple/Mac systems though I prefer not that system unless there is no other comprehensive option.
I don’t much like Windows either , thus I am asking here ,
will the Linux operating system accept the various test and measurement software ?
or do the Manufacturers of test and measurement software only design and manufacture for Windows and Apple/Mac operating systems ?

Yes , you are correct , I am really not knowledgeable about computer systems ,
thus any information will be appreciated ,
or a direction to a Thread in this Forum , or to anywhere , where the above has been comprehensively explained.

It seems there are a lot of people, besides me of course, who have been wanting a good quality PCB for XBOSOZ. Well I have been pointed to several thread and tried to glean as much as I could. This is a completely no area for me, and my first pass at a Pass design. 🙂

First of all thank to all those who have paved the way. They did the real work here, this is very small thing compared to what those people have done.

So thanks to these people in Particular:

Nelson Pass, of course.
Metalman, Terry Aben whose circuit I followed.

But I am sure there are others who deserve credit, and to those I apologize for not knowing who you are.

Anyway they say a picture is worth a thousand words, so here are 2000 words worth.

First the Schematic:

Never used except for a 1 1/2” shorter lead that was trimmed off in anticipation of an amp build. Usually bought at $110 or so, but I can sell it for $75. Shipping from southern Ontario.

I dug out one of my old amps and remembered why I put it away in the first place:

I need the DIN power/control cord.
I think this is the part #:

Amp: Kenwood KAC-8020
Part # E30-1433-08

It's an 8-pin round connector (din), it is what runs up to the head unit (the input)
If I remember correctly, it also has the remote wire on it.

Any idea where I can get one? I found parts web sites, but most of them say 'discontinued'.

I could probably make my own cable, but I don't have the pin-out information either.

I hate to keep it moth-balled because I don't have the DIN cord.

Thanks,

H.


Here's a picture of the amp and the DIN connector:

I purchased an assembled 6AU6 6AQ5 SE Amp from China. It worked as assembled but was not correct. The PCB was not correct. I had to cut traces, add correct capacitors/resistors for the voltages and rewire correctly the power supply CRC filters. I have questions about the feedback connections circled in red in the attached picture. I have not seen this before having feedback connected to Pin 2 the suppressor of 6AU6. And I have not seen the R2 240k resistor feedback connected to the suppressor before, but to Pin 5 the plate. Are these connections correct or is this an error too? Or if correct how does this affect the circuit? >>>I have corrected schematic to show 5Y3GT and the added 10uf 400V Capacitor. ALSO Attaching Original Schematic<<<

I am thinking to have muses volume into b1k pre amp.
I found muses volume PCB.
https://nw-electric.way-nifty.com/blog/muses72323vol.html
https://nw-electric.way-nifty.com/blog/files/muses72323vol_sch_20210131.pdf
I only use for simple volume control without opamp.
Is this worth to build against Meldano's kit
or anything comment or tips?

Intrigued by how well my cheap 'n cheerful ESP P-06 build worked out, I've been playing around with an even simpler RIAA phono preamp made with an opamp > passive EQ > opamp. The PSU is repurposed from an earlier Hagerman Bugle build (+/-15VDC, using 7815/7915 ICs).

I believe I have a circuit that 'sounds good'. Now I'm playing around with the output capacitor and pulldown/load resistor.

I've looked around at other similar projects, and have found a lot of variation in values chosen for the output DC blocking cap (Cout) and output pulldown/load resistor (Rload). Many traditional 1980s-style designs use a large value capacitor with lower value resistor. For instance, I found one Graham Slee design which uses 10uF for Cout and 47k for Rload. Rod Elliott shows a few example designs that use Cout = 22uF and Rload = 22k ohms. But then the Muffsy and an ELENCO design use Cout = 1uF and Rload = 100k, while the RJM VSPS uses 2.2uF and 100k ohms. On the other hand, the TNT Solidphono goes to the other extreme and uses Cout = 0.33uF and Rload = 330k ohms, like early 1960s tube circuits would have used.

I'm left with the impression that the choice of values is less important than the resulting time constant. Is that correct?

One thing to take into consideration is the load impedance presented by the device to be driven by the phono preamp output.

- If that's going to be a preamp or amp with a 10k ohm volume control, then a 1uF output cap will result in an F3low of 16Hz, which would introduce a noticeable rolloff of low bass. (Perhaps that's desirable, though, as a rough 'n ready rumble filter?)

- Looking at the other extreme, if Cout = 22uF and Rload = 22k, and that feeds a 10k volume control, even though the resulting load (22k//10k) will be only 6875 ohms the F3low will be down at 1Hz. The Graham Slee values of 10uF and 47k results in F3low of only 0.34Hz, but if the load is a 10k volume control, F3low for that would go up to 1.9Hz. Perhaps that was the design goal there?

Let's say I'm not sure what my RIAA preamp will be driving. It could drive a class D amp with a 10k ohm volume pot on its input, or it could drive my living room stereo with an autoformer volume control (AV) which maintains a very high impedance load. Should I try to find a compromise solution that works OK with both? Or should I optimize for one extreme and assume it will be OK for the other?

I did have 1.5uF and 100k and that was working fine into the living room hi-fi w/ AVC. Into a 10k ohm load the F3low would go way up to 16Hz, but it's not a problem into the AVC.

Now I'm trying a 6.8uF 100V (big) film cap with 56k, which is predicted to have F3low of only 0.4Hz into a light load.
Should I change the Rload to 22k, so the F3low is about 1Hz? Would that conform better to best practices?

What are the pros and cons of bigger vs smaller value C and bigger vs smaller value R in this part of the circuit?

I understand that smaller value capacitors have advantages of lower inductance, lower ESR, etc.
Also, the smaller value of R allows the output cap to charge/discharge more quickly, reducing turn on/off thumps.

PS - Forgot to mention... I've read that a lower value of capacitance (e.g. 1uF) for the output DC blocking cap will have a higher reactance at low frequencies, so can make the circuit more susceptible to picking up hum from its interconnect cabling. Is that independent of the value of Rload? If that's true, and a higher value capacitor (e.g., 10uF) will help reject hum pickup, then perhaps that's an important issue for a standalone phono preamp as opposed to one that's built into a full-function preamp or integrated amp?

Are there other issues of importance?

Hello friends

Looking for a pair Scan Speak tweeters D2608/91300 in good conditions

Thanks
and regards

Josoeff

Various used and NOS/NIB valves for sale.
1) Two Sylvania 6S4A's matched in emission - 22.8mA/22.8mA one is yellow print, the other white. £20
2) Electro Harmonix tests good - 24mA. £10
3) Standard tests ok 21mA - £7
All tested at Va 250v, Vg1 -8v, Ik 24mA = on spec.
4) 11 Mullard PL504's NIB, not tested as I haven't a base. £7 each
5) Various used ECC88's, tested good to ok, pics shows various brands inc Brimar, Mullard etc. £4 each
6) A pair of ITT CV391's (807's in a different package) test good. £10 each
7) Pair of ITT CV428's ( 807's etc). £10 each.
8) Four 813 top caps, ceramic, new. £8 the lot.

Postage to UK mainland estimated @ £3,all valves will be well packed, happy to knock a few £'s off for multiple valves etc.

TFL, Andy.

I am researching EL34 PP amplifiers in the hopes of building one in the near future. Google throws up about 873 billion different designs though and I just want to make sure I'm not building an unproven (or even dangerously out of spec) design with brand new and very expensive components. My main amp is currently a Maggie 9303 EL84PP modified to Dave Gillespie's design and I am very fond of the sound. I need something that will fill a larger room though and I happen to already have a matched quad of EL34s. I think my preferred setup would be a dual monoblock in separate chassis from each other. Other than that I really don't care so long as it's a respected design and not overly complicated. Anybody want to point me toward something?

Hi All,

Just bump into these Chinese semiconductor website

http://www.xlsemi.net/products_AB_audio.html

Definitely have the numbers familiar?

Hi,

have some ALPS RK27x adapter pcbs left for sale.



Pot and connector strip not included, this is just the PCB.

Cost: 2€ each pcb

Shipping in a padded envelope (not tracked or insured):

• germany (without islands) : 2,50€
• world wide (without UK, sorry guys to much paperwork for the customs): 4,50€

After I unfortunately killed my Soekris dac due to an accidental short circuit, I am looking for a replacement as I got used to its sound. Could also be the latest rev.7 1021 board, as it's architecture s similar to the OEM board.

Hi all,
Some time ago I built a small battery powered sine wave signal gen to pair with my handheld oscilloscope for basic signal tracing. It's this one https://www.valvewizard.co.uk/siggen.html and seems to work well. But then I thought it would be nice to have a square wave output too so I added this one https://sound-au.com/articles/sqr-f14.gif from https://sound-au.com/articles/squarewave.htm in particular the sine to square converter scheme. This way I could have common frequency control for both waveforms but independent output level controls. The problem: The sine wave output is 0,7V rms and apparently this is not enough. You can see that the square is not symmetrical. When I feed it with another sine gen that does 2,6V rms things get a lot better however it seems it needs just a little more than this to be perfect. My conclusion is that it has to do with the Schmitt trigger input threshold but I wasn't able to understand the datasheet on this. So, the question is if there is anything I could adjust to make it work with 0,7V rms.

PS. The delayed rising time is intentional according to Rod Elliott's teaching. Also, the square out as is works to trace ringing, frequency response etc. It is also very good to tell if the DUT is inverting. 🙂 But it does not look nice...

Clearing out some stuff that I have no use for from a recent bulk-buy of parts and projects. Prefer shipping within EU, but world-wide shipping is of course possible.

**************
- Sjostrøm Audio SSR-03 no-compromise regulator PCBs with pre-mounted AD825 opamps.
On sale from Sjostrom for around 50 EUR, asking 30 EUR each. 5 Pcs. available.
https://sjostromaudio.com/pages/ind...ssr03-sjoestroem-super-regulator-power-supply


- Sjostrøm Audio DCT-02 DC-trap PCBs for buzzing toroids. Asking 10 EUR each, 2 pcs. available.
https://sjostromaudio.com/pages/index.php/hifi-projects/109-dct02-the-dc-trap-high-end-style

- Sjostrøm Audio SST-03 softstart PCB. Asking 15 EUR, 1 pc. available.
https://sjostromaudio.com/pages/ind...146-sst03-softstart-for-toroidal-transformers


- Original cViller F5 amplifier and PSU PCBs. Asking 15 EUR per set, 2 sets available.


- F1 amplifier and PSU PCBs. Unknown origin but looks fine, and in sealed shrink-wrap. Asking 15 EUR per set, 2 sets available.


- Aleph 5 clone PCBs from KK audio. Looks like an older version, unfortunately no PSU available. Asking 15 EUR for the pair.


- Original BrianGT Aleph-PCBs (can be used for Aleph 3 or Aleph Mini IIRC). Asking 12 EUR per pair, two pairs available.


- Original BrianGT/Chipamp snubberized LM3886 Gainclone PCBs (sorry, I have no spare LM3886s!). Asking 15 EUR for a set of two amp and two PSU-boards, 2 sets available.


- Zen V4 clone PCBs from jimsaudio, unopened. Asking 12 EUR for the pair.


- Jung/Didden Superreg-PCBs from the diyaudio-store. [SOLD]

Hey
Quick question does anybody know how similar they sound ?
I haven't heard r2905 970000
But i do like revelator and illuminator tweeters sound.
Price difference is rather small but r2905 97000 is very simple swap out tweeter.
R3004 660000 i must re-route front panel.
I have listened sonus faber amati furutura speakers and there is 2905 tweeter but i don't know wich one i suspect 930000 or 950000 becouse detail is little bit missing ?
I have own(ed) d3004/602000, r3004/602010, r2904/700005

Amplifier:
Musical Fidelity P270

Problem:
Lightning struck near my apartment at night.
The next day when I tried to turn on my device, smoke started pouring out.

What is broken:
This is an image from the web and the problem area is circled. This is where the capacitor and resistor are located. The value of the capacitor can be seen and no problem, but the resistor is so burnt that the value cannot be read. There are two resistances. One on each side. This connects the amplifier ground to the mains ground.


Situation:
My cousin is a good electronics mechanic.
Since the device is very old and out of warranty, and I don't have the money to send it abroad to an authorized service center (I live in Serbia), I let him fix it.
Since everything is burned, you can't tell which parts are involved, he needs an electronic circuit diagram.
I couldn't find an electronic circuit diagram on the internet.
I was looking for an electronic schematic of this device from the manufacturer. They forwarded my email to the authorized service. Of course they told me to bring it to them. There was no talk of an electronic circuit diagram.

Possible solutions:

• One of you has an electronic circuit diagram (or a link);
• one of you has the same amp, (take some photos of the problematic part).

Many many thanks

(Google Translate...) 🤐

Hi everyone,

After quite a few years not building any speakers anymore it started to itch again recently. With DSPs getting more affordable I wanted to try building some active speakers. I own a Sonos Connect so my idea was to connect 2 MiniDSP 2x4HD with a Toslink Optical splitter to the Connect. In that way I’d try to create some sort of high end Sonos speaker 🙂 So I started drawing something in Fusion 360.

For the look and feel of the speaker I took some inspiration from an existing German brand of speakers. I like the slightly angled baffle look and I wanted something that could cover the full range so it needed to be large enough to fit a decent subwoofer. My living room furniture is made of white oak and I have a black leather sofa, so this made up the choice of materials for the outside. Baffle with waveguide milled out of solid oak and panels with leather polstering.



Choice of drivers:
I personally like the sound of soft dome tweeters. I also wanted a good controlled directivity so I decided to integrate a waveguide. I went for the Scanspeak D2604/833000 after seeing the test results here. I intend to cross over at around 3 Khz
For high mid I chose Wavecor WF152CU14. I liked the low distortion of the driver and the look of the phase plug 🙂. If it helps in directivity we will see later. It should handle the range down to 500 Hz. For low mid I designed in 2 x Wavecor WF152BD03 in separate closed chambers of around 3.5 liter. They should handle down to around 125Hz.



For the low end I was looking what I could do with a volume of around 40 to 50l behind the drivers and came across this which I could integrate. So a 26W/4558T00 close to the floor on the one side and a passive radiator 26W/0-00-00 higher up on the other side to make it fit in the narrow housing. With the MiniDSP I will have the flexibility to adjust the response to the room a bit.



The setup is made of separate sections, which are screwed together with threaded inserts, so it is easy to take apart and tweak. Most of the walls are double MDF with some damping in between.



The amplifier is made of 4 UCD400OEM modules together with a SMPS1200 and is mounted with heatsinks underneath in the base of the speaker. I created a PCB with a simple gain stage to connect the amplifier modules and to fit some connectors for the drivers. Cooling air can enter from the bottom and exit on the side through venting holes which are covered by the cloth hiding the subwoofer.
I’m still looking to find a way to connect the baffle without the visible screws. I might try to do it with supermagnets. If anyone has other ideas how to do this, please let me know.
Next is starting to build something. I might do some youtube videos on progress if people are interested.

Hi.... as title suggests. Any experience out there?

I've used bypass caps on Mundorf HC series before, but wondered if there was any advantage to doing this on M-lytic AG series?

Not sure what the physical differences are between HC and AG caps and how a bypass cap might influence.

Thanks

I am currently working on a two-way ported passive design to act as my surrounds. This is my third design project and fourth build project (I built a predesigned Micro-Marty sub).

My first design and build was a fairly large two way build for my main L/R with Dayton RS-225-8 woofers and RST28A tweeters designed for a college project. I designed them with the help of a super friendly redditor. All design work was based on factory data and diffraction/enclosure simulations. It turned out good for my expectations.

My second build was a MTM center speaker using the same tweeter as before, but two RS150-4 woofers. Same thing, all just theoretical/simulation data used to design.

This go around I am using the RS150-8 and RST28A, but am trying to improve my process by taking and using my own measurements.

For FRD measurements I have a Dayton EMM-6 and Scarlett 2i2, I built a turntable/spinorama style stand to allow for off-axis measurements while keeping my drivers at the center of rotation. For ZMA I am just using a make shift DATs with resistors and my 2i2. I follow this guide here.

So far I have learned a decent amount and made quiet a few improvements that I will then propagate to my previous builds.

1. During initial impedance measurements I had a weird peak/dip in my woofer's impedance. Found it to be a cabinet resonance. Gluing a brace/strip of MDF between my two side walls fixed that. I am sure that is an issue for my 8in mains that can be fixed.

2. I found during initial farfield measurements of my tweeter that I had some pretty nasty peaks and dips from about 2-6 kHz. I found out part of my issue was having a surface mounted instead of flush mounted tweeter. I fixed that, and my results got better but still some improvement left on the table. I at the same time went ahead and flush mounted the tweeter on all my previous builds.

3. In a continued effort to fix my tweeter dip, I rounded my edges with a 1/2in radius roundover bit. I will eventually roundover the edges on all my other builds too. The result I got from this was slight improvement in my 2.5-4 kHz dip, but not much. It did also decrease the hard fall off I had above 10 kHz by a significant amount.

Here is an album that shows some of my measurements throughout my process and what the actual speakers look like.

So now I am trying to figure out what else I can do or should be looking for before taking a final spin measurement and starting crossover design.

I still have a significant dip between 2.5-4 kHz. I believe the biggest culprit here is having a symmetrical/centered tweeter. I know the obvious answer is make my tweeter off center, but that also sets me back a lot because it means cutting off my baffle, cutting a new one, and then repeating the counter sink and round over process. I was suggested at looking into a waveguide if I don't want to rebuild the baffle entirely, but I haven't found any that would fit on my 7.5in wide baffle that also has the proper mounting for the RST28A. I will likely look into the SEOS-8 for when I go to revise my mains build, as it will fit on those.

There are a couple of things I know now that it is slightly too late for this go around, unless I want to back track a lot, that I will use going forward. Making my tweeter off center and getting my tweeter as close to my woofer as possible.

So my questions now are:

1. Is there anything I can do to help mitigate my dip that is almost certainly (VituixCAD diffraction simulator agrees) due to my centered tweeter, without cutting off and rebuilding my entire baffle? I have been suggested 3d print a waveguide, but that is a whole can of worms I am not sure I want to spend the time learning about, plus I have heard wave guides have their own trade offs. As well, I was told I could put some F-13 felt around the tweeter, but then I have to find a way to make that aesthetically pleasing.

2. I made my tweeter flush, could I expect any benefit of making my woofer flush as well?

3. How do I properly handle merging my near field and far field woofer measurements seeing as I have a port? The previously listed VituixCAD guide suggests taking SPL nearfield measurements from the woofer and the port, but I am not sure how to handle merging both of those with the farfield.

4. I have largely just been looking at my SPL and ZMA results for things to improve. Are there other factors I should investigate before carrying on?

I am selling a pair of Bianco 15" woofer for open baffle use. I just bought them from TLHP ( France) and open the box to look at one of them.
I realized that the OB I was about to try is too big for my room and low WAF. They are new and didn't think I need to post pictures as they've never been used.

Asking $275 shipped to USCON.

Like a lot of people on this forum I want a quality measurement microphone but can't really justify a full professional set up.
B&K (or clone) external polarisation capsule + proprietary preamp body + external power source is the laboratory reference option but is expensive, inflexible and inconvenient (not easily portable).
But the cheap Electret Condenser Mics tend to distort and lack stability over time.
So what are the options in the middle?
1. Use a B&K style capsule connected to a 48V Phantom powered body to provide the 200 V polarisation and the preamp (actually an impedance buffer)
Gefell and Josephson make such bodies but they are expensive and would be somewhat difficult to DIY.
[Edit] So it's a nice option if you have the money to buy ready-made "top of the line" hardware.

2. Use a B&K style pre-polarised capsule connected to a P48 V body that only has to do impedance buffer duties.
These capsules are fairly common and almost a commodity item, made by B&K, DPA, G.R.A.S., Gefell, Norsonic, ACO Pacific and PCB Piezotronics, so there's a reasonable chance to find one at a reasonable price.
This is a simpler system and naturally there are more commercial options.
ACO Pacific, and PCB Piezotronics do "measurement mics" and DPA market their's as an omnidirectional, flat response "studio mic".
MicW is the studio mic division of BSWA (like DPA to B&K) and claims to have a decent measurement mic.
These are less expensive but still not cheap ~$500 for the body.
I have had a look at some of the JFET buffer circuits from Borbely and Bob Cordell and it seems it shouldn't be too expensive.
[Edit] So a DIY kit could be almost "top of the line" and more than adequate but far cheaper.

Since option 2 is my choice - has anybody seen a DIY "preamp" body kit or have ideas for the circuit?
I am tempted to try a fully complementary circuit but a kit makes sense, if only to simplify the mechanics, so I don't have to cut odd sized capsule threads and find Teflon insulators and similar obscure parts.

David

Hello, i'm starting a new speaker project using Markaudio drivers. First a small desktop cabinet using the chn-50 in a bass reflex enclosure, for this i need an amplifier that can deliver 7 watts in 4 ohms, it can be class A or AB.

The second project will be a bit bigger, using the CHR-70 in a pensil style enclosure or some other form of floor standing cabinet, for this i need an amplifier that can deliver 20 watts in 8 ohms, i think for this one a class AB amp will be ideal, maybe something around the lm3886 chip.

What would you guys recommend for both this projects? any help will be greatly appreciated, thanks in advance, cheers

GONE

I'm giving away a centre speaker and a pair of floorstanding rear speakers. Pick up only in the Newcastle NSW AU area.

Centre: Vifa NE19VTS-04 + SB12PAC25-4, Vb=5 litres, F3=71Hz, 417mm W x 142mm H x 200mm D, 2nd order xo, comes with spare tweeter, grey grille cloth, oak timber veneer, birch ply top.
Rear: SEAS 27TFFC + Peerless 850488, Vb=7.3 litres, F3=60Hz, 270mm W x 416mm H x 140mm D, classic series xo, grey grille cloth, designed as mirror pair (tweeter out) to sit beside the sofa, beech timber veneer.

SOLD

Before I list this on eBay I'm listing it here for $150.


For sale is a 10" Subwoofer based around the Peerless 830452 XLS driver with a Parts Express 300-793 Subwoofer 250W Amplifier with remote control. This is side firing to the left and the B&W dimpled FlowPort is down firing. Made from Tassie Oak veneered MDF and both the inside and outside are sealed with a polyurethane finish. Cleats are used in the corners and the baffle is braced behind the driver. Comes with the remote control, IEC power lead and the amplifier manual. Used in a HT system in a 4.2m x 5.7m room with 2.4m to 4.2m ceiling height and provide plenty of bottom end for movies and was used at sensible levels.

Vb=40 litres, F3=27Hz, 376mm x 376mm x 476mm high.

Pick up only in the Newcastle NSW AU area.

Hello all!


I've been given a NAD C315BEE by a friend on the vague hope to repair it. From my diagnosis there is a distortion/muffled sound on some sounds. It's much more noticeable when the music has more sounds (ie. playing a sequence of sine waves from a youtube video from 10Hz to 20K Hz does not cause a problem).


All the inputs suffer from the problem, so I ruled out the input multiplexer circuit. The problem is also consistent across the full volume range so I also ruled out the power amplifier. Also, it happens on both channels.


I'm very dumb in terms of electronics but I started poking around and decided to test the voltages provided by the PSU board. From my tests the multiple voltages provided by the PSU board are ok (-45/+45V; 8V).


I then turned my attention to the +17V/-17V circuit (service manual, page 17) I noticed that the +17V rail is OK BUT the -17V is reading -1V ! (Amp on, no speakers connected, no inputs). Also the 5V regulator is working fine. I took the measurements between J169 (DGND) and J13 for +17V and J90 for -17V



For reference:



I measured +35V and -35V on the right side of the circuit, so that rules out the big capacitors after the full bridge rectifier. So, this leaves me with the -17V rail and I'm at the limit of my electronics skills here 🙁



I removed C330 and C315 for ease of access which appear ok (I'm unable to measure 2220uF with my meter but since the -35V rail is OK the cap should be ok too).


I then removed the capacitors from the left side: C308, C309, C310. All checkout fine. I finally removed the Q305 transistor but I'm unsure how to test it.



So, I would like to ask for your assistance. What can I test, is there an obvious culprit for the missing -17V? Should I replace all the bottom transistors ? (Q305, Q307)? Should the top part components also be replaced?



Something appears shorted here but then again I'm measuring -1V between the -17V rail and GND which (I think) rules out that possibility


Thanks!

I want to replace an iPhone 5s that does a good job of recording far-field sounds with its MEMS mic and +10.00 mic gain. It is probably doing AGC and noise cancellation.

I could get a Zoom H5 with a Clippy XLR ECM mic (Primo EM272 capsule) using phantom power to provide 8v to the mic or the Clippy ECM mic (same capsule), but only 2.5v from the mic/line input port of the H5.

Or I could get either of those mics, but use a single-board computer connected via USB to a Scarlett Solo or RODE AI-1 audio interface that connects to the mic via XLR or instrument port.

So far the budget portable recorders don't provide the Clippy capsules with enough power via the mic/line input port (Zoom H1n 2.5v, Sony PCM-A10 3v).

The Zoom H1n with onboard mics is not good enough to replace the iPhone: low audio levels (+39dB mic input gain) and noisy XY mics.

I see that on Rod's ESP site, he measures his MOSFET amp as having noise of <2mV RMS (unweighted -54dBV). A voltage seems to be a sensible way of measuring noise, but I'm not informed enough to know how this value compares to the SNR quoted on most commercial amp specifications (e.g. 116dB), so i'm struggling to put it into context. I realise this will be an incredibly basic question for many of you, but for thicko here is it comparable or convertible in some way? Or is it good/bad/indifferent?

For those of you who have built your own version of these, I'd also be interested to hear if you felt it was quiet or noisy compared to other amps, and if you made any mistakes or improvements etc related to noise/hiss/hum and so on of the completed amplifier. Obviously there'll be other sources of noise too (interconnects, upstream equipment etc) but this question is particularly about the amp (and PSU) built to Rod's official guides.

I'm interested in this particular amp because I already have the official PCBs and most components, bought some years ago but then never built. I'm trying to decide if I'd want to build and use it for my next application: an active configuration involving potentially sensitive drivers rather close to the listening position. In this case it seems very likely that noise of lower levels than normal might become quite audible, and I really dislike hearing it.

Thanks,
Kev

Say you have a biwire speaker with a low freq section with a 4ohm impedance, and mid/tweeter section 8ohms. You biwire it to a tube amplifier that has both a 4 and 8 ohm winding, connecting the speaker sections to the appropriate transformer output terminal.

What's the effect?

Amp came in stuck in protection mode .

I pulled the output driverboard and the amp powered up .

Amp is producing +- 12 volts to all opamps .
Amp is also producing rail voltages .

Any ideas where to start checking ?

Looking for $50 + shipping, which may be at least $20.

Hi I would like to sell my lovely AYA4 (2018) fully built-up by Audial. It is the model with USB to PCM input and SPDIF input
The condition is almost as New
220V input

Euro600 for your consideration, shipment will be depended on where to post to.

Last year during first lockdown, I build a TAS3251 Class D amplifier by joining a project of DIYA member JMF11. This amp is named NetAMP, build threat is: [design log] Neat 2x170W I2S in, I2C controlled, integrated DSP amp (TAS3251).
I was very happy with the result and decided to use it as my main amplifier in the living room. I then decided to build a preamp, then an enclosure. One design goal was to have something user friendly (easy to use) intended to all family and friends usage.
As the project as reach a point where I can confirm it is viable, I can publish information and progress in this thread

NeatAmp Pre handle:

• Remote management of NeatAmp board, including fault and error handling.
• 7 sources: 2 spdif, 2 toslink, 2 Bluetooth (1 open and 1 dedicated as a bridge with the TV), and 1 USB.
• Bt sources are 44.1 or 48k, spdif and toslink 44.1, 48, 88.2 and 96k and SUB will be 44.1 or 48k.
• IR receiver (NEC & Sony protocols).
• UI made of an OLED display and two rotary encoders.
• Soft start / power control for the SMPS.
• User presets memories.
• Auto power-off.

NeatAmp Pre allow user to manage:

• Volume, Mute, L&R balance, mono/stereo, DSP presets, balance of the input level of each source, IR remote code learn.

The block diagram shows Pre is made of four different boards: one for the rear panel with the inputs connectors; one for the front panel with display, IR receiver and encoders; one for the main board and the last for the power supply controller. This design mostly address large enclosure, this choice was dictated by NeatAmp board size and also the SMPS size. Boards are connected via flat cables and JST connectors.

NeatAmp Pre is built around a STM32F303, an AK4118 and two CSR Bluetooth modules. CSR, now Qualcomm, provide AptxLL (Low Latency) codec which is required for TV sound.

Power distribution is a bit weird, during development I encounter overheat issues on main board +5V reg and I installed a pre reg on the front panel board. This could be simplified.

NeatAmp Pre manage NeatAmp through a serial link. For this purpose, NeatAMP receive a specific software which manage low level TAS3251 duties including error management and communication with NeatAmp Pre. NeatAmp Pre also run its own software. Both software are made with ST suite: STM32Cube, they’ll be available on GitHub once acceptable quality level reached.

Current project status is ongoing, but everything now is stabilized and is viable.
What remains to do?:

• Power controller board. I am not satisfied with the current one, I’m currently switching to something closer to the one of Mark Johnson with an ‘always on’ supply made of a AC/DC converter module. I’ll update later when tested ok.
• Software modules:
  • Storage and management of the DSP presets
  • IR Learn
  • Shut off screen
  • Some minors enhancement
  • USB Audio (will be made last after the enclosure)
• Make an enclosure.

Block diagram is attached.

OPA627 VS Burson V5 VS Burson V6 Classic op-amps perceived quality of musical reproduction



Images show OPA627, Burson V5 and Buson V6 Classic op-amps and their relative size and Cyenne Dac 3100 with OPA627 and two V6 Classic op-amps in sockets.

In exchange for two Burson Classic dual op-amps I writing my honest review of their performance in comparison to OPA627 and the Burson V5s I already own. My system is composed of an Asus Gaming Computer running JRiver media center 28 which I modified to up samples all files to 96 KHz which are stored on a Passport external hard drive. I have an external Musiland external soundcard I use with optical out through a glass optical interconnect to a Cyenne 3100 DAC with three replaceable op-amp sockets. The DAC output goes to a Cambridge Audio CXA60 amp. I have Mordaunt-Short Aviano 6 speakers and a Wharfedale subwoofer. The Mordaunt-Short speakers are very neutral and I can hear differences in op-amps in the DAC typically. Lm4562 DAC chips came with the Cyenne Dac originally, but I found that despite their low THD, they sounded clean, but lean in expression. I previously owned a Maverick D2 DAC with 3 OPA627 op-amps and found them more textured and open. So, I am reviewing the OPA627 as a baseline vs the two Burson op-amps to see any difference. My dedicated music room is highly acoustically treated and has a fairly straight equalization curve along the x axis, according to Room EQ Wizard, without any equalization required in the JRiver application to achieve this.

Above example of ceiling room treatment in dedicated music room.

I plan to look at the factors Buson uses to describe their op-amps: transparency, details, color and texture, and dynamics and soundstage when reviewing the op-amps previously noted.



Definitions:

Transparency: the sound is reproduced accurately without adding much extra.

Detail: hearing the small, fine sounds of a recording; the most delicate parts of the original sound.

Color and texture: Tone color or texture of a sound is the quality that allows you to separate two instruments playing the same instrument. It is hearing the different layers and harmonies of music.

Dynamics and soundstage: variation in loudness between notes in the spatial sound image presented to the listener.

The Cyenne Dac has 3 removal socketed op-amps. One serves as the combiner op-amp for the DAC. The other 2 are gain op-amps. The gain sockets will vary from OPA627 to the Burson V5 to V6 Classic. I will listen to a playlist of Flac files with a minimum 44.1 KHz and make note of the above features of transparency, etc. for any sound characteristics noticeable for each op-amp.

The flowing listening set began with OPA627 in all three op-amp sockets followed by the Burson V5 in the gain op-amp sockets and OPA627 in the combiner op-amp socket.

Just Like A woman, live 1966 by Bob Dylan.

This is a beautiful 6:45 version of his studio song which only totals 4.42. With three OPA627 op-amps the song sounded less detailed than I remembered. I choose my music in this evaluation for its familiarity. Reference material I believe has to be used so elements like detail and transparency can be assessed as different from a typical presentation. Dylan’s guitar and voice had a sadly muted color and texture. It is actually a beautiful version of a song in which we hear Dylan during a grueling tour in 1966 play an emotionally different version of his studio album while leaning over his stool and expressing longingness and compassion during the song, not the derision found in his studio album. But the OPA627 did not convince me I was listening to a live performance of a wasted Dylan generously sharing his feelings of tenderness with the audience.

The Burson V5 turned the concert and song live. We hear the echo in the room and his harmonica notes are more present. The detail is enhanced. The whole texture and color of the song in his strained voice emoting his regret, not his signature emotional sneer. I was enveloped in the large soundstage during the recitation.

Visions of Johanna by Dylan, the studio track from Blonde on Blonde

The song sounded less textured as well with OPA627. Details were diminished. The notes were compressed and the soundstage was narrow. The music was colored, but with all the instruments the same color.

The Burson V5 again brought out more realism and detail. The bass is strong and Dylan sounds real, unfortunately the soundstage is medium but with clear textured sound separation with sharp details and accuracy.

Hotel California by The Eagles, MTV Live.

With the OPA627 texture is diminished in the guitar work and soundstage is small. Details seem muted. Henley seems to be singing through a cloth. There is a lack of presence.

The acoustic guitar beginning Hotel California with V5 sounds beautiful in its richness of color and texture. Its presence jumps out. The bass is stronger and more defined than with the OPA627 and the music seems more accurate. The instruments keep increasing as the music plays, but textural space is created for them. The vocal could be a bit more forward. But a great performance.

Maggot Brain by Funkadelic

In Maggot Brain with the OPA627 the echo in the left speaker of this beautiful guitar solo brings me right into the studio. The OPA627 shines when outputting the distorted gritty sounds displayed on this cut. I would like a little more transparency here but otherwise great performance.

With the Burson V5 the vocal is more authoritative at the beginning. The drum slap and echo is more detailed and colored. The solo is more in our face but cleaner than with the OPA627, although I like the OPA627 distortion guitar solo sound better. But the sound separation and soundstage are excellent with the Burson V5.

Early in The Morning by Harry Nilsson

Nilsson’s bouncy bluesy keyboard and infectious voice and lyric wants to be heard. But despite some spatial openness and slight texture separating the notes and voice, it was a boring presentation with OPA627.

The Burson V5 again has more detail and bass, but still the song lacks a natural and intimate soundstage. I want it to feel live, but it doesn’t.

The Burson V5 performed better to my ears and with my equipment. It was less fatiguing than the OPA627. The frequency curves on the OPA627 and Burson V5 were identical with REW. The great details, texture, dynamics, presence, etc. with the Burson V5 made no change in frequency distribution. They were both flat.

Burson V6 Classic performance

The Burson V6 Classic has been reviewed frequently and most reviewers indicate it is a good fit for folk and jazz, but not necessarily for rock. I found this to be false in my evaluation. It is clear it is more evolved in design than the earlier produced V5 I already owned, particularly the better ventilation qualities on the V6. I chose the V6 Classic for review since the V6 vivid is based on the Burson V5 I already own and they seem to share many characteristics, if the published qualities in the bar graph are accurate. The V6 Classic seemed different enough from the V5 to interest me in hearing its advertised sonic qualities. Since the OPA627 was found lacking in previous listening tests, I filled the integrator op-amp socket of my Cyenne DAC with a Burson V5 and the gain op-amp sockets with the two V6 Classic op-amps I received from Burson.

Joni Mitchell The Asylum Albums CD 1 and CD 4

I could hear Joni’s deliberate pace in the music that reveals a more realistic music experience with a musical texture which was beautiful and a clarity that was a joy to listen to. Joni in her live performances on CD 4 sounded present in my listening room with a sweetness to her voice which was big and clear, yet textured with a beautiful timbre to her voice I had not previously heard; while the instruments encompassed me. The bass was strong amongst rounded vocals that echoed slightly, revealing the music hall acoustics. I had never heard such a beautiful voice, although I had heard Joni many times. Her phrasing was accomplished.

Roger Water, Amused to Death

The first five tracks on Roger Waters album, known for its soundstage and imagery, were listened to with the Burson V6 Classic op-amps. The spacing and imagery of the sound was revealing. The texture and dynamics of the music, with pounding drums, electric solo and the chorus singing was three dimensional. The best I have heard it. The presence of Rogers is real. I had oddly read that V6 Classic op-amps were not good for rock, as previously noted, but I had a different experience, the realism the Classics added to my system was nothing short of fantastic. The sound wrapped around the back of my head. Roger seemed to be singing in my face, he is so close.

Early in The Morning by Harry Nilsson

I was so disappointed in the OPA627 and Burson V5 presentation of Early in the Morning I wanted to give the V6 Classics a chance to properly present the song. With the Burson V6 Classics in the gain op-amp sockets, finally the song sounds real. The keys of the piano can be heard vibrating and Harry is singing next to me. There is a brilliant texture revealed by the Classic op-amps positioning the struck keys, voice and distant bass. The realism of musical timbre is heard with an energy in the sound not previously heard.

Kill City by Iggy Pop

Kill City with Iggy Pop followed, with Williamson on guitar. It is a textually dense and energetic album. Iggy is in your face while the listener is enveloped by this forceful performance. The songs are catchy and floor tapping. This is a vinyl quality performance in sound by the Burson V5 and Burson V6 Classic combination. There is an absence of distortion, only the authentic presentation of instruments and voice. A total musical presentation which envelopes you with its presence, but maintains its clarity and details. The tone of each note is cleanly offered, while grandly performed by Iggy in his musical prime.

Avenue B By Iggy Pop

Avenue B by Iggy is live and his spoken introduction echoed as he spoke in the theater. His guitar strings vibrate in front of me. His voice is huge. The texture and details of the sound is so real I believe I am in the theater. Details in his performance are brilliantly revealed in this live performance which I had not previously heard. It was obvious now the sounds were always present, but only revealed now with the V6 Classics by Burson in the Cyenne 3100 DAC.

Clearly many things affect my perception of the V5 combined with the V6. The V6 Classics are so much more realistic and texturally beautiful than the Burson V5. I am sure though they complement each other with their revealing musical presentation and clear sonic image and details previously never heard on my system. I would pay for the V6 Classic op-amps if they were not offered free for a review.

I do not know how Burson was able to design the V6 Classic to produce such a musically beautiful presentation, suitable for rock or folk in my view. I am convinced from my experience that op-amps can change the sound signature of a DAC for the better. The best op-amps I have found to my taste have been made by Burson. I realize an op-amp interacts with the DAC circuitry and another DAC might produce a difference in sound quality, but in my system, the sound was superb and these op-amps will remain in place.

The value of discrete op-amps is denigrated by many, however many such as PS Audio co-founder Paul McGowan mention the intolerance of heat of IC op-amps due to size limitations; while discrete op-amps are not limited in this respect. He further states that discrete op-amps can be designed with a certain sound in mind and the use of the highest performing circuits and endless designer choices. Of course, the success of the project depends on the ability of the designer.

Specifications can not alone explain an op-amps appeal. Claims that discrete op-amps have more distortion than IC op-amps is usually used by critics to support the claim discrete op-amps are not worth buying or using. Gaskell’s 2016 dissertation at McGill university mentions that many designers and critical listeners believe that low level distortion may impart a beneficial tonal, timbral, or transient quality. Small amounts of distortion in a signal affect the “warmth” of the sound. He lists several studies showing that distortion is preferred by many listeners and his dissertation’s conclusion is that low level capacitor and op-amp distortion was found pleasing to many listeners.

Metrics can only replace subjective testing if they correlate well with a jury response. Measurement equipment in audio has to correlate with what people find pleasing in music, not what measurements such as THD say is best. Gaslighting someone who says something sounds good when it has poorer specifications is not a way to improve the science of psychoacoustics in the development of measures which quantity what consumers find pleasing.

Finally, Harman’s “How to Listen” is a computer-based, self-guided software program which teaches listeners how to identify, classify and rate the quality of recorded and reproduced sounds according to their timbral, spatial, dynamic and nonlinear distortion attributes. Other critical listening programs are available and can be used to improve the evaluation and improvement of audio systems through trained listener evaluations of products. Many audio companies do this to improve speakers, DACs, Amps, etc. Improvement of listening skills may improve the opportunity for agreement between critics as to what sounds good. One study found audio salesmen had the best listening skills, better than music critics.

I would be happy to review any discrete op-amps that manufacturers wish to have honestly evaluated.

It's been a bit hectic in the shed since I decided to give skiing away. If you think building speakers is expensive try skiing in Australia.
Anyway I just counted up what I am doing and I think I need a bigger workbench.
I have a RoadKill set I am testing to determine real world power handling; now playing Chill-Out annual 3 non stop.
I need new bedroom speakers and i am cutting wood for a small 3-Way
I have 2 big cardboard boxes set up as well. One a Bloody Big Box; again a 3-Way and a smaller 3-Way using the recently purchased Peerless dome and in the back of my mind is changing the small box to a big 4-Way.
Although I am having fun I probably start more projects than I'll ever finish.
Anybody else have a similar affliction?

Good morning all!

I just heard from Peter at Audio Sector and he's currently selling pcbs only. No worries, I'm fine with buying my own parts. I've heard nothing but good this about this kit but might as well make the opportunity to make a few improvements.

What you your thoughts on ways to improve these kits in terms of components selection? Better capacitors or diodes, etc. I have experience buying components to spec but no idea how they might affect sound in audiophile equipment.

Also, who the hell is selling lm4780 chips right now?! I can't seem to find them anywhere

Thanks everyone!

Hi, I recently acquired a KEF KUBE 104/2 active bass equalizer. I know it was designed to use specifically with the KEF 104/2 speakers. But, I’m thinking to use it with the non-KEF speakers—ADS L1590/2. Regrettably, the brochure and the owner’s manual of the KUBE 104/2 have no technical literature anymore. Does anyone know its specifications, such as tuning frequency and Q? Also, it would be really appreciated if anyone know how to modify its tuning frequency and Q.

I needed a small cheap mic preamp to add an input for a project and I was surprised to find very few kits for that (at a reasonable price at least). So I decided to roll my own. It's a bit overkill for my current application but I wanted something I could re-use later on.

It's a mix of datasheets for the ina217-ssm2019-that1512. It has provisions for phantom power and a volume control in between the ssm2019 and the output if desired.

Values for c8-r9-r10 depend on application.

PCB size is 5*3.5cm. All connectors molex kk.

I'll order the pcb soon. So if anyone has corrections to suggest, I'll gladly listen.

Hello everyone,

I just mounted the Vifa/Peerless XT25TG30-04 on a Visaton 148 R waveguide and thought that I'd post some pics of how I did it in case someone is interested.

First here is the couple:







Due to the slightly curved front plate of the XT25 a small ~2 mm gap remains between the tweeter and the wg which I thought might not do any good to the sound:





A cheap solution was found from blutack or what ever this sticky stuff is called in English. I made a thin donut and placed it to the very edge of the opening in the waveguide:





I also put three small blobs of blutack on the tweeter face while I aligned the wg to make sure that it stays in place while I mark the holes.




I don't have any pics of the alignment process itself, but it's of course very straightforward. I used a 4 mm wood drill bit to push small marks on the wg wg back plate as the bit fit the holes nicely and was thus centered well enough automatically.

Some of the holes go close to the original mounting holes with thread inserts on the wg, but the original holes are not at the same distance from the center point as the XT25 mounting holes so they can't be used.

I first put some white paint around the mark to make the mark more visible for drilling and used a normal 2,5 mm metal drill bit for the hole.






3,5x13 DIN 7981 screws and washers were used to mounting:









And here is the result from the front:











I think they came out quite nice.

No idea how the sound is yet, though.😱

As per title i m looking 4 woofers with spec Fs no less than 25 hz and xmax 9 mm at least . Also 15 inch minimum . I m from Europe , Romania .

Thanks

Hi all, I'm getting ready to build an audio sector gainclone in monoblock configuration (ordered the lm3875 kit but sounds like Peter may be sold out of those, so maybe lm4780 instead?). I see many people recommending 300va transformers for these, however being a firm believer than anything worth doing is worth overdoing I ordered 2400va [Edit: Two 400VA transformers!] units. I'm now having concerns that these might cause issues due to being too big. Are these fears unfounded? Is it safe to oversize transformers so much? Thank you!

I have for sale the following evaluation boards from Tripath.
All evaluation boards include the the cables and plugs, power
supplies are for the power section.
RB-TK2350-70€
EB-TA0103-100€
RB-TA0105-130€
Payment by PP shipment for Europanean Union
For more information PM

Hi all was wondering if anyone might have the FRD & ZMA files for the SPE21. Parts express is out of the DAT V2 and the new DATS V3 is not in stock until November and i just can't wait that long if anyone has those files i will pay to get them. Also is anyone in the Tampa Florida area that can measure the speakers for me i will gladly pay you for your help. thank you

The cheap "Party" boxes have been returned by my middle daughter.
They started whizzing and squeeeaking and became unlistenable when turned up to party volume.
I made these from some roadkill boxes I picked up and all the cheap hot melt glue seams look like they have dried out and vibrated loose.
OK I can fix that but I promised my neighbor a pair of cheap speakers for his shed and I want to re-use these repaired boxes.
I also want to use the little 4" midrange in something else.
My mate is a saxophone player and I know he likes a hot top-end in his speakers as he is always turning up the treble control when he's here listening.
I have a pair of the Peerless 830656 here which will work but I'm wondering if the bigger 830657 would be a better choice if I can squeeze it in.
Using two cheap 8" woofers and crossing around 300 to suit the existing coils I have, the tweeter I want to use is a cheap one and needs to be crossed around3K.
I think 3c to 3k is more suitable for the smaller Peerless but just double checking with the experts here

I share my ~shop with my brother and he is making these for fun.

https://en.wikipedia.org/wiki/Icosahedron#Convex_regular_icosahedron

What would anyone think the downfalls of using this shape for a cabinet be?
I,m thinking that a four inch bezel, 3 to 3.5 inch cone full range might be fun. To have a facet big enough to mount a four inch the inside edge (using 1/2 inch material) is 8.5 inches giving a volume of 0.7753665740741 cubic feet.
(I love to leave all those digits as if my measuring devices are N.A.S.A approved and calibrated.)



Any thoughts??

Jeremy

I'm building three line arrays with 6x GRS clones of the BG NEO 8 per speaker. I bought the Joseph Crowe Neo3 tweeter horn CAD file and elongated the design to fit the NEO 8. I then 3D printed the design in two different widths. There has been some discussion that blocking the 2 outermost slots in the NEO3/NEO8 speakers improves off-axis dispersion, so one of the designs only allows the center two of the four slots of the speaker to be exposed. I also tested two different sizes of rear chambers. The idea for my line array is to make very large (but shallow) speakers that will fit behind a 100" acoustically transparent projector screen. My modifications of Joseph's original design eliminates the nice round overs that curve to the back, so that the horns would fit flush in a large infinite baffle style speaker.

Here are all of the measurements together.



The complete Line Array would be 6x NEO8s stacked like below.


Here's the two designs, One Narrow/2-Slots Wide, one normal width/4-Slots.



These were rather rough prints with 1.0 x 0.3 mm layers. I printed in ABS because it sands and glues well. I spent minimal effort on sanding.



Here's the flat baffle I tested along with the large rear chamber (Old 6x9 Car audio box), rear chamber (green 3D print), and the two horns (2 and 4 slot).



The small chamber had a weird resonance around 2.4k The difference in level may not be real, as I forgot to test the small chamber, and had to re-set everything up to get the measurement. All of the other measurements below were done with the large chamber.

Small (blue) vs Large (red) chamber:

Large Chamber Distortion:



Small Chamber Distortion:



The flat baffle with flush-mounted GRS NEO8:



Straight-on measurement (0 degrees) of the flat baffle (red), narrow horn (green), wide/normal horn (purple).



15 Degrees. flat baffle (green), narrow horn (orange), wide/normal horn (red).


30 degrees:
Interesting that the narrow slot horn did help off-axis dispersion, but only above 16k.


45 degrees:


Flat baffle (no horn) 0, 15, 30, 45 degrees


Narrow Slot Horn at 0, 15, 30, 45 degrees


Normal Width Slot Horn at 0, 15, 30, 45 degrees


Final thoughts:

The normal-width horn did not have a very smooth output which was surprising, but it did have the same increase in output from 1k to 3k as the narrow horn.

I was also surprised that the off-axis levels diverged lower (1k vs 4k) with the horns.

Interesting that the narrow slot horn did help off-axis dispersion, but only above 16k; also interesting that the narrow slot was actually the loudest everywhere.

I don't think that I'm going to end up using the horns in my line array. The more gently upsloping response of the flat baffle should play well with low end lift that you get from a line array. And the lack of the sudden increase in response around 1k (that the horns have) should allow me to cross over the flat baffle mounted speakers lower (500-700 Hz).

Joseph already posted about a horn he actually made specifically for the NEO8. His wooden horn was much larger, and helps extend the usefull range. My 3D printed horn raised output down to ~1200 Hz, but Josephs's larger horn increased output down to 300 Hz! He posted about it here: https://www.diyaudio.com/community/threads/grs-pt6825-8-planar-test-results-300hz-horn.379684/

The fact that a horn-loaded NEO 8 can play down to 300 HZ is amazing. I'm convinced now that there is really no point to the NEO3. The NEO8 plays the upper range just as well, but allows a 2-way speaker with a lower crossover and probably more headroom as well.

Would anybody else choose one of the horns over the flat baffle in a line array? What do you guys think? Big thank you to Joseph Crowe for making his CAD files available, makes this silliness possible.

The audiophile reissue label IMPEX (essentially, a major player in the same marketplace as Mobile Fidelity) last month reissued on Crystal Disc CD the David Hancock-engineered former North Star, former MFSL, former JMR/John Marks Records title "Songs My Mother Taught Me/Romantic Music for Violin and Piano," with violinist Arturo Delmoni and pianist Meg Bachman Vas.

This CD starts with an optical-glass substrate. (That makes for a very heavy CD.) The pits are embedded in a special plastic that is cured under pressure and light (I assume ultraviolet light), and the reflective layer is gold. The liner-notes booklet was signed and numbered by Arturo Delmoni. The packaging is beyond deluxe. The Crystal Discs are made one at a time; each one takes hours, I am told.

The target market is well-heeled audiophiles in Asia, where Dr. Delmoni is something of a cultural demi-god. People put up YouTubes of their stereos playing the North Star LP of what I call "SMMTM" (Songs My Mother Taught Me) just to show off that they own the North Star LP.

North Star LPs have gotten up as high as $400 on eBay. Verified sales, not asking prices.

The CD data on this release is from the Bob Ludwig 1993 digital transfer directly from from David Hancock's 30-ips, half-inch two-track original, in-machine, at-session tapes. David had a unique asymmetrical combination Noise Reduction and Playback EQ encode/decode. Bob Ludwig is the keeper of those keys. Arturo and I attended that session.

To me, the Crystal Disc has more detail and more body. I do plan to send an aluminum JMR CD and the Crystal Disc to Bob Ludwig for his evaluation.

The limited edition was pre-sold about 80%; I don't know if there are any left.

I am not posting this to brag. I think that what is remarkable is that a young recent law-school graduate with no real direct relevant experience managed to put the moving parts together for a shall we say, near-totally underappreciated young violinist to make a recording. 40 years ago.

Thank God money was so tight that we could not afford digital!!

If SMMTM had been recorded in 1982-vintage de-facto 14-bit digital resolution, nobody would care at all today.

In early May of 1982, I never would have dreamed that 40 years later, people on the other side of the world would pay so much money to get just that last little bit closer to one precious moment in time.

Which, thanks to David Hancock, Bob Ludwig, and Abey Fonn at IMPEX, remains ever frozen in amber.

https://www-joyaudio-com-tw.transla...=zh-TW&_x_tr_tl=en&_x_tr_hl=en&_x_tr_pto=wapp

If anyone wants more background, my memoir about Herb Belkin of MFSL tells the story.

https://positive-feedback.com/audio-discourse/mfsl-herb-belkin/

amb,

john

Pair of new, but tested Dayton Audio DC160 "Classic" 6.5" woofers. Very nice, smooth sounding woofer with great bass response for their size. Have used these in Paul Carmody's "Classix II" DIY speakers and love them. Paul has also designed a TMM which uses two of the DC160s and the Vifa BC25TG tweeter.

Asking A$90.00 for the pair, including standard tracked postage within Australia; also have a used pair, taken from my Classix II, work fine but some cosmetic flaws in the surround and rear gaskets; asking $60.00 including standard tracked post within Australia.

The DC160 is also used in other DIY speakers such as Dayton BR-1 kit with the Dayton DC28F tweeter and Dennis Murphy's "Affordable Accuracy Monitor", which is a BR-1 with revised and better crossover. Or the "Dayton III" MTM with the DC28F. Not the most attractive looking woofer, but that's what speaker grilles are for!

Reason for sale: too many speakers!

Links:

http://www.theloudspeakerkit.com/dayton-audio-dc160-8-6-1-2-classic-woofer

https://sites.google.com/site/undefinition/floorstanding-speakers/classix2-5

https://sites.google.com/site/undefinition/bookshelf-speakers/classix-ii

http://murphyblaster.com/content.php?f=pe_br1.html

https://www.parts-express.com/Dayton-Audio-BR-1-6-1-2-2-Way-Bookshelf-Monitor-Speaker-Kit-300-640

Thank you for looking!

Geoff

I, usually, prefer to post after a given project has been made and tested, but, this case is different.

I have been requested to make a DC to DC converter. There are many IC's and modules online. I have found this : https://www.amazon.ca/Converter-0-5V-30V-Powering-Speaker-Amplifier/dp/B08KZYSRRG

This can be used without any, additional circuitry, yet, I have decided to make something, which is posted here : https://drive.google.com/drive/folders/1LME3Dg6xhWQB1pt2SPSd0jWdyPnyTgpQ

Please, comment, suggest, correct, inform, etcetera.

TWEETERS ARE SOLD EV T-35A tweeters $90 per pair plus shipping. Taken from my home stereo. Resistance measurements in pictures. I have 2 pairs, and they are very similar. Take both pair for $150. Shipping will be 10x8x6 inches and 4 pounds, shipped from 28601.

MIDS STILL AVAILABLE EV 1824M mid driver with 8hd horns. One pair (A and B) are in excellent condition for $200 per pair plus shipping the other pair (C and D) have small toddler finger indentations on the screen but work and sound fine, $150. Taken from my home stereo. Resistance measurements in pictures. I have 2 pairs, and they are very similar. Take both pair for $300. Shipping will be 16x12x12 and 20 pounds per pair, shipped from 28601.

View attachment 1010061View attachment 1010062View attachment 1010063

I've been playing around without serious building.
I've been using a Behringer CX crossover for approximations and an old Rotel multi-channel amp.
The amp is OK but being an older amp [ the RMB 1048] the power is limited.
I've managed to put a little money aside and I'm thinking about getting something with a little more Oooomph so I can use it double duty as my party amp as the old ones are slowly losing power and I can't be bothered getting them rebuilt [ I'll be able to sell them tho as "Vintage amplifiers] so looking at second hand multi channel amps which units do people think offer the best bang-for-buck?

I'd like to opt for multi channel amps here to keep the space used for power to a minimum.
I'd like to spend less than $1k- if I can.

Hello everybody, newbie here:

I'm looking for a pair of bookshelf speakers for my PC (not for monitoring, just play games and listen to music), something better than the typical 2.1 Logitech kit, but in a 2.0 fashion trying not to miss the bass so much.

Searching on internet the DIY project "DINAS - Do I need a sub" was pretty promising:

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• Amp: Lepai LP210PA
• Subwoofer: Tang Band W5-1138SMF
• Woofer/mids: Dayton Audio RS100-4
• Tweeter: tn28b

Until I came across these comments about the amp:

• "The Lepai LP210PA is based on TPA3118 chips without heatsinks so don't use with 4ohm loads! use TPA3116 instead"
• "The power supply that usually comes in the kit is completely insufficient to give full power"

I'm not sure about the TPA3116 as alternative due I'm currently using one on a Bluetooth chinese board from Aliexpress and it has noticeable background noise and a 'hiss' noise when a bluetooth device is connected (maybe is another component and not the amp itself):



On top of that i'm from EU and the Lepai LP210PA ($63 on Parts Express) costs 105€ at soundimports.eu. Almost double the price 😵


So my questions are:

• Suggestions?
• Alternative /better components?
• Is there a more convenient way to sound my PC instead of buying 2 amps + 2 good power supply + DAC with volume knob? I've seen some monitors that have all integrated in a single plate:

Thanks!

Hello everyone. I’ve been a long time audio enthusiast and have finally found time to upgrade it to a hobby. I’ve been a toolmaker for 30+ years now so tinkering with things comes east, but I have a big learning curve ahead of me when it comes to the electric side of things.

My DAC just burned the transformer and it looks like a custom made one with five secondary windings. No doubt replacement won't be found on normal RS/Element14 websites. Anyone has a clue to find replacement? Please see the attached photo.

New but tested pair of these well known tweeters, asking A$50 including standard tracked post within Australia.

Product link:

https://www.wagneronline.com.au/tym...udio-speakers-pa/bc25sg18-04-69688/976499/pd/

The BC25TG family is used in many DIY speaker projects. Have used these in Paul Carmody's Classix II with the Dayton DC160; and in two way speakers with Peerless 830656 and SB Acoustics SB16PFC woofers. Really nice sounding tweeters.

Does not include the mounting screws, as have used in another project.

Reason for sale: too many speakers!

Thank you for looking

Geoff

I know this is an odd request. So, I'm putting it in the Full-range section where oddball speakers are the name of the game.

I'm replacing the incredibly cheap factory 3.5" speakers that came preinstalled in the top of my enclosed shower unit.
The back of the speaker is infinite baffle, using my entire bathroom as its enclosure.

While the front of the speaker fires into a medium-closet sized shower enclosure.

I know I won't get any bass. But can you think of any drivers that would excel more than something designed for a tiny speaker box and would work from I guess, 120hz-12Khz (or better)?

I'll need something that isn't un-treated paper.
Because it won't get wet but there will be lots of moisture in the air.

Original speakers:


Per my digital calipers, the cutout diameter is a tad over 3.3".

I can’t tell if these transformers would be useful for audio builds.

Any help???

Hi,
To make a long story short, I'm currently in i a situation where I have plenty of tubes and output transformers to play around with but very few mains transformers (and no budget to by new ones everytime I get another idea). Many of my tubes are european sweep tubes with from the P series with 300mA heaters, which allows a bit of creativite thinking when it comes to warming them up. I also have a decent stash of medium to large mains toroids with in the 2*25-40V range and dozens of 160V 680uF Rifa electrolytics.

My latest idea is to use a 77V 5,5A (single secondary winding) 400VA toroid to produce +100Vdc @ 300mA to heat a quad of PL36 (25E5) while simultaneously producing 200Vdc @ 250mA or so for B+. The 4X8-type voltage doubler looks right for the job, but are there any hidden pitfalls to look out for when using this schematic at higher currents?
My instinct tells I would need diodes that can take a bit of flogging.

http://www.bunkerofdoom.com/lit/4x8/index.html

Another option would be to use a regular voltage doubler to produce a single 200V rail and use a quad of PL519 (40KG6), although such an amp would rely heavily on the Vkf ratings of the tubes and not go unnoticed on the electricity bill.

It started here : https://www.diyaudio.com/community/threads/scryer-or-how-f8-met-mighty-sissysit.388550/

After that one, couldn't avoid my own greediness to try this combo too

Though, clever as I am, God's ways are his own and Buddha is everywhere........ meaning - better name for this one would be Mad Hatter, you'll see from 4R load behavior ........

Ok, for starters usual set of graphic files

Schematic is aggregate, find separate ones for pure Mos and for Schade later

Hi everyone, this is my first posting on this site and was basically wondering if anyone has had any experience with the 'hegel sound engine' from http://www.profusionplc.com

from the pdf, the device looks pretty simple to build and has some good specs. check it out, what do you all think?

Hi,

using a biamped system with set.amps on the fullrange panel i want to make the basspart switchable to fullrange
bypassing the resistor crossover after the transformer
If that works i can run my system all day (maybe with compromised sq) and not only 2 hours in the evening.
Are there issues regarding voltage if i use 230V rated switches? As often Wago-clamps are used i think that should be ok
soundwise

have a nice day
Carsten

I mounted my Emilar EH 330 - EC 600 today and made a "Kenrick video" for you to enjoy! 🙂

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Several weeks ago I started to write code for implementing FIR filters under LADSPA, and mentioned it HERE.

LADSPA plugins were never intended to do intensive processing, or to have highly variable CPU loads between calls to the plugin by the LADSPA host. This means that implementing an FIR convolution inside the plugin itself is probably not advisable, or even possible. So I have taken a different approach:

The FIR convolution is performed in a separate process that is entirely independent of, and is running asynchronously from, the LADSPA plugin and host. I call it an "FIR engine". This separate process is launched using an operating system call as part of the LADSPA plugin setup process. Parameters and other info are passed from the plugin to the FIR engine using a "control" FIFO, and other FIFOs are used to pass unprocessed and processed data.

Performing the FIR convolution in a separate process has several advantages. The LADSPA plugin, once up and running, only needs to put/get data into/out of the FIFOs and this keeps it very computationally lightweight. The FIR convolution can only be performed when the desired length of data has been obtained via LADSPA. Because of the real-time nature of the audio processing, this takes Ndata/sample_rate seconds. For example, at 48kHz if you process 16384 data points per call it will require the FIR engine to collect incoming data for about 0.35 seconds before it can perform one convolution. During that time, the LADSPA plugin might have been called 16 times (given a typical frame of 1024 samples). The speed requirement of the convolution can therefore be as slow as 0.3 seconds or so, by which time the next 16384 samples have been collected and need to be processed. This can be helpful when using low powered Linux hardware such as a Raspberry Pi. Additionally, the OS is free to schedule the FIR engine process around other processes, and there is no need to run it at a high priority, etc.

As part of the set of of the plugin and FIR engine, a uniquely named directory is created in the tmpfs (files in memory) that comes with all Linuxes. The FIR engine collects and writes to this directory a file that lists the mean and longest "cycle time", that is the time to process the data and return it to the LADSPA plugin. The cycle time latency can be obtained in a test run for a given platform and FIR filter set, and then this is supplied to the LADSPA plugin as a parameter during normal use. This latency timing info is used to set up internal buffers so that underruns are prevented.

Because the FIR engine is a separate and independent process and because some LADSPA hosts do not correctly tear down the plugin (by calling deactivate, etc.) I have written the FIR engine to self-terminate and cleanup after itself. This includes deleting the tmpfs directory in which it was operating, and the FIFOs. Once this process is completed there is no sign that the FIR engine was ever there. A pair of error logs are written to the tmpfs and not deleted, however, on reboot the tmpfs filespace is wiped clean. The user can manually delete the error log files anytime. Since these only are used to record fatal error messages, it is not likely they will grow to any appreciable size.

I am currently coding up the FIR convolution using FFTW but have everything else functioning well using a dummy convolution function that simply passes input to output in the FIR engine. I hope to get a test version fully up and running in a week or so and will post updates as I have them.

Hi

Apologies for yet another noobie-question, as I have no knowledge of all this but am curious…

I will build a 3w speaker from a kit in the near future. This will be my first ever.
The designer wasn’t too specific about the cabinet except for the front-baffle and the volume.
While contemplating various details, the question came up wether changing the cabinet‘s geometry (making it „conical and a little deeper) without changing the volume or front-baffle would affect the sound?

I wanted to try Miro's AD1862 DAC. Apparently the only legitimate source fo thse chips is Digi-Key and I had to purchase 14 of the DAC chips. I've already sold some of these but do have 4 chips left. I paid $25 each for the chips and will sell them for the same amount. I also had 10 of Miro's fine PCBs made by JLCPCB. I will give these for free with the chips. Please send me a PM if interested.
Cheers.

We used a lot of these in the 80's and 90's, I believe they were sold as Digikey's "house brand". Decent quality metal films with copper leads. Anyone know who the actual manufacturer was?

I just got a spare Denafrips ARES II board. I bought it used ("in working condition") but I doubt it is the case. The initial plan was to use it in a DIY DAC perhaps with a lithium battery power supply. But first I need to power-up this board, repair it and test it. Can you help me with that ?

I am looking for information about the original transformer and how it is wire on the board (PIN 1 to 8) ?



Here is the top of the board:



I already see two problems on the board :

Hi folks, this is a strange problem I've never come across before. I stuffed a YJ tpa3116 ( black/blue) board in an enclosure with a 2 way input selector switch and a 10k tocos cosmos pot. I've used this setup before without problems but this time I'm getting quite a loud hiss from both channels that isn't coming from a source and dies away to almost nothing when the pot approaches max (nothing playing) in the max position the hiss can only be heard with my ear to the speakers but at normal listening positions I hear the hiss from my chair across the room. The same pot worked fine previously with a buffer so I'm kind of confused why it's not ok with the the amp board. I have a 20k miniature stepped attenuator I could try but I can't see how that would make any difference. Any ideas would be good to hear as this is both annoying and confusing.


Cheers D

Hi,
I had to sell DBX 234 XL 100% working conditions,use only a few hour only ,
They are not no more using after brought fusion 503 .
Price $50 plus world wide shipping

Best regards
Myint

Hello, I have a heavy tonearm - I want to buy 100-200$ a good MM cartridge with low compliance! I ask for advice
No more than 10

Big Vent Relex for the Mark Audio Alpair 12pw. Comment from builder (he may chip in)

I can confirm that they sound absolutely beautiful. Very very good.

Click to expand...

Contach me or Scott about getting acess to the full planset.

dave

After looking at all the existing options I can find for adding an audio ADC to a RPi, I've decided to go ahead with making (or at least, trying to make) a RPi hat based around the ES982x. It will be pretty similar to the datasheet circuit, using ESS's own ES9311 regulator to power the ADC chip. The ESS datasheets use OPA1612 op-amps for the input stage, and I allowed for them, however the first prototypes will use ES9820 + NE5532. The main design requirements are:

1. Single-ended connectors onboard, with headers for diff. input
2. Use I2S, not USB
3. Hopefully >100dB THD+N - significantly better than other options available
4. Parts available (not that easy)

It's really for people who want a high(er)-quality I2S input to their Pi, because there are high-quality USB ADCs (Cosmos) and ~90dB THD+N hats available (e.g. HiFiBerry ADC+DAC and this open source WM8731-based one). I also think that a Linux driver for the ESS chips could be useful for someone else.

There will be a few parts of this project that are new for me (especially the driver). Here is the first version of the schematic... I welcome feedback and suggestions. For example, one thing I'm not sure about is whether the resistor values can be left the same when substituting the NE5532 instead of OPA1612.

FYI, my own use-case is to provide analog input for a RPi DSP+xover for 3-way active speakers. Analog input selection is done on a relay board inside the case and the RPi USB ports will be exposed through the back panel, so a USB interface is not gonna work.

Hi. I am interested in a pair of Audax PR120IL Horn tweeters. Please let me know if you have an unloved pair available. Thanks!

Looking for matched thermaltrack trannies for my FC 100 Roender amp .
Or if someone has a complete kit of this amp and want to sell it let me know .
Thanks

Patrick

My background is in music production so I know very well all of the commercially available mixing / mastering monitors. I want to explore building my own.

From the monitors I have used I would like to take the general characteristics of:

- the highend from Genelec S30D ribbon tweeters

- the midrange from ATC scm200 soft domes

- the Lowend from Barefoot MM27 but with extended response

For my purpose I would like to be able to go to fairly high SPL's and also frequency linearity is crucial. Budget wise I want to be uncompromising but not needing to go into extreme esoteric territory.

1. What is the most suitable Ribbon Tweeter? (Raal 140-15D / Founteck NeoPro10i / others?)

2. What is the closest available driver to the midrange dome of the scm200's? (Volt VM752 / Morel SCM634 / others?)

3. Not sure what LF drivers would complement the above but would like 2 x 15" drivers in a side firing configuration (a la barefoot MM27) (Volt RV3863 / others?)

Would appreciate thoughts on this configuration with professional audio mastering in mind.

Currently designing a new studio so there are no space limitations - the studio will be big enough to accomodate any design.

Current thinking is to use Hypex amps / power supplies / DSP. But also open to other suitable pairings for the above.

Most likely closed cabinet and not ported design but this also not finalised.

Thanks in advance for any advice for this novice !

I have a pair of Metaxas Solitaire Amplifiers that drive my Infinity Kappa 9 speakers
The amps were sent to a reputable Perth electronic service outfit ,but were deemed unserviceable by the technician.
The amps in question were revealed on the techs' Hall of Shame
https://liquidaudio.com.au/hall-of-shame/
So I have decided to rebuild myself , but the question is - Is it a Bridge too Far for a novice to undertake?
I have plenty of time at my disposal as I'm retired .
Any comments would be gratefully received.
Thank You.

The analog level meter has been designed for use as a recording level meter for a reel to reel tape recorder. But it can also be used as a standalone device. The LED bar can contain up to 50 elements per channel. To reduce the level of interference, static indication is used. A feature of the meter is fully digital signal processing. That allows you to make the meter metrologically accurate, with an exact match of the dynamic characteristics. All parameters are programmed using special software on a computer.

Each meter channel contains two signal processing branches, the readings of the first one are displayed as a bar, the second - as a dot. This can be used to display two values, such as average + quasi-peak, or quasi-peak + true peak. For each of the branches, you can individually set the gain, integration time, response time, decay time, hold time, fall time, peak hold time, peak fall time. The scale type is set by assigning an individual threshold level to each LED with an accuracy of 0.01 dB. This allows you to build a linear, logarithmic, S-shaped with a stretch near 0 dB and other scales. A set of all parameters, including the scale type, is stored in EEPROM as a preset. There can be 4 different presets in total. They can be selected with the control button. The second button can turn on the indication of statistics of maximum levels, as well as reset it.

Here we will talk about how simple things can be made difficult 🙂

Block diagram

The block diagram of the level meter is shown in the figure below. This diagram shows one stereo channel of the meter. The input signal is fed to a differential amplifier implemented on an op-amp (Meter50_v4_sch.pdf). It performs two tasks: it shifts the signal by half the ADC scale and allows you to implement a balanced input. The signal shift is necessary because the built-in ADC is unipolar, its input range is from 0 to the reference voltage. A balanced input is desirable even if the signal source is single ended. It allows you to take the signal relative to "analogue" ground and eliminate the interference present on the ground of the meter circuit. If the level meter is made as a separate device, then balanced inputs with XLR connectors can be implemented.



All signal processing is done digitally by the STM32F100/103 microcontroller. To avoid the use of an anti-alias filter at the ADC input, as well as to be able to register short signal peaks, the sampling frequency was chosen quite high - 96 kHz. With DMA, ADC samples are stored in a buffer in RAM. When half of the array is ready, an interrupt occurs, the data is read and processed.

DC removal filter

The first operation on digital data is to remove the DC component from the signal. This must be done before the signal is sent to the detector. With a meter dynamic range of about 60 dB, even such a small DC component as 0.1% of the ADC scale can be equal in magnitude to the useful signal and distort it.


Detector

Next, a full-wave rectification of the signal is performed. This is the easiest part of processing. When the DC component is removed from the signal, the task is to calculate the absolute value.

Filters

Next are several filters that define the dynamic characteristics of the meter.


Integration time

To obtain a given integration time, the detector output must be smoothed with a filter. The smoothing filter is not a simple low-pass filter, but a more complex filter with different charge and discharge time constants. The charge must be fast to provide the specified meter integration time. And the discharge must occur slowly to provide a given return time, which is much longer. Therefore, the integration time filter is turned on only when the signal rises; when the signal decreases, it is kept at the same level. A separate filter is used for signal decay.

The difference filter equation has the following form:

y( n ) = y(n-1) + A*(x( n ) - y(n-1)), where A = 0.0083 for Fs = 96 kHz

A 32-bit accumulator is used to obtain the required calculation accuracy. Filter coefficients are 16-bit unsigned. They are defined as A = 65536 / 96000 / tau.