I've built a number of amps over the years based on David Tilbrook's AEM6000. They've all been really good, reliable, low distortion, quiet amps. With the newer ones I've started departing a little from David's topology, initially by dispensing with one side of the VAS, which I built as a set of 50W modules that work very nicely, and now by realising that by dispensing with one side of the VAS, I can use current-mirror loads for the preceding stage, increasing the load impedance for that stage significantly and allowing for significantly better distortion performance.

I started discussing this in this thread http://www.diyaudio.com/forums/solid-state/288479-aem-6000-ready-build.html. Chalky pointed out that my VAS current wasn't well defined, being set from one side of a current mirror, so I've been working since then to get a handle on how to stabilise the VAS current without throwing away the gain. This initially involved using current sources and sinks tied back to the previous stage's tails, and more recently using a resistor across the mirror to define the collector voltage on the mirrored side at the cost of some gain.

While at it I changed out some of the VAS and preceding stage transistors, using the KSA1381/KSC3503 for the VAS, and IMX8/IMT4 duals for the preceding stages. Tail currents are provided by the cheap and cheerful MJE340/350.

I think I've reached a reasonable compromise with this, and I'm working now on layouts.

Here's the simulation schematic:



My goal with this is 1ppm THD at 1KHz/50W/8R, without huge supplies. I've set a somewhat arbitrary limit of 100mA bias for the LatFETs.

I've achieved this and then some. With gain set to 20dB, THD at 1KHz simulates at 0.4ppm, at 10KHz I achieve 3.5ppm, and even at 100KHz THD is a manageable 0.04%.

So, an explanation of how things work. The first stage is a completely conventional matched JFET (SST404), with cascode provided by an IMX8 dual and resistor loads. Tail current is set to 5mA, of which 3.5mA is used to bias the cascode and the rest flows through the JFETs.

The second stage is where I depart a little from David's topology. It's a symmetrical differential pair. Rather than the resistor loads of the AEM6000, I use a pair of current mirrors. Tail current for the differential pairs is set to a healthy 9mA. Dissipation is low enough that I can continue to use the IMX8/IMT4 duals both for the differential pairs and the current mirror loads, which is nice, as I really need a good match here, not to mention that these little things are lovely and fast, with plenty of gain at just the right current. I just use MJE340/350s for the tails as I don't need speed here but do need some dissipation.

Note R30 & R31. These 47K resistors waste some gain in this stage to ensure the collector voltages of Q17 & Q18 reasonably closely follow their mirrors. This allows us to define the VAS current by noting that the collector voltage here is defined by one Vbe + the current flowing through R18-22.

The current in the VAS is set by two mechanisms. The first is that for the VAS transistors (KSA1381/KSC3503 - Q14 & Q15). This is set by their emitter resistors. They have constant current loads (Q19 & Q20), whose currents are set by their emitter resistors, recycling the reference from the preceding stage tails.

The loads must be set at a lower current than the VAS transistors, to ensure current flows through R28, and hence the bias for the final stage drivers is set. I chucked a diode in here (thanks PB2), because if the VAS transistors and loads aren't biased correctly there was otherwise nothing to stop current from flowing the wrong way through R28.

I run the VAS transistors at 10mA, and the loads at a tad over 7mA, leaving the balance of a little over 2mA flowing through R28. This is then tweaked to give me 100mA driver bias.

Compensation is straightforward. A couple of caps across the VAS transistors, some RC across the first stage diffamp, and normal phase-lead compensation at the feedback point. I can push the gain all the way down to 3 (1ppm at 10KHz!) without it taking off, so I'm confident that it can be reasonably easily stabilised at more usual voltage gain levels.

Anyway, I think it's cool. I'm gonna build a couple, and I'd appreciate critique, especially of my logic around the VAS current stabilisation mechanism, as this is the area where I'm least confident.

Like so many projects these didnt see the light of day and havent for a few years. The guy from Adio-gd set these up but they are adjustable. Please dont ask how because I have no clue and would of forgot by now anyway! I used to do a lot of work on TDA players and build my own supplies for all the crucial parts but for some reason I purchased these, must of been for a special favorite. Disclaimer. I have never put power to these. The solder on the inputs is from him setting current. I know these arent going to apeal to anyone but people like me so hopefully if they sell you can check the voltage is correct by looking at resister values, my days of DIY are long over.

£20 plus post to any country for the lot.

Edit: I found the label in the box, the missing one is +5v 100ma

Hi, I'm from Worcester UK.
I'm into bang and Olufsen or anything really.

Hi there all,

I own sinds many years, a Cambridge Azur 840C Cd Player/DAC , which has shown quite some problems, mostly due to poor quality Chines Electrolytic capacitors (the problems All being located at the power lines of the DAC's (AD155 's) the Op AMP's and the Muting circuit.
The digital sections have so far worked flawlessly.

I would like to address this problem of poor electrolytic capacitors in All powerrails once and for all, instead of having to repair it about every 3 years.
(Once it works, it sounds very nice, hence I'm still holding on to it).

MY question:

There are a total of 21! voltage regulators (Type LM78xx, LM79XX, LT1761, LT1763, LM1117, each with a 10uF 35V capacitor at the Input and Output.

These 10 uF 50V capacitors have a short lifetime expectation due to poor quality, but also because they are mounted immediately next to hot heat sinks (a clear design error) and also due to poor ventilation possibility (due a suboptimal enclosure design), resulting into high operating temperatures inside the enclosure.


Better quality capacitors (e.g Rubycon 105 deg. C) typically behave better, but I am contemplating to cure this problem Once and for All, by using film capacitors e.g. 2.2 uF or 3.3. uF 50V WIMA MKS Polystyrene capacitors.: MKS2B043301H00KSSD WIMA - Capacitor: polyester | 3.3uF; 30VAC; 50VDC; Pitch: 5mm; +-10%; MKS2-3.3U/50 | TME - Electronic components


In view of space limitations, I cannot use higher values (I already have to install them at the bottom side of the board).

Could you please confirm whether such a lower value (3.3 uF film capacitor instead of a 10 uF electrolytic could work ( I have read somewhere that they are only used to suppress possible oscillations of the voltage regulator).

Please note that the 5 main power supply rails are each buffered with 4400 uF 50v (each 2x 2200 uF capacitors).

Appreciate your opinion / feedback!

Hi,
I bought these pair of Japan made surplus speakers. I intend to use it for my DIY amp. However I didn't realize that it comes with 2 pairs of wires. Is there a way to convert them for single wire use? something like parallel or maybe series parallel combination.
Thanks!

New member here.

Background in RF Comms mostly and many years in broadcasting. Now retired.

🙂

This is a summary of my approach to designing discrete-transistor op-amps for audio. These op-amps favor open-loop linearity, bandwidth, and PSRR over gain.

IC op-amps are invariably designed for the highest possible open-loop gain (>=100dB) so that the closed-loop gain is almost entirely determined by the feedback network. IC op-amps are well-suited for precision applications up to frequencies of about 100Hz. Above that, the gain must be rolled-off and performance deteriorates.

In addition to audio being fast by op-amp standards, the feedback networks used in audio are always linear and often time-invariant. Audio can tolerate errors of 1% as long as the errors are linear and time-invariant.

The special properties of audio give rise to a new (old) approach - design an amplifier with good open-loop linearity, bandwidth, and PSRR, and apply moderate feedback. An amplifier with 60dB of open-loop gain can come out ahead of an amplifier with 100dB of open-loop gain if the starting point is 40dB better. Keep in mind that we are not trying to beat IC op-amps on all metrics, just the ones that matter.

Naturally, I have omitted all details. This approach is applicable to class A amplifiers where distortion can be made arbitrarily low through an appropriate choice of operating point.
Ed

ETA: I added a graph in post 63 which illustrates the idea.

Humm, to be precise, I am not a professional audio development engineer, but an amateur-I saw more strange ideas in DiyAudio forum than in China, and I thought it worthwhile to discuss these ideas, so I participated in it.

For amateur purposes, our production may be irrational, non-cost-optimized and incomprehensible. PCB is my drawing board-I hope to have a good communication result on a compatible platform.

Hi, I'm currently servicing a PA amp, K.C.S PS-900 (it is a DAS P-900 on the inside, just rebranded outside).

EDIT after running several tests:
The relays don't click. There is one in the psu and one for each channel.
The PSU one tries to flick, there is a tiny movement but not enough to close the contact.
The ones on the output have no movement at all

If I disconnect the PT (about 100V AC) from the rectifier/ relay board all relays work.

If I leave it connected and manually close the output relays they stay closed and I have sound...so I don't know what I'm supposed to look for.
It also had a hum on the output once relays were closed but that is pretty irrelevant for the moment I think.

I replaced the amp board relay driver transistors and electrolytics on one board as it was cheap, i also checked the power transistors just to be sure. Amp side should be fine. And since it is a problem that interests both channels its definitely something happening above.

My tape deck - same model as to find under
Le forum de Vintage Audio Laser • Afficher le sujet - Remise en état d'une platine cassette Yamaha K-720
produce sometimes unwanted stop while "play" mode (resp. change the direction after activiating the auto reverse function).
Fast-forward and fast-rewind are fine.
Take-up reel pulses while in playback are present.
This issue is mostly to observe, if the cassette tape goes a bit too stiff (C90, C120)
A second tape recorder device, which is the same model, shows the same unwanted effect. By the use of C 60 cassette tapes the effect is much less frequently.
Those reel, that winds up, remains through stopping reel motor. The other reel, driven by the active capstan, unwinds for a second longer, and then comes the stop-mode or the reverse mode.
The behaviour is the same as to observe, when the cassette to an end is, i. e. the tape is completely unwound at one side.

The question is, which current flow goes in the reel motor in good working condition (the maximum permissible value before the reel motor driver goes shut down and stop the motor).
If I know this value, it is possible to determine, whether the electronic components or the motor itself is the reason for this unwanted effect.
Thanks for your help in advance.
Maybe it is possible to upload a schematic diagram.

Discovered that the Linaeum dipoles from some Optimus LX5 nest nicely on the top of the ADC MS-650. I’ve been listening to them and have been pleasantly surprised how well they add airy-ness. I’ve crossed them using a 6.8 uF cap at about 3,900hz which is a little lower than what they were originally (6.2 uF).
What have others done with the Linaeums? Any pointers?

I thought I'd share this article about Ulrich Sourisseau, a maker of DIY vinyl cutting lathes.
It’s a great addition to the big commercial record companies where small bands sometimes have to wait a year to get a handful of records pressed.

https://www.theguardian.com/music/2025/feb/25/vinyl-carver-lathe-cutters-home-cutting-records-craze

https://www.vinylrecorder.com/

SOLD ! Sorry to all that PM me

Hello,

I have a QA402 in perfect shape and working condition I bought directly from Quantasylum for a project never completed. So I'm selling it. I am located in France so I would prefer to sell it to a European guy. Price asked 250 € + shipping.
Feel free to pm me if any question
Jean Claude

Hello diyAudio community,
Finally I registered at this site, deliberately didn't say joined as for last couple of years I read several threads here throughout as observer, so I already feel like part of it.
Particularly I liked 10F/8424 & RS225-8 FAST / WAW Ref Monitor from Xrk and DLH Amplifier: The trilogy with PLH and JLH amps from Diego. This is just to mention few of my liking, there are 100's of great posts here. The size of this community is overwhelming for me, it seems that everything is already digested, maturated and answered in every possible way . So much experience , knowledge and innovation!. Ill stay humble with my things, I know something, but by far not a master in this subject.

I started this hobby in 1980's as teenager (I'm 56 y.o. now) by dismantling parents audio devices for parts. Oh Lord how sorry I am that I destroyed radio from 50's with EL34 output stage and 3 x 8" paper cone speakers then. Radio was mono and I needed stereo, it was also only AM so no use for anyone... But amp was great, and speakers could be also if the moths didn't eat membranes while laying in garage...😱
In early 1990's I quit my university (mechanical engineering) and got job from necessity, some money allowed me to continue without scavenging for parts, this book was something available and as you can see from picture, read over and over many times..


Then I learned some high end stuff about opamps. Bought my first OP27, SSM2017, TLE2027 and NE5532. But there were few misconceptions that I learned about later; they teach about ground planes as solution for all ground issues (while its not) and encouraged my addition to tantalum condensers, which also proved wrong with time.
At that time, 1993 - 1994 , I bought KEF construction kit and built preamp, 3x24db/octave xover and 6 channels of amps (4 were standard 3 stage (bass and mid) based on MJ802/4502 transistors, and for highs I made double LTP x one more double LTP + IRF530/9530 mosfets for tweeters) . It worked!

End of 1990's and beginning of 2000's the internet became a blessing, I started to read articles from Nelson Pass and became religious , not that I am religious other ways in any way 🙂. At the same time my financial situation was great, so:

• I bought my Sonus Faber Signums, I have serial number 18. Still my reference when I want to check my sanity.
• Built fully differential preamp based on IRF610 LTP and 24pole rotary switch volume control , inspired by Nelson's Balanced Zen
• Build what I was always dreaming off; a proper turntable. here are 2 pictures, from design phase, and finished (white mass is limestone):

• Built an super quite fully balanced passive RIAA preamp to match also newly acquired Benz Micro Ruby 2 naked.... by the way this cartridge is awesome and still with me after 20+ years.
• And final effort was Nelson Pass inspired amplifier, its some 300 w fully balanced monster based on IRF610 LTP and IRF240/9240 x5 (total 40 IRF's for stereo) . Its only 2 gain stage (LTP + output) and I never seen it done like that. Here are few pictures from 2005:

This is still my reference amp, even I measured it just lately and THD is not very impressive, but slave rate definitely is.

Since then, until few years ago, audio was in big competition with life and other hobbies which are many. I was happily listening to what I have and said I will never build speakers again as its too much for me.
But since Covid, many things changed. My professional occupation (management consultant in packaging industry) slowly went down the drain, last job in 2010 gave me burnout an now I have 8 y.o. son (which is blessing), We went from city center to stand alone house on the hill and I finally have place to disassemble all my hobby boxes.
Due to those circumstances, now finally I read Douglas Self book, many articles here, lot from Elliot Sound products (great guy, sorry he argues with Nelson, both views are legitimate) , Rane and so on... and started again.
Here is latest , still in fine tuning project; 10" woofer in a reflex tube + Visaton B80 in a ball, active amplification of course;


I would like to contribute DiyAudio by one article (to open subject about what I did, with hope for improvement ideas from the group) as I cant write all at once.
What do you think would be best for start, please let me know:
1. Limestone plinth and plate turntable with super heavy bronze bearing + fully balanced MC + passive RIAA phono stage
2. Nelson Pass inspired (not his design, its mine but I had him in mind) fully differential 2 stage amp AB class
3. FAST 10" ala sonotube (but this one is from sewage pipes) + B80 widerange in a ball (made from Ikea salad bowls)
4. Rebuilding an amplifier; one PA 6 channel amp is on my desk now in process of upgrade. While it was decent product in itself, I counted over 40 changes I made. This is not about circuit diagram on a PCB, but about mechanical things, wiring mains, wiring PS, transformer, various connectors,,,, wiring ground (I said it, I see many heat about grounding here) .. So its an guide for starters.
5. Sounds as off subject, but forum is called "audio" and not "hifi"; and audio radio for elderly, visual impaired or blind people. My mother (she is 95 and last 20 years suffers from macula degeneration) inspired me to make blind friendly device to read audio books, Not an easy item to find. For other listening sourpusses I bought her 2 Tivoli radios and that does great job. But how blind person reads audio book... Is this an subject?

Pls let me know if anyone wishes me to start any of this?

Many thanks,
Dražen

I've been fooling around with my old Shure M67 four channel mixer. It sounds pretty good really. It lacks headroom and has too much gain in some circumstances like micing drums. I like the fact it plugs into the wall with a permanent power cord, has a master VU, mic input transformers, and individual master and channel volume controls. I also have a place in my heart for discrete preamps and their low noise characteristics. However, I'm thinking of gutting it and addressing the shortcomings with a new PCB of my own design. I would improve the dynamic range by using the 34volt PS that's present rather than the 14v one it uses. Also ESP has published a lovely two transistor preamp with great noise/gain/distortion figures. It requires a mic input transformer to get the gain up to reasonable levels though and to be noise free with long cord runs.
I would reuse the transformers already in the M67. I have no clue how they rate against modern types.I can easily check the bandwidth with my signal generator and O-scope. When do they saturate is another question. I suppose I could check that too by overdriving. Apparently the sonic gurus like the sound of mic input transformers for the compression affect.
On the other hand I could leave that piece of vintage gear as is and remake the whole thing using a mic input transformer available commercially.
Looking at mic input transformers I'm bowled over by the prices.
Neutrik advertises a mic input transformer, the NTE10/3. It's only ~$15.00, a far cry from the 3 to $500,00 you can pay for others.
Anyone have experience with it?
What's so special about the expensive types?

Looking to purchase a APT Holman preamp and a pair of amps in good working or restored condition. Must be willing to Ship.

Hallo,

I have a question about max SPL with bassreflex and closed boxes.

I tried to calculate a ~70L boxes with Wavecor WF223BD in BoxSim and suprised by the output.

As you can see in the images, I've added 3 such speakers in a 40cm box, one closed (red), one BR tuned to 30Hz (blue), and one tuned to 40Hz (green). In the efficiency plot, the blue plot looks good.

For the maxSPL plot, I've set the max themal power of the speakers to above 1000W, to make sure we see Xmax-limited results. Just as I'd expect, details like volume of the box drop their influence on max SPL then.

But isn't only about 97 db max SPL @50hz a suprisingly low number for a bass of that size? And how does BR manage to get ~12dB maxSPL more @fb given an Xmax limit? Wouldn't +6dB be the outcome to expect?

Hi, I'm in the process of designing a high-quality audio interface. I'm already planning on using Polak-Groner composite op-amps for the ES9039Q2M I/V converter, (possibly using lower cost OPA1612s instead of the OPA2211s and the newer OPA891s instead of the THS4031s) and was wondering what you guys thought of using the same composite op-amp buffered with a current-boosting push-pull stage as mentioned in this article as an ultra-low distortion headphone amp?

I've posted the proposed schematic below, which with 10pF of parasitic load capacitance is simulated to have a phase margin of about 40° with a 32ohm load, though that becomes a dangerously low 15° at 16ohms. With the selected transistors the amplifier will do about 24Vpp into 16ohms, which is frankly complete overkill for all but the most demanding headphones, but I wanted to design it to power pretty much anything under the sun, including planar magnetics. The distortion is simulated as -180dB, though I suspect in reality that will be significantly higher. The gain is a fixed at 3.39, mainly because I can't figure out an easy solution for digitally-controlled gain switching that doesn't significantly impact distortion.

My questions (besides thoughts about this amplifier in general) are as follows:

• Is it necessary to include gain switching with modern DACs' more advanced digital gain control and if so, is there a reliable, low-component count way to do it while maintaining these distortion figures?
• Am I absolutely insane for using composite op-amps because SNR and THD+n are limited by the DAC itself?
• Am I missing anything besides Aol/β and phase-margin for stability criteria?
• Is the maximum Vpp I have for line-level (before the headphone amp) of 8.272V way too much? I know +4dBu is 3.472V, but also heard that most pro-audio has around +10dBu of headroom.

Sorry for the long post and all the questions, thank you in advance for all your help!

Which shielded cable would you use for signals inside your amp or preamp chassis? Do you have a specific brand and model?

I'm looking for cables which are easy to work with for a semi-competent DIYer (me) and have excellent shielding.

I know that the Sound Of Cables is a big, hot topic, and I don't want to get into that debate. I will read all your opinions, and I hope you'll forgive me if I will quietly accept those which I can understand, given my (limited) knowledge. Therefore, for instance, I will select pure copper conductors over silver, in cables. I will ignore skin effect claims, and will happily work with solid-core or stranded conductors, depending on which are more easy to work with, break less easily, get soldered and desoldered more easily, etc. I will avoid cables which have capacitance or inductance which are too high for audio (do these things matter in the lengths of inches we need inside a chassis?) To give you an idea of my preferences: I blindly and happily buy cables made by Blue Jeans Cables and look no further.

I'm looking for two-core+shield type of cables, I guess. The better the shielding, the more I'd like it. Why two-core+shield and not coax? It is because I prefer to run signal ground separate from shield. I feel that shield is often more noisy than the signal ground, if I keep them separate. I have had remarkable success with long runs of unamplified mic output running through 2-core+shield cables, because I kept the signal ground separate from the shield.

Blue Jeans happily tells you which 2-core+shield cable they think are great for audio, for their XLR cables. But then, I would think those would be for external use, with thick outer jackets and mechanical protection. What version of those are available for internal wiring?

I searched for cable choices discussions on this site, and read those which I could find, but nothing is addressing the issue I'm asking. They're either asking about how to connect the shield, or getting into areas which I cannot understand.

Which cables do you use?

When I look at the Digikey catalog, I am seeing some 2-conductor cables which specify "1 pair twisted" and others which don't. I'd always pick the twisted over the untwisted -- is there any reason to do otherwise? I know twisting increases capacitance, but will it matter for lengths of a foot with circuits which are not at the hairy edge of unstable? Are the ones which don't specify "twisted" really untwisted? These are some of my doubts.

Looking to purchase a matched pair of SML BUZ900 N-channel power mosfets for a sensible price.

i recently picked up a REL Stadium III subwoofer and the PO buggered the plate amp by connecting the speaker to the wrong terminals. The P-channels tested good using a diode test on my Fluke DMV. In ideal world I'd replace them also, but looking to minimize my investment at this stage.

Thx!

I have been reading a few threads on this site regarding this topic which has answered some questions but then inevitably led to further questions. To be honest I am starting to go around in circles. To avoid the "paralysis by analysis" scenario, I thought I should try and break down my areas of concern as follows.

I need something that is easy to learn and intuitive to use, but also has most of the important functions I will require. I realise this can be a balancing act.

Most people mention Arta & REW as being excellent but then I hear other people on this forum state that they think USB mikes aren't the best. I think someone mentioned that they don't measure speaker phase? ( I could be wrong here, but I am new to this and could be mis interpreting/mis remembering things, so bear with me). I believe phase measurement is important for good imaging?

Is there a decent set up that will cover all the measurements I require including phase, that is also friendly enough to learn as a first timer?

I don't want to over complicate things but at the same time I don't want to invest time and money into something that only does part of the job.

Any help (and links) that will help me make a sensible first purchase will be appreciated. I should also perhaps mention that I am based in the UK.

Thanks

Paul

IRF044 3P5V
8 pcs. available
used; pulled from working UPS.
60V Single N-Channel Hi-Rel MOSFET resp. REPETITIVE AVALANCHE AND dv/dt RATED HEXFET TRANSISTORS THRU-HOLE

Since there's been a bit of interest in these things lately, I went and corrected a couple of small errors in the drawings, and updated the names. Just forget the old joke names, those were never intended beyond an audience of about 3-4 people... 😀 I plan to delete the old drawings to avoid confusion. They were basically draft versions anyway. This may screw up some links in the other threads.

Anyway, here are the links to the plans. They can be downloaded under 'More' on the menu.

Karlsonator "6"

Karlsonator "8"

Karlsonator

How important is it to dampen inside of the box?
What happens if we not?
Will it impact the sound?

Distance between SPDIF output and AES3 input is about 35ft/10 meters.
I'm gonna need:

• RCA to BNC cable
• NADITBNC-M (Female BNC 75 Ω input - male XLR chassis connector 110 Ω output.)
• 110 Ohm AES/EBU Cable or just regular XLR cable?

Or are there any other, better solutions for this?

Passionate about sound since always, I first went towards the sound system, and dj, which I still practice.
I had home cinema with davis speakers. notably the chatenoy model from boulanger.
then 4 or 5 years ago I was given a pair of kef 104/2. wonderful speaker.
wanting to have them restored, especially at the level of the bass speakers with their special assembly with a donut in the center
i took them to a small vintage hifi store, we listened to them with real amps, by a home cinema thing. what a discovery….we left with a preamp and an amp
today 2 years later we therefore have 1 main system with:
b&w 801 series 80 speaker recaper restored
jeff rowland coherence one preamp recaped
jeff rowland model 3 mono block amp recaped
counterpoint da 10 ultra analog dac recaped
bluesound node 2i streamer
upton audi switch, with master clock
project 6.9 vinyl turntable
secondary system quad 33 and quad 303 set
acoustic research ar38s speaker
smsl su6
another 1 with philips speaker 22rh427
amp adcom 555
receiver sansui 1000x in preamp
dac 3d lad master 500
xduo bluetooth
and then the last one
speaker b&w 802 series 80 modif at the filter level
amp yamaha pc2602
preamp hafler 101
dac audio gd reference 7
xduo bluetooth

relaxed hifi, with great open-mindedness, who likes to test, lots of stuff, with the “cables”, the switches, all the stuff that “improves” the sound to see if…..

Hi, I'm razer, a 50 years old's french guy.
I play guitar since I'm a teenager and I'm an electrotechnic teacher.
I mostly like tubes amps, as most of guitar players do, especialy for my hifi setup.
I have several guitar tube heads amps, some are very old, but I'm using less these days with the benefits of modeling stuff and D class amps associations, and because I get older and older with back issues, lightweight cabs D class powered are my choice right now, I'm building them myself.
I have audio tube amps too, a cibot 2xPP EL84 and a antique sound lab PPx6L6 (converted to 5881). I got them out of service and repair boths.
I write english like a french cow and I'm sorry about it, but it's worth trying, I'll progress !

Hi folks!,
I have a Tascam professional CD-tape combo recording deck. Yesterday, I purchased a brand called Verbatim because the store had no Maxell, TDK, or Memorex products in stock. The problem is that with the Verbatim brand CDs, there is excessive skipping on playback. Heard that there have been lots of complaints about Verbatim CD products. If any of you folks have experienced this problem, please let me know. I’d like to find out which of the other 3 brands I mentioned above fare best besides Verbatim brand CDs. Please help me out on this subject.🙏🏻

I normally build high voltage tube preamps but couldn't resist trying the China 6J1 kit even tho I didn't expect much at all. There are few versions, I got the blue one in the pic here. I must say I'm quiet impressed with the sound, it is very clean, no top end loss (that I can hear), smooths out audio little (for a single stage) and can double the output so it's more than a buffer. It is useful little preamp.
But I'm running it at 16V because that's all I had and on the back of the board is printed plate Voltage 28V. I'm reading 38V. Either they made a mistake or maybe it sounds good because it's getting more Voltage. Yes I did drop the heater Voltage resistor to 6.3. Anyone build this thing? What plate Voltage do you get at 12V and what are general thoughts on these 6J1 buffers?

based in the west of Ireland... New to the forum and diy in general... I have Lii Audio 15in open baffle and Sonido drivers in Tannoy Cabs

Program Manager: “Engineers never think the design is finished”.
(The Program Manager thinks all is done if the prototype isn’t in flames)

Hello, DIYers!

This is intended to be a series of posts about several trial batches of DIY measurement microphones that I built. I was playing with the design as a possible product for Dayton Audio. No decision was made yet on whether Dayton will actually pursue this, but they are ok with me writing about how these could be made (by some more skilled DIY constructors).

It will be partly a how-to, partly a technical design discussion, and partly a documentation about obsessive engineering behavior.

The microphone is an ultra-slim, small acoustic profile, wideband device with very flat response even without an individual correction file. A short cord connects the mic wand to its electronics board that provides power (9V battery or 48V Phantom), as well as response flattening circuitry and drive for balanced XLR and RCA outputs. The design capitalizes on the repeatability of MEMS chips’ sensitivity and frequency response.

-------
ERRATA:

• some later posts mention using 38AWG enameled wires for wiring the chip. That should be 36AWG. AWG38 would of course work electrically, but won't have much of any stiffness and so is difficult to handle even with tweezers to position onto the pads and tack-solder. A little stiffness helps a lot. Even 34AWG would probably be ok, but 36 was easier to find. For example, TheElectricGodmine.com item #G27281
  https://theelectronicgoldmine.com/p...jlFgBwuZ5ccgkjYya1pgI-1UYeRFbfSuqFWYDJH3ljWOi

[index into relevant later design posts:]

1. Frequency Response curves for mics without adjustment
2. Arguments Pro and Con
3. 3D Printed Mounting adaptor for MEMS chip
4. Equalization Method
5. Performance Specs
6. Assembly Fixture
7. Bypassing and Ground wire
8. Assembly of the Microphone Wand
9. Skill Level needed
10. How Calibration was Done on test run mics
11. Equalizer/Interface Box and related 3D printed parts
12. Files download package

STL, GERBER, Lists, schematic, notes are now available 9/13/2024

Recently retired, I now have more time to fiddle with my home theatre equipment, which has been dormant for quite some time.

Current gear:
Marantz SR6011 9.1 channel receiver,
Rauna Leira II, Front speakers
Jamo Concert Center speaker
Home build (Seas based) surround speakers
Home build sub (from now retired magazine "high fidelity") with (unknown) Peerless 15".

Greetings,

I recently picked up a sweet F20 and I'm wondering why Mitu would of decided on such a low (150mv/5k) output on the fixed outputs. I just installed it and listened to it for a week or so. And thought gee something must be off as the presentation was not well balanced. Like a fat bottom with murky mids and dampened highs. So I looked up the specs and was shocked to find such a low level. So I switched to the variable out(500mv/5k) and that completely changed thing for the better as the cloud has more or less lifted. Nevertheless both seem to be a bit low as I see 1000mv is more of the usual norm with many designs.

Which has me wondering if I shouldn't consider possibly altering the output stage to raise it. Plus I'm debating replacing the final 4.7uf coupling cap to a film type to see if that cleans things up a bit. Of perhaps I should just leave it stock.

Any thoughts would be greatly appreciated.

DD

Can it improve speaker/driver quality if I will attach addition magnet to it back?
How it will affect vintage speaker with lower magnet?

Also find it hard to remove the magnet from not needed drivers (any size) (better to buy pair).

I've lurked here off and on over the years, mostly for the tube amp content. Currently seeking assistance repairing a Technics SU-V9 that won't turn on. Will post more details in the Solid State forum.

Hello diy audio community. I currently have a Technics SU-V9 on my bench that I'm attempting to repair. I know my way around tube amps well enough, but my knowledge of anything transistor-based is sorely lacking, so I've come here seeking advice.

I believe the main problem with this amp is related to the protective circuitry which engages the main DC voltage rails. The amp powers up and doesn't blow the fuse, but neither of the internal relays click on.

I'll attach the service manual which has the schematic for reference… There are two transistors (Q601 and Q606) which I believe activate the thyristors for the DC + and - rails (D601 - D604), but I don't think they are turning on - Q601 collector and Q606 emitter both read 0V when they should be +16 and -14, respectively. Tracing things backward, Q603 and Q605 don't appear to be conducting, either. Neither does Q505, which seems to be what drives Q603 and Q605. Q505 has 0.1V on its emitter, when it should have 4.7V. Finally, some of the pins on IC501, the muting/relay drive circuit, don't show the correct voltage, but at least there's 3 volts on pin 9 (DC power to the chip itself).

And that's about as far as I've been able to get with it. Like I said, my knowledge of solid-state transistor logic is pretty weak...

Some other observations, which I'm not sure if they should be cause for alarm: the DC + and - rails after the thyristors read about +20 and -30 volts, and drift by +/- a few volts every couple seconds. There is also a very slight ticking sound coming from the power transformer, which is in sync with the fluctuations on those DC + and - lines.

The AC voltages from the power transformer secondary seem okay and are stable (I think around ~40V for one pair and ~50V for the other). And the DC + and - supplies prior to the thyristors also seem okay and stable, about +47V and -46V.

Phew, sorry for the long post. If you're still paying attention, any ideas what might be causing this behavior or what to chec
k next? TIA!

Hi,

Almost one year ago I posted information about my audio player:
http://www.diyaudio.com/forums/pc-based/273684-another-raspberry-pi-radio.html
http://www.diyaudio.com/forums/construction-tips/273690-woodware.html
Since that time I redesigned the hardware and software components. Now it's based
on Raspberry Pi 2 and Amp+. The software part was changed completely. And now
the project has its name - 'Peppy Player'.

All the details about this project can be found here:
https://github.com/project-owner/Peppy.doc/wiki

And here is the summary:

This is DIY project which includes three components: hardware, software
and woodware. All three components were created for this project from scratch.

Here are the key features of the hardware component:
* It is based on the popular single-board computer Raspberry Pi 2.
* High quality audio achieved by using integrated Amplifier module HiFiBerry Amp+ and Sony speakers.
* The Hardware has six "senses" to control its functionality:
- Mouse
- Keyboard
- Touchscreen
- Infrared Remote Control
- Rotary Encoders
- Any computer in a local network or mobile device with Web Browser

Here are the key features of the Software component:
* This is application written in Python.
* Peppy provides Graphical User Interface for audio players running in a headless mode. Currently Peppy supports 'Mpd' audio player.
* Embedded Web Server allows to control audio playback from any Web Browser.
* The default touchscreen resolution is 480*320. This is the resolution of the TFT used for this project. Though UI is dynamic and can scale to any screen resolution.
* Currently Peppy has only Internet Radio functionality. In the future releases support for playing audio files and streams will be implemented as well.
* By default Peppy has playlists containing free radio stations for English, French, German and Russian languages. Users can add their own stations to the playlists.

The key features of the Woodware component include:
* Original custom design.
* Made of solid wood (Cherry and Walnut).
* Natural finish - the variation of French polish.

I hope that the information about this project will be useful for all
DIY developers as it brings together many different aspects of developing
hardware, software and woodware fro Raspberry Pi platform.

Enjoy!

Sorry this isn't actually about a diy amp, but a triode labs. But this is probably the best forum to help me out.

My knowledge is fairly limited here. When I turn my amp on there is an intense "crackle" (the one thats kind of like radio static) that is actually causing major excursion on the driver - enough that I won't be turning it on again.

To me it sounds just like a tube that's gone out, I've heard this before, but I switched all tubes to the other channel and it's still in the same channel. That seems odd to me.

Could this be a bad transformer? Tune socket? Simple cable connection?

I'll have to bring another (cheap) speaker in to try any troubleshooting I guess, but any pointers would be appreciated.

This started as a project to simply replace the turntable... Then the amp was obviously shot and at that point the tuner was a lost cause to so it was gut and start over.

The mounting of new components is clunky for now as I work on finding someone to make me some actual mounting brackets etc.

I will post a few messages as a thread of sorts. Open to any additional suggestions or recommendations before I start wire management and buttoning it up.

Starting point:



The back off:



After a couple of removals:

Hi my aleph j Zen mod build is going not so great i put together everything but apparently i must have something wrong as i cant manage to get a click pop or anything at all to tell me im getting power to the unit i think i have the power supply connected correctly but i get no power when plugging it in

When i first opened for visual, no supply fuse., A toasted resistor divider which goes to lower 4 A6 pnp devices. Did a quick continuity check looking for CE shorts on A6s. Found 3 with BC shorts.
As show in pic, I proceeded to pull all 4 outputs. 3 were shorted as measured in circuit, the single survivor was limping along with ~ 2 mA collector leakage.Of Collector. All four will be replaced. Route cause is unknown as of now.



Pic of backside. Rated E for everyone.



Turned my attention to amp board. At first glance, I noticed the roached resistors. Both appear to be 1k ohm. 1-2W wirewound. I feel they are too closed to the board for my liking. I Will be replacing these with a higher wattage wire wound resistor and lifted further from board for better airflow. There are absolutely no markings on pcb. 🙄 A total of 18 transistors

10 x MPSA42’s——>MSP42’s
5 x MPSA92’s———>MSP92’s
2 x MPS6571’s—>ZTX694’s(1100 hfe)
1 x 2N4250—->KSC733C (ECB)

Note all are EBC orientation. I had to bend pins on 733C. Mind orientation.
Use schematic.

Diodes 1N4148x4 and 1N4007 x2

Dozen or so caps replaced. Go through all passives and verify values.

There're a few other pieces of silicon......... consisting of 6 diodes.

3 electrolytics and some film and tant capacitors.

5k bias trim pot. No trim for offset other than front end cascode diff pair and base mirror.

In this version CFE1078 a pair of moto 42/92’s is used as pre driver. Not as beefy as i would have expected.

This unit looks untouched so i suspect I'm looking at original components. Assuming this unit was used hard and has seen Venus type temperatures ….very impressed that it made it to year 2023.

The MPS6571 cascode devices and the 2n4250 concern me. I plan on testing each device. I suspect these have some high beta properties which may be difficult to match with modern replacements (IF NEEDED).

I’ll need to inspect the supporting circuits to see what else is damaged. I did notice the 47uF/50V is now open.

A long time ago, sound experts had to use their ears to do their job, which was hard. They wanted a better way to measure speakers and rooms. They could have used advanced math, but they didn't have the powerful computers that were needed back then (and still don't exist today).

In 1985, the iNTEL 80386 CPU just had just been released. Doug Rife founded DRA Labs in 1986, and released MLSSA in 1987. It was software AND hardware to go hand in hand for the PC. It came with sound card with a 12bit ADC that fit into the computer's ISA slot, and was capable of sampling in excess of 100KHz.
(IBM and compatible PCs didn't have ADCs at that time)

Around the same time a company called Audiomatica in Italy releases Clio, which it's own HR-2000 hardware that plugged into a PC's ISA slot + software to run in DOS.

These hardware/software solutions needed a 32bit PC, as well as also an optional math co-processor 80387, which costed US $800 in 1987)… $2000 in today’s money) just for MLSSA to run.

At this time computer were at their infancy, but anyone doing any real work had to opt for that math co-processor. When the 80486 CPU came in 1989, it had a built-in math co-processor, and a complete PC with monitor, keyboard and costed around US$3,000 (US$7,500 in today's money). Sounds like a lot, but it seemed like a bargain compared to the original IBM PC which has meagre performance.

Moving forward, in 2000, Angelo Farina with his AES paper, discussed the logarithmic sine sweep, aka log chirp, borrowing from radar technology, created by Sidney Darlington in 1947. It was better than the previous method… yet still people said these measurements didn't match what they actually heard.

In the early 2000s, Michael Tsiroulnikov invented a "divide and conquer" method that allowed acousticians to use better math, without spending too much money or computing resources and named it FSAF.

What is FSAF?

"Fast Sub-band Adaptive Filtering (FSAF) is a technique used in audio processing to measure and separate the music from everything else. It is a proven technology that has been tested, and used in the real world in the field of acoustic echo cancellation.

Have you experienced this technology before? Good chances you probably have. Have you ever witnessed someone talking to a smart speaker to say e.g.

“Hey Google, what’s the time?” whilst the smart speaker was still playing music?
How was the microphone able to hear what was said, above all the loud music that was being played?

Here's how it works in the context of listening to, and testing audio devices, like speakers.



1. Splitting the Signal:
FSAF divides the audio signal into smaller frequency bands, called sub-bands. Think of it as dividing a picture into small pieces, like pieces of a jigsaw puzzle. Each sub-band contains a specific range of frequencies, making it easier to analyze.

2. Adaptive Filtering:
The technique uses adaptive filters that can adjust themselves in real-time to changes in the audio signal. This helps in accurately measuring each sub-band.

3. Measuring distortions
In each sub-band, changes in the signal can be treated as additive noise. By using a mathematical method called Least Squares, we can estimate the true response of the speaker and the room for that specific sub-band. This estimation gives us a clear picture of how the audio should sound without the added character, effects or noise from the speaker.

4. Combining Results: After analyzing each sub-band individually, the results are combined to create a full-band response. Think of it like putting all the puzzle pieces back together to see the complete picture. The added character, effects and noise calculated for each sub-band are actually discarded in this step.

5. Full-Band Distortion Measurement: The full-band response, which is now a more accurate representation of the speaker and room's true sound, is used to identify the character, effects and noise. These distortions are the leftover “residuals” after applying linear filtering.


Thus, FSAF allows one to play music through a DAC/ADC, an amplifier, or a speaker, and “subtract” the original input from it.

What you have left over is the “residual” - parts that weren’t in the original audio file.

So what can expect to hear with FSAF, when measuring your speakers?

For starters, yes you will get your frequency response.

You will also be able to listen to distortion. Harmonic distortion, intermodulation distortion… all audio that isn't the original file being played back. This includes noise, the Barkhousen effect, ringing, echoes… even faint sounds that you had been accustomed to and your brain may have tuned out e.g. cars driving by, the steady tick of the second hand of a distant wall mounted clock, it’s all there.

For a long time FSAF was used in industry in acoustic echo cancellation, but as of June 2024, it was incorporated in REW 5.40 beta32. It is still in beta testing...

Hi,
I have the ALPS27 Motor Potentiometer that I bought some time ago. I got it with a circuit for the remote conrol. But Ihave been looking all over to get a circuit diagram so that I can determine exactly what the J3 connector is used for. But I can not find anything. Any help with this is greatly appreciated.

I've a DSL710 that has the usual no disk issue. It's been written that it's really a detector issue, but no matter. After opening the unit to my surprise I to found a HOP-1000 laser instead of a HOP1200s that I expected and had previously ordered. Can someone confirm or dismiss that the two lasers are interchangeable,? The player at present plays factory CDs but no CDR or DVDs. So most certainly it's the laser unit....

Hello,

Name is Chris and I am working on taking a 1950's Zenith console and replacing the long-dead guts with newer (albeit pretty cheap for now) components. Keeping the the existing speakers but replacing pretty much everything else.

I will be posting in a few areas looking for specific help.

Thanks in advance!

Good afternoon everyone. I sold or gave away all of my programs but, bass box pro. Am looking to make a slot vented box for a pair of Mark Audio drivers I’m very fond of. The CHN-50s. I believe they are the best fullrange driver out there for the money hands down. So what program can I use to get the best results? Sorry if this has been asked before? Jeff

Whenever visiting DiyA, the first thing I do is scrolling to the bottom of the page to set the theme color and time/date format to my liking, but if I decide to log in and after a while log out, the theme color and time/date format cookies gets deleted at log out, would it be possible to exclude the following two cookies from being deleted at log out:

• diyaudio_language_id
• diyaudio_style_id

I have tried to use couple of cookie extensions in my browser and lock the cookies but it doesn't always work.


BTW, the placement of these theme color and time/date format settings at the bottom of the page and the behavior is rather annoying and frustrating when one have to reapply these setting several times a day, here's what the procedure look like right now:

• Scroll all the way down to the bottom of the page (or use "End" button on the keyboard)
• click on theme button
• page represents the available styles in a pop-up window, but... AT THE TOP OF THE PAGE far from the bottom
• ok, make a large mouse movement to the TOP OF THE PAGE to select the style...
• page reloads and GOES BACK TO THE TOP, hey I am not done yet!!!
• ok, one more round of this nonsense circus... scroll all the way down to the bottom of the page
• ... click on the "YYYY-MM-DD / 12-hour clock" and select the one used in our region.

...rinse and repeat...........

• page represents the available styles in a pop-up window, but... AT THE TOP OF THE PAGE far from the bottom
• ok, make a large mouse movement to the TOP OF THE PAGE to select the style...
• page reloads

Not fun.

Hello all. David Seymour here. Long-time stalker, first-time poster. My current project is a bass guitar combo amp using this Sunn 200S preamp from EffectsLayouts.com coupled into this cheapo Chinese mono power amp stage. Meant to be small enough for solo practice and lugging around, but powerful enough for small ensemble gigs, maybe even with a modest drummer. Heatsink, cabinet design, and 8Ω speaker selection is done, but I need help figuring out the power supply. I'm definitely not a qualified electrical engineer. I'm handy with schematics and a solder station, but I know better than to mess around with power mains.

The Sunndering preamp is designed for DC 9V, presumably 500mA is plenty. The Chinese power amp PCB says it accepts wide power supply range, dual voltage DC +/- 20V-90V (not sure what that additional 12V input is on the bottom right.) I'm thinking 200W, 45V or 60V would be best, right? Trying to keep costs down; it's a bass guitar amp, after all, not a HIFI or audiophile grade application.

Is there such as thing as a ready-made solution for this? If not, can anyone point me to some existing DIY solutions, schematics, BOMs, etc? If not, who is able to design such a thing for cheap?

Thanks and cheers! 🍻

Preamp stage:


200W Power amp stage:

Longtime audiophile now venturing into DIY audio projects….

Hi guys! Are there any DIY tapped kick horn projects anywhere in the world? Or have such projects already been realized and I haven't found them? Or to put it another way: There are no TH kicks because it is perhaps impossible to construct a tapped horn as a kick horn? Or has no one tried to develop one yet? Questions, questions! I'm hoping for a little exchange of experience from the experts!

Best regards

I'm building a new system using a SEAS 4" Full Range and a SEAS 6" woofer. I'm trying to determine the point on my preamp volume control that represents enough power to blow the 4" driver (coil melt or cone damage) so that I don't get to or cross that boundary during listening sessions. 90% of the time I'm probably listening in the 65 to 85dB range, which I know isn't close to the power handling of the speaker. However, I'm sure we've all experienced a time or two when very loud is very good and inhibitions aren't properly in place. 🙄

The speaker has a 3.3ohm voice coil resistance. Handles 40W long term and 100W short term. If I use the equation of W = V-squared / ohms then it looks like somewhere around 11V amp output is the stopping point. I'm wondering if I can put my voltmeter on AC mode across the speaker terminals and figure out where the volume equals 11V and don't exceed that number on the preamp. (I also have an oscilloscope if that is a better tool to use.)

I'm asking in the Class D section because I'm using an ATI AT524NC amp that puts about 350W into 3.3 ohms at full tilt.

This amp plays good audio but sometimes it will just start clipping.It does that even when there is no audio playing.

Any suggestions?

This amp power up has rail and regulated voltage.I drive a signal in the amp but am not seeing no oscillation on the outputs.Driver ICs have supply voltage.

Any suggestions?

Hi. I don't mean to sound like a Gil Scott Heron song, but I'm new here, will ya show me around?

I'm tying to get the maximum level from a Source player that has both options of RCA's and XLR's outputs.
I would like to unbalance a balanced source, does that mean a straight short between pin 1 and 3 or ...

Decoupling capacitor between pin 1; Negative and pin 3; Gnd ? If so what type, material, size?

I'm used to studio gear and this Source player is a quasi balanced device, too sensitive for my likings.

O/C between 1 an 3 results in buzzzzzz

David Seymour here. Long time stalker, first time poster. I studied music production and audio engineering many years ago, am just getting back into it. I've designed or modified many stompbox circuits, mainly bass distortion, octave fuzz, reamping. Working on a few more primo balanced mic/instrument preamps. I'm a bass player, but no longer have a bass amp, so I'm currently building a combo amp from components designed by smarter people than myself. I need help with the power supply section, which I'll be posting about shortly.

Thanks and cheers everyone for generously sharing so much expertise!

Hi all,

I guess I am posting this because it is a requirement to post.
🙂

Anyway, I never posted anything since my skill sets are still very novice and I never really had a question or challenge that I needed to seek an answer for. However, that is changing so I wlll be posting my question shortly that I hope someone can answer and lead me down the right path.

Dave

Hi everyone, and nice to meet you.

My name is Jan Zuidersma, I am from South Africa.

I have a little mix of vintage hifi equipment, and I also enjoy doing DIY audio a lot.

Looking forward to learn and share with you people.

Hi. I'm from Ukraine. I need help finding some service manuals.

I'm looking for recommendations for a quadrature rotary encoder with detents that has nice high-quality feel. Ideally, I'd like somewhere around 16-25 steps per revolution.

I've used the Bourns PEC11 series previously, and these work ok but feel kind of cheap.

Anyone have any experience with Nidec or Grayhill, or have other recommendations? I was looking at the Nidec REC20-25-201-1.

Thanks.

Hi DiyAudio community !

I’m really happy to finally join this forum after countless hours of reading fascinating threads. This place has been incredibly helpful as a guest, and I can’t wait to get involved as a member!

I work as a fabricator in metal working field, in a small but cosy workshop. I mainly work around art work production and design for other artists. As well as this exciting job, I'm interested in diy practices and open design ideas.

Last year, I discovered the joy to have a listening room. After a lot of different vintage hifi setups, it was time to build my own !I’d love to share more about it in a dedicated thread.

✌️

We got a new sky glass gen 2 tv on saturday and it seems that the only audio output is through eARC and my dac doesn't have a eARC connection. For using normal 2 channel stereo would it affect the sound quality going through a device that enables me to go from a eARC cable to a optical cable.

Hi everyone..I have been trying to find the volume control for the headphone amplifier output for Philips CD/Magnavox CDB 650..
The control is Alps ZV103A...I need to know where to buy it..Needing 4..
Thanks.

I started this thread to post the spreadsheet that I use to keep track of vacuum tubes that I have dealt with. I keep it updated as needed. I have posted older versions on diyAudio before, but they get lost in the crowd.

I have used this spreadsheet for years and have fixed several mistakes. I can't guarantee that there aren't any more. Use at your own risk, and consult a tube manual if in doubt.

I made this spreadsheet for my own use over a period of at least 20 years. It is not always consistent or complete since it was intended for my own use.

If there are any questions, requests, or comments feel free to post them, especially if you think that there is an error.

I'm sure that there are more goodies hidden in a dark corner of my 20 TB NAS drive box. I'll add them as found.

Hello,
I'm a novice at speaker enclosure design, so please forgive me if this is a dumb question. I'm designing an enclosure for a pair of small (3.5") full range speakers to use in a bedroom/office setting where high volume is not needed. I currently have the speakers in enclosures with a cylindrical port sized to achieve what seemed to be the lowest I could reasonably go without getting huge peaks and valleys in the transfer function curve using an app I found online.

While I like the sound of the current setup. I want to change some things to improve the overall setup and want to try to improve the bass output while I'm at it. Now, I'm looking to construct a multi-chambered enclosure with the full range drivers on each end of the front baffle and a 3" mini subwoofer in the center. To get the lowest bass output I can with a flat transfer function curve with the sub I'm considering, I need a very lengthy port that will take some creativity to build into my enclosure given the size limitations I have placed on it. I'm looking at using a labyrinth port design to achieve this.

After running different port scenarios in my app (single port, multiple ports all of different sizes, lengths, round and rectangular), I've concluded that a single port out of the speaker chamber gives me what I think is the best, flat response until it hits a steep drop off at the low frequency end. The geometry constraints of the overall cabinet makes it challenging to fit the labyrinth port in the box efficiently.

Finally to my question. Is it acceptable to have a single port out of the sub chamber, then Tee it a short ways downstream, then have two runs switch back and forth, terminating in two outlets? Can I count the length of port tubes on each side past the Tee toward the total length of port, or does the Tee somehow reduce the effectiveness of what comes after it.

Thanks for any advice you can give.

After some experimenting I'm left with some spare M2x input stages I'd like to sell.
All these were in my amplifiers working before their removal.
I have four IPs9's and two Cedarburgs v2.

Asking $20ea. plus shipping.

Hi all,

My Rotel RA-611 produces excess hiss - enough that it's audible from across the room with the volume at zero. I believe the issue is in the Tone/Preamp section, since if I pull the connectors to the main amp the hiss is gone.

I've traced through the circuit with an audio probe and it's nice and quiet until the collectors of Q204 and Q208. From this point forward there is the loud hiss in the circuit. I thought it might be the 50 year old transistors, so replaced the 2SC644s with KSC1845Fs, which made no difference. All of the voltages match the schematic, the capacitors have been replaced and the carbon resistors have been replaced with metal film - also made no difference.

Please find the schematic attached.

Many thanks in advance for the help!

I found a dead Krell FPB 200 on sale for cheap. The seller wrote that it goes into protection immediately.
from past experience i knew, its gotta be a short somewhere in the output drive.


what too heavy and those handles at the back make no sense, it messes up the front frace when placing it down.
the fins are also quite pointy and sharp, you cant grab it there.


So i started taking it apart.

Its well built, easy to take apart. a few screws here and there, and everything comes off in modules.

but its huge, its heavy. and its overbuilt as heck.
i mean what you need so many transistors to drive the output stage?




and its microprocessor controlled (biasing trough optocouplers that switch in and out resistors. it measured the rail and output voltage and currents at all time.




If its one of these, i might be screwed. Unobtanium TO-3's



i didnt wanna do this to it, if possible:







But after measuring several spots, i found no short.
so i kept digging, taking apart this PCB sandvich, comparing Left and Right for meter readings
and found a reading that wasnt equal on one so i looked whats hooked it to.
a shorted transistor, but after lifting the legs i noticed the short was still on the PCB

turned out to be a shorted silver Mica capacitor (1100pF)

after replacing it (all eight), it started working again!


for good measure i gave it a new set of electrolytics, the old ones where still good but the new ones are slightly lower esr and may last a while longer now.



After listening for a while.. i noticed this thing is a damn room heater.

60W in Standby!
180W when turned on..
1600W when theres a signal going in without even a load connected.. Class-A of course.

i listened a few hours and my room went to 35c.. nothing i want in my room 24/7.

it sounded good, nothing that immediately kicked me off my chair but alright for an class-a amp. may not have the right speakers for it.

i found my "jaboost" opamp-amp that ive built afterwards sounding slightly less fatiguing over long sessisons. and you dont want to leave the room afterwards.
https://www.diyaudio.com/community/...her-dont-expect-miracles.423889/#post-7929116


Its gonna find its place, someday, somewhere.

These have been unavalable for some time. It is a circuit for EQ of dipole woofers.
I am hoping someone has one they would be willing to part with.
Thanks

I have played around with similar simple circuits in the past for fun with numerous BJT Darlington Pairs as single package or discrete. Usually obtained around .0006 to .001%
1Watt THD1 using the classic 5534 as the driver. Typical expected 18 to 27 watts with op amps that you might tippy toe into the 22 volt area. ( Yeah Whatever more like 15 to 22 watts) to be safe.

A circuit design example was requested. So I made sure I was not blowing smoke up my own butt. Put together a more final circuit with hopefully real world stability margin.
Then run it up close to crest power and see what THD20 ( 20 kHz) was. Then post another sim amp that people love so much. Somebody might do a PCB ( probably not) and remind me diode thermal tracking makes everyone yell at the clouds. Until someone just has fun and is actually creative enough to build one. Woo Hoo

I like using diodes sometimes, because its fun and also easier with op amp drivers to use Boot Strap current source for the hardest swing you can get. Maybe live the Fantasy of getting 25 to 27 watts into a 4 ohm load. It actually does swing rather well driving a Darlington pair. The simple bias adjustment circuit will basically get you going dead centered at 50%. If you want to run 85 to 100ma for high frequency numbers about 55 to 65% will do it.

Yep T0-220 package for power transistor Using 80 volt 80 watt, 50 MHz fT with around max 1 amp current expected hfe should Hold 80 to 100 and yes minimum is rated 35 to 40 like a zillion other T0-220. D44VH10(NPN),D45VH10 (PNP) manufactured currently by On-Semi



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

Bandwidth 1 Hz to 514 kHz stability phase margin 50° ( thumbnail click to enlarge)


Clipping Behavior 4 ohm load ( thumbnail click to enlarge)


No Overshoot / Gain peaking 20 kHz 4 ohm load ( thumbnail click to enlarge)


THD1 ( 1 kHz) 2.83V output 1 Watt

THD20 ( 20 kHz) 2.83V rms output 1 Watt


THD1 ( 1kHz) and THD20 ( 20kHz) with 300mV input and 900mV input 8 ohm and 4 ohm

8 ohm load
============================
300mV input 8 ohm Load 1.2 Watts output
============================
THD1 .0004 %
THD20 .003 %
============================
900mV input 8 ohm Load 11.2 Watts output
============================
THD1 .003 %
THD20 .007 %
-------------------------------------------------


4 ohm load
============================
300mV input 4 ohm Load 2.4 Watts output
============================
THD1 .0004 %
THD20 .006 %
=============================
900mV input 4 ohm Load 22.5 Watts output
=============================
THD1 .004 %
THD20 .01 %
-------------------------------------------------

Slew Rate Measured 10 kHz 10 Volt Peak to Peak , 5 Volt Peak Delta X Delta Y 10% 90% 551 Nano Seconds
= 14.5V/us Slew Rate

Hello!
Very light background in diy from building Eurorack synth modules and from hanging around with circuit bending and sound art people. Newbie woodworker, synth noodler, ex-dj, ex-concert & event photographer, hobbyist fruit and berry gardener, some sort room acoustics weirdo/nerd, wannabe dry stone waller.

I have a small dedicated listening/media room in an old wooden house with quite studio'ish diy acoustic treatment. Current speakers are my first diy speakers, B&C 15CXN76 elements in 120l (net) bass reflex enclosures (Piste15 kit by Samu Saurama/audiovideo.fi), a stereo pair of Eminence LAB12 in ~32l closed enclosure sub/low bass speakers to compensate speaker placement related problems and a Genelec 7360 to bring some roundness to the lowest half octave with nice support from the room. Livingroom has Genelec 8340.
As an old raver / dj I'm interested in "party mode quasi-PA, but HiFi" speaker builds and I'm aiming to build a small sound system for my wife's 50th birthday party, she is a lot heavier basshead than me, so I need all the Emotional Support SubWoofers.

looking specifically at Radian:

https://radianaudio.com/collections/ribbon/products/lm8k-wide-band-planar-ribbon-transducer

it says the conductor material is "aluminum foil"

but aluminum is a material that suffers from fatigue when repeatedly bent. in a planar the "voice coil" and the "suspension" are the same part - so you now pretty much have aluminum suspension. the only other drivers i can think of that have aluminum suspension is Beyma Bullet Supertweeters ...

but Beyma Bullets have replaceable diaphragms, in fact i replaced it on mine ( due to issues unrelated to aluminum fatigue ). it doesn't seem like you can replace the diaphragm in a Radian driver.

would this be a cause for concern in a high SPL application ?

does GRS planar also use aluminum foil or do they use copper ?

does it make a difference for durability ?

thoughts ?

EDIT: i should add that Beyma Bullets have recommended frequency range of 5khz and up, while the Radian is recommended down to 250 hz so it would probably see a lot more bending of the diaphragm, though that's just a guess, not scientific analysis.

I'm building a open baffle/dipole speaker and want to try a rear mounted tweeter. The problem is the only extra tweeters i have are different, similar sensitivity though. Both are 4 ohm. The amps im using are ice power amps, is it safe to wire them up in series and try it out?

Hello friends. For a long time already, about 5 years I had unfinished acoustic systems, but recently I decided to finish them all the same. The speakers consist of a bass segment, which in turn is built of a double membrane stack, each with 4 stators and 3 membranes, a total of 6 membranes and 8 stators composed by isobaric. As well as a high-frequency emitter. Everything that I do I will gradually spread here.

For me, hum and noise are the most important aspects. I may not be able to hear THD 0.1%, but I definitely can hear hum if there is any.

Below is a simple way to implement balanced input. When the resistor ratio matches between positive and negative side, it gets optimal common mode noise rejection.



I am not going to convert my amp to balanced interface, but let's see what if we use the same principle to the existing RCA unbalanced interface.
Let's replace the opamp with a power amplifier. Like this, you can see it does reject common mode ground noise.

1. Ground noise is canceled when R1/R2 = R4/R3.
2. Don't bridge the signal ground to the chassis.
3. R5 is necessary, R5 needs to be << R2 so that the negative feedback ratio won't change much when the signal ground is floating in air.
4. Be careful when adding volume pot directly to the input of the amp, it is equivalent to an out-proportional R3. The noise cancellation condition no longer holds true. The work around is put a unit buffer after the volume pot.
5. As the input impedance of the signal ground is very low, 10 Ohm in this case. I would not call this a balanced interface. It is a still unbalanced interface. However, it has the capable to reject some of the ground noise. It will be an improvement to the existing RCA interface.

PS: C2 should be equal to C1 to get optimal results. Thus, the improved version is C1 = C2 = 22u.

PPS: In some situation, it might not be easy to put extra R3 or keep R3/R4 ratio constant. For example, a volume pot. I post a compromised version at #11. Although it cannot cancel out ground noise completely, the ground noise won't get amplified, either. Thus, only the differential signal is got amplified.

It has been a long road designing the Emerald Physics 600.2SE amp, but what we have finally achieved is totally worth it and an amazing sounding amp. This amp has enough power to satisfy any speaker one could imagine or need. I want to thank Jan Hofland for his help on this project as he designed the HyperSET tube buffer that feeds the 600w class D amp the audio signal. We are now finally starting production of this hybrid amp - a 600wpc Class D amp with a SET tube buffer (E88CC or NOS 6N1P). Accepts balanced or SE inputs. Selector switch for tube or SS buffer. The sound is powerful, smooth and clean with just the right secret sauce of a SET goodness. The grip on the bass is incredible. Built in integrated logic controller for graceful startup and shutdown to allow proper tube warmup and no thump. Chassis is satin finish bead blasted and black anodized CNC’d aluminum with a luxurious feel and heft (5mm thick panels all around). Real galavanometer VU meters to show power levels and quality Neutrik XLR/TRS connectors and premium Viborg pure solid copper binding posts.





Have a listen to this amp with some Vanguard bookshelf speakers:
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Hi! my wife buys a lot of bread boards for institutions and in the last two years the quality has severly dropped. Like the holes are not clean and contact is bad and so on. Do any of you have a reliable source for good quality bread boards?
Thanks.

Hello DIYers,

I'm new to this hobby and planning to make myself a set of home theatre speakers using my 3D printer. Maybe later also a sub if need be 😉

I saw this sub/mid range woofer, the TangBand W5-1138SMF, which seems to have really deep low end. I've seen it paired with the Dayton Audio RS100-4 ("Dinas" speakers from this video).

I really liked the concept and tinkered a bit in WinISD and VituixCAD. However, now I'm wondering if there wouldn't be other options to replace the RS100-4 with something more geared towards high end and let the Tang Band do the job in the mids.
Would you have anything else to recommend? Mainly cheaper. I had issue finding something that would crossover nicely at 1 kHz or so.

Also, how bad would the distortion be if a tweeter was to be crossover too close to Fs with a DSP ? I've read somewhere that it might be less of an issue than with a passive XO.

Anyway, happy to be part of the adventure!

Intrigued by good reports (Scott Wurcer and EUVL) on BF862 JFET I also got some. The idea was to build a preamp based on JBOZ and B1 concepts with gain of 2 (enough to drive F5) and ability to drive 5K next stage impedance. Also, 6-7 V of undistorted output signal is needed.
This is what I ended with and it's all I hoped for, it sounds great, better than 2sk170 in same topology - I suppose it's due to much higher transconductance and smaller parasitic capacity. The only "problem" is that BF862 is an SMD part but that's easily cured with small adapter PCB.
This is the preamp and PSU schematic:

I've always considered 10dB as the 'nominal' value for line stage gain in a preamplifier. Does that sound about right to you?

Simple question: Why do most (almost all) negative feedback amplifiers use Miller compensation rather than lead/lag? The latter has a much flatter Bode plot of the OLG which, I believe, should increase the slew rates.

What’s the disadvantage of a lead/lag (other than the need for larger caps)?