Marshall Jubilee misbehaving

I have a recently acquired 1987 Marshall Jubilee (model 2550). I've recapped, retubed and rebiased it.

The amp is a "sort of" channel switching amp. There's no independent EQ but there is a "clean channel" and a "lead channel". The lead channel introduces an extra gain stage and a diode clipping circuit. The amp has one weird behavior -- when on the clean channel, there is distortion that fades in and out / level jumps but only when the lead master pot is up. With the lead channel volume pot on zero the clean is perfect.

Here's the relevant part of the preamp schematic. Circled in red (because I think they're relevant) is the lead master pot, and the switch (actually a relay) that selects the channel.

1696861114853.png


With the switch in the position shown, the amp is on the lead channel.

The input signal comes in via V1A. The clean route goes more or less directly (ignoring the "rhythm clip" circuit) to the grid of V2A. The lead signal goes to the grid of V1B, and the signal off V1B feeds the diode clipping circuit and passes (via the lead master pot) to the grid of V2A.

When the channel switch flips to clean (the other position, not as shown in the schematic) the effect is to ground the lead signal heading into the diode clipping circuit and also to ground both sides of the lead master pot.

My thinking is as follows -- if both sides of the lead master pot were actually, properly grounded, the position of the lead master pot could not make any difference to the signal downstream as there could be no signal at the wiper of the pot. Since the position of the lead master pot does make a difference to the signal downstream, the assumption that both sides of the pot are grounded must be incorrect.

Therefore that channel switch (which is actually a 36-year old relay) can not be properly closing and grounding the signal and that component is likely the cause of the problem.

I'm not an experienced amp tech by any means, so any comments / critique of my thought process so far would be very welcome!

Aluminum vintage-looking case for Raspberry Pi/rpi "pifi"?

I'd like to add a small screen and a not-ugly case to my raspberry pi streamer (uses volumio). The only thing close to the aesthetic I'm thinking are the sold out ones here. Not sure if aluminum or steel is more commonly used.

Does anyone know of a company/source for the sort of machined metal style cases?

There's probably a proper word for this aesthetic/era but something (anything) like this:
1697136934465.png

What would cause no sound and a square wave output?

Hi all,
I am trying to get a test circuit up and running based on this chip.

https://www.diodes.com/assets/Datasheets/PAM8019.pdf

But i am not having much luck at the moment

Here is the input signal to the amp.


IMG_20231013_234729.jpg


An here is the output signal to the speaker which i get no sound from and a very suspect square wave.


IMG_20231013_234709.jpg


Any ideas what might be going wrong here to give such an output?

Notes:

I have left the PL pin unconnected
I have not implemented any under voltage protection as i assume this is optional.
I have connected pin 8 directly to VDD 5V rail

For Sale Hagerman Bugle 2 phono stage

This is for a hagerman bugle 2, that i have modified. the biggest mod is that it now runs off of 2x12V NiCad battery packs. It is dead silent.

The front has an on/off switch. When its off, you can press the button to get a readout on the battery's voltage. Anything around 24V is fine.

Opamps are LM4562, i dont know if they were the original specified opamps. They are DIP8 pkg, so should be easy to roll your own.

Charging the batteries, when needed, is simple. Open up the case, the batteries have easy quick connectors. Just hook it up to your charger (not included, but they're inexpensive) and charge one at a time. See circled section in the picture. The current draw is small, so you dont need to charge often.

soldering isn't my best work, so I'm open to offers.

$125/open to offers
bugle-1.jpg
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bugle-5.jpg
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Three Crossover Options for my Purifi PTT5.25/Beyma TPL-75 Project

Dear Music and Audio Friends,

Solen has simulated three different crossover options for my Purifi PTT5.25/Beyma TPL-75 Project.

P-B Xover LR2 : LR2.png


#1: Both legs 2nd-Order Linkwitz-Riley, crossing over at 2400Hz.

P-B Xover BW3 : BW3.png


#2: Both legs 3rd-Order Butterworth, crossing over at 2400Hz. The arrow points to the little bump at the 2400Hz crossover frequency. Which Solen would rather avoid, I take it.

P-B Xover BW2 : LR2.png


#3: Low-Pass: 2nd-Order Butterworth, crossing over at 2400Hz; High-Pass 2nd-Order Linkwitz-Riley, crossing over at 2400Hz.

If anyone wants to see the complete workups (driver specs, parts lists, schematics), please let me know.

All thoughtful comments welcome.

john

Choosing a BR alignement in relation to the driver's Qts ?

Hi,

So I have a 12" PA driver with a .38 Qts; Fs: 42 hz and EBP = 110 so BR advised. Vas : 113 L, limited Xmax : 7.5 mm : again in favor to BR load choice

Is there bass alignement that are adived based on the Qts driver number ? The Qts seems too big for a SBB4. Not sure a Bessel is the best if transcient (group delay flat enough) is what matters before low pass (passive) filter tweaking not talking about a Fb at 10 hz below the driver Fs if Bessel BR load is chosen !

Which alignement filter would you choose as a start (before low pass filter tweaking) with such Qts which is at the limit before flat and non flat alignement.

My basic understanding is room gain in most of the rooms doesn't really matter above 80 hz, yet matters belllows. Seems most of the time a smooth anechoic magnitude is good enough for most rooms. Something more close to the sealed 18 db slope. (so with BR slope nearer to 24 dB than higher BR slope)

So following that idea, still with my basic understanding about BR load: bigger sized box and more gentle slope.

I admit such a driver should be used as a uper bass-midrange driver. BUT, in an average listening living room, it is "maybe" large enough despite high Qts 42 hz.

What BR alignement would you use as a start before filter tweaking (group delay flatting) if transcient is favored as well as second order low pass filter ? Qts is increased with 0.5 ohms with a good coil and some length from the amp? SSB4 (more advised with Qts near 0.32) ; Bessel (but very low Fb) ?

BR : -F6 is circa 45 hz ; sealed : circa 78 hz.
BR : -F10 is circa 35 hz ; sealed is circa 45 hz.

Again, room gain seems more important below 80 hz in most average living room. And often we have to focus on the 80 hz to 200 hz for that tigth slam psychoacoustic wanted sound (again my basic understanding; Stereophile measurements, etc)

Which trade off BR alignement would you choose going with a passive filter with such driver datas for a bass driver in a 3 ways, please ? (as a start before group delay flattening with a second order passive filter).

My main concern is to use this 12" PA in a living room as a bass driver with a low pass from 200 to 300 hz and making strong enough the 100 hz to 200 hz area where the snap, tigth feeling is.
As going passive I am of course worried about the high pass slope I shoud use for the midrange above!

Hope it is not too much confusing question.

Edit : better to change the driver or at the opposit going active amp with DSP (but I fear the group delay when playing with IR bass boost with such low 7.5 mm Xmax)

new Dynaco PAS upgrade kits forum

Hi all,

I created a new forum where we can discuss about all of the Erhard Audio Dynaco PAS upgrade kits, https://www.erhard-audio.com/EA-Forum/index.php
This forum is very new, so there are not many members as yet, but there are a few posts covering some updates and changes we've made to some of the upgrade kits.
This is an exclusively Erhard Audio PAS upgrade kits forum only, for any general Dynaco PAS preamp questions or other make/brands of kits, it is best to keep discussing those here at diyAudio.
I look forward to seeing you at our new forum.

Holger
Erhard Audio

ps. yes, I did get permission from diyAudio to post this 🙂
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Trying to repair a Meridian 500 transport

I'm trying to repair a dead Meridian 500 transport for a friend (VAM1205 version). It came to me with two faults:

1) Badly struggling to eject and close.
2) Failure to read discs.

I've now got the drawer action working well but it still doesn't read discs even after I replaced the optical block with a brand new one.

With a disc loaded, if you press PLAY you get the message 'LOADING' on the display but sod all happens, no disc rotation at all. This was the same with the old optical block too.

I haven't been able to find a copy of the service manual so I'm flying blind with this one. If anyone is familiar with this machine and knows of some common failure modes I'm all ears! You might save me a lot of trouble.

Sony CDP X555es tick-tick-tick-tick

Hi: I've searched for this particular symptom without much luck. Hoping for ideas.

This repeated sound suddenly appeared when playing a CD. Sound is a group of five (?) quick ticks that repeats when playing CD. It is quiet between tracks.

Affects all my commercial and home-recorded CDs.

I'm quite sure sound is electrical not mechanical.

Tick sound appears on all tracks and in both channels.

Player works properly, gets a TOC quickly, fidelity is normal.

Tick sound level is well below music level but easy to hear.

Tick sound appears in RCA and headphone output. I haven't tested optical output, I don't have a converter.

I replaced laser ~10 years ago.

Anything I can add? Thanks in advance,

dizmayed

macMini (HDMI/USB) to diy DAC (RCA S/PDIF) what's my options?

Hi friends!
So after a long journey playing around with analog, I'd like to turn my eye to this DIY Dac I got from a former DIYA-member.
Problem is, I need some conversion.
Source is a mac mini 2018, with either USB or HDMI out, and the dac only has that SPDIF RCA plug.
I've found some cenverters, M2tech for example, most are way beyond my budget.
What's my options?
Can I connect something HDMI to RCA?
Is there a simple DIY jig available? (I am not very good at electronics, and nothing in digital, although I build a few things...)

thank you!

Stick-on fin extenders to improve heatsink performance?

I'm trying to help my friend fix an overheating problem with the voltage regulators, causing instability and crashes. We know improving airflow will resolve the problem, but I'm trying to find a way to improve overall dissipation capacity as well. Could I use a thermal adhesive to glue longer metal strips to each fin of this heatsink to improve its dissipation capacity? That seems easier than trying to remove the heatsink and install a larger unit onto the chips directly. Thoughts?

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PCM1794 Mono mode

I'm a bit puzzled on the PCM1794a Mono mode: it's clear that in Mono mode, the one channel's analog output is inverted to allow balanced operation. However, it's not clear on WHICH channel is inverted! Absolute polarity is important in my application as I'm building interchangeable DAC cards for a DSP application where the polarity needs to be consistent.

When the DAC is set to Mono mode left channel, I would presume the right channel's polarity is inverted, and vice versa for the other way round.

Table4 on p17 stipulates that a digital full-scale positive is represented by a full-scale positive voltage output when using the datasheet standard opamp I/V and summing circuit.

The Mono mode in Figure27 p22 then shows how the output of the Left output in Mono mode is called "OUT+" yet it goes to XLR pin3/cold.
Similarly the Right output is called OUT- but goes to XLR pin2/hot.

The question therefore is:

- In mono mode, is the polarity inversion applied to the opposite or selected channel?
- Does this mean for a dual-mono application with 2 identical circuits but one strapped for left and the other for right, the one channel would need to invert its polarity at the XLR plug to retain phase?

SIT Choke Loaded Follower

This project, like most DIY, is based on the work of others. Nelson, Zen Mod, Ben, Mikey and Lynn, were all a huge influence on this project.

Starting with Nelsons 2015 Sony VFET front end, further modified by Lynn for his version of the XA25. I have been using this front end on many amps as it gives a nice 20dB gain and has an adjustable harmonic spectrum. I normally set it for minimum H2 and let the output stage add the honey. I have this frontend driving one of ZM’s DEF R3 output stages, a big SIT-Mu Follower, a ZM Redneck DEF with a Pass SIT1 and now the Choke Loaded Follower. This has helped me in evaluating the output stages as they all have the same input/gain stage.

This Choke Loaded Follower is following Nelson and Mikey’s work and then all of Bens excellent posts. It does incorporate a buffer for the SIT. At first I was concerned about this additional stage but it worked so well in ZM’s DEF R3 that I figured I would give it a try. It’s a good thing. IMHO It add additional speed and transparency over direct dive of the SIT and the bias is nice and stable.

The reason for this amp is for the summer when I need something that dissipates a bit less heat. The Big SIT-Mu Follower is excellent sounding amp but at 185W per channel heats up the listening room. Ben said that he liked the sound of the Choke Loaded Follower and at less than half the heat dissipation it seemed like a good thing.

The caveat with the Choke Loaded Follower is that adjusting the VDS is problematic. You can change the bias and this changes the drop across the choke. I played with different supply voltages and the higher you are, up to about 37v the lower the distortion. 32 volts seemed like a nice compromise and that gave about 30V across the SIT at 2.5Amps. I have included a graph to see how bias changes the distortion with a fixed 32v supply.

Lastly I added Nelsons De-Thumper, just because. I’m sure my vintage Altec 515’s will appreciate it.

Oh I like to see my work, (modern art?) so it’s all in the open. It also makes it nice for adjusting and testing.

I hope it inspires people to try the Choke loaded Follower. It sounds great!

Jim and ZM like pictures so here are a bunch…

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Sub filters - Linkwitz Transform etc

I built LW's Transform on a pcb from Rod Elliot (sound-au.com). Pcb no 71.

I used LW's values rather than rod's as these were a closer match for my subs.

These are the lower closed box section of what were once, a pair of ML Aeon hybrid electrostatics. (I sold the worn out panels and crossovers on ebay).

First chart is the ML response without any eq, second one is with the LW Transform in circuit. I installed it into an old Arcam Alpha IIs tape loop.

Charts to followML NO EQ.jpgalpha2moded.jpgmlsub2.jpgmlsub1.jpg

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6AH4 Preamp..A great option for line stage

Started researching a good line stage to match these 1625 power amps. After a lot of reading, stumbled on the low mu approach. The more I read, the more it made sense. So, found this design by Thomas Meyer, which I found intriguing. Was initially going with a 12B4, but I love the sound from octal tubes, so went with Mr. Meyer's design.

The parts cost forced me to re-consider this project. However, after a talk with my friends at Heyboer, found out they could make the iron for a reasonable price.

Since I wanted this to come out great, enlisted my buddy to build it. He is much better at this type of work than I am. My skills are OK, but nothing like his. Since he was intrigued with the design, he happily accepted the job.

The only real change made was using a Khomozo stepped attenuator connect to the input vs. the transformer volume control on the output. Used a slightly beefier power transformer, DC'ed all the tubes, and went with 6DE4 damper tubes.

This preamp sounds better than anything I've ever heard. It's shocking just how good it sounds.

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Markaudio 7MS and CHP90 enclosure options

I've been looking at enclosure options for the Markaudio 7MS and CHP90.
Ideally I would like to build one enclosure suitable for both, but I recognise that this might be far fetched.

What I've found interesting is that opposed to a regular BR, some of these more complex enclosures have wide backwave output in the mid bass, mitigating baffle-step losses.
Just look at this one from @nandappe :

chp 2.jpeg


chp.jpg


And here is a comparable one for the 7MS:

7ms 2.jpeg
7ms.jpeg


I've also looked at the Okapi compact MLTL for the 7MS: https://www.markaudio.com/wp-content/uploads/2022/08/Okapi-7MS-compact-MLTL-metric.png
Would this achieve some of the same effect? (wide backwave output up to 700-800 Hz to mitigate baffle-step)
Is there an enclosure that would be suitable for both the 7MS and the CHP-90? (adjustable duct length etc)

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Help!! Understanding datasheet and signal bias

Hi All,

I am really hoping someone will be kind enough to shed some light on the following.

I don't have much experience with amplifiers and am struggling to get my head around a specific aspect of the data sheet for the PAM8109 chip

https://www.diodes.com/assets/Datasheets/PAM8019.pdf

1697120232574.png



Pin 4 in the datasheet is labelled as "4 -- Bypass -- Bias Voltage for Power Amplifier"

My 5V audio source is already biased to 2.5V.

My question is:

1. Does this chip accept a signal that has already been biased.
2. What is the chip asking me to supply on pin #4
3. The example application seems to imply that its 0V connected to ground via the bypass capacitor.

Am i good to go ahead and feed in my existing signal, and the bypass pin would only be fed a voltage if the signal was not biased and then would handle the bias internally if a reference voltage was applied to that pin.

I feel like i'm making too many assumptions and just guessing at this point.

Thanks in advance and apologies if i am misunderstanding to the point my question makes no actual sense 🙂

JVC Victor QL-Y7 turntable

I recently took a chance on this Victor QL-Y7. When it arrived the platter would start to spin but then stop again. The series of switches along the front panel seemed erratic so I squirted them with contact cleaner, whereupon the whole thing stoped working completely. I managed to get one of the switches apart, and the little metal contacts at the bottom were corroded, so I cleaned them up but it made no difference. Sticking a meter across them (removed from the PCB) reveals the resistance when energised ranges from 200 to 400Ω. I figured they were shot, pulled them out and replaced them with new parts, albeit of an entirely different design.

I've replaced most of the electrolytics in the deck now (just waiting for a couple I didn't have). Some of the larger main supply ones were toast.

The platter now turns again as soon as the turntable is powered up, but none of the switches do anything. The two on the motor unit itself, which select 33 or 45rpm, are touch switches which apparently rely on capacitance to work (except they too are unresponsive). I'm assuming the ones on the front panel aren't, as they're plastic, but the fact that the little nipple inside them that bridges the gap between the two metal contacts is made of a rubbery material is making me wonder whether there's something odd about these switches. Does anyone know what the score is with them? I might reinstate them, having now cleaned them all (they're a fiddle to get apart).

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DC offset problem

I know this is car amplifier, but beside the inverter it is a solid state.

Before i use tda2050 with 24v dc ct inverter in my car, no pcb i solder the resistor and caps straight to the pin as short as posibble. Sounds good, but it is hot and i think it have potential fire hazard.

So, i buy this cheap amplifier and replace most component with a better one. This is what i do:
1. Change all the 1/8w carbon resistor with 1/4w metal film.
2. Replace original capacitor to better capacitor. i use elna re3, silmic ii and panasonic fc. For smaller value i replace mylar with generic mkm and philips mkt. I replace mono ceramic with 2kv ceramic.
3. Replace the original jrc4458 with original ne5532 from ti. Yes, they got both input protected with transistor configured as diode.
4. Change supply for opamp from zener with lm7815 and 7915. I measure the voltage at +15.1v and -14.8v. I use two 220ohm 2wa resistor in parallel before the regulator.
5. Rewind the inverter toroid from 24v to 32v. Replace the original pair of 4700uf 35v capacitor with 10.000uf 50v+1000uf.
6. Add another gain stage before the amplifier with resistor divider at the input, i use ne5532 set around 9x gain. I use shielded cable around 10cm, i also ground the shielding wire. The total gain is around 2.5x. i do this because the source voltage output is to low, if i increase the volume i notice distortion using 50hz test tone.
7. Change the final transistor on the channel that i use for my woofer and tweeter. Before it was 2sc5198/a1941, i replace it with 2sc5200/a1943. It got 2 way passive crossover.

It works, no noise albeit a little muddy. Maybe because i use elna silmic which need some burn in period. And yes, i know it is improper using electrolite as input caps, but i like it. The original caps is generic eletrolite anyway, so it might be an improvement.

The problem is, i got 120mv dc offset on one ch. The other 3 was below 20mv measured. On this ch i use the original final transistor, it was the 2sc5198/a1941. I recheck all the resistor value, nothing wrong. I also resolder all the solder joint. The one thing i notice is 0.01v diffrence between the base of the 2sc5198/a1941.

Did i miss something? Do the 0.01v diffrence between the final transistor pair base is the problem or should i look somewhere?

Here i attach the before and after pic. i add the gain stage later on and is not in the after pic, it was before hf filter on the amplifier input(it was 2.2k resistor with 100pf to ground, the input resistor is 10k):

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Onkyo M504

I recently purchased an onkyo m504 and have received some help on connections and for that I am grateful as I cant even obtain a manual. Would anyone know if I can just use standard rca type monster cables for the direct inputs to receiver or is there a special cable designed specifically for this type of setup. Next concern is ,is it acceptable to plug this unit into a joule protection outlet with other units sharing same one? The outlet bar is a good quality piece by monster. Your suggestions appreciated thanks .

  • Locked
can I change title of my thread?

Hi,

My thread came back to life, but the title doesn't reflect properly the content. I'd like to edit the tile.

https://www.diyaudio.com/community/threads/tgm8-an-amplifier-based-on-rod-elliot-p3a.245619/

Current Title:

TGM8 - an amplifier based on Rod Elliot P3a​


Proposed New Title:

TGM8 - my best amplifier (incredible bass, clear highs, no fatigue)​

Yet another funny Chip-Amp

Yet another funny Chip-Amp: The Deviant

Hi there,

I have already described a number of amplifiers based on chips not initially intended for audio, mainly voltage regulators:

http://www.diyaudio.com/forums/chip-amps/176052-now-regulator-chip-jlh-amp.html
http://www.diyaudio.com/forums/chip-amps/192934-se-class-regulator-chip-amp-madness.html
http://www.diyaudio.com/forums/chip...oss-tringlotron-regulator-chip-amplifier.html
http://www.diyaudio.com/forums/chip-amps/193214-class-chip-amp-now-complementary-version.html
http://www.diyaudio.com/forums/chip...ator-chip-amp-family-welcomes-new-member.html
http://www.diyaudio.com/forums/chip-amps/175457-just-fun-regulator-chip-amplifier.html

This time, I set my sights on another category of circuits, vertical deflection chips for CRT TV's.

These circuits contain a number of function blocks, including a power amplifier.
In its normal application, it is rather far off from an audio amplifier, being current output and featuring flyback handling circuitry, but it can very easily be converted to an audio amplifier.

To that end, the flyback generator is left apart, and the PA supply is directly connected to the power supply.
The inverting input is available at pin 10, and is connected to the feedback divider formed by R3 and R4.
The DC output level is set by R5.
There is a compensation cap, C3 connected to an intermediate stage of the amplifier. There is also a 220K resistor, but it does seem to be useful for audio; it probably helps control the circuit during the flyback period, and its only apparent effect is to reduce the loop gain.
The inverting configuration is slightly inconvenient, because of the lowish input impedance, and the fact that the amp goes into unity gain when the source happens to be disconnected.
To avoid instabilities in this scenario, C5 provides an HF path to the ground.
Note that the feedback resistors could be made significantly larger without problem: the typical input bias current is only 100nA.

Now, the $1,000 question: does it actually work?

Yes!!!!, and rather well: it provides a good clean output, is well behaved even with a 10KHz squarewave and has a power bandwidth extending to 25KHz, quite OK for audio.
With a 34V supply, the output power reaches 13W at the onset of clipping.

To cope with this power, I have improved on the original heatsinking method: I have made a U-shape copper heatsink link by bending a power bus strip.
It is soldered to the tabs, very close to the case to minimize the thermal resistance, and the underside is pressed against the bottom of the IC, with thermal compound in-between.
In those conditions, the chip has no problem delivering 13W sinus into 8Ω. It wouldn't be prudent to go lower, as the maximum peak current is already exceeded.

Could other function blocks be reused in an audio context?
The oscillator is probably ruled out, because of the positive feedback.
It would be tempting to use the flyback circuit to boost the supply voltage, like in Philips chips, but bits are missing, and it requires an inductive load to operate.
The internal regulator (pin 7) can be used to supply other circuits, like a preamp or a tone control.
The buffer, between 12 and 1 is probably perfectly usable, I will certainly try it.

One thing to be noted: this circuit does not include a short-circuit protection, and I had the occasion to verify it the hard way: an alligator clip went loose, brushed very shortly on something else, and it was gone.
Funny thing, it didn't go dead short, unlike 99.9% of semiconductors: instead, it went open circuit, something must have played the role of a fuse, either a bonding wire or a metallization.
This means that I didn't have time to carry out more detailed measurements, like THD for example, but anyway, it looked quite decent.

In short, a good little chip-amp requiring very little passive components and working surprisingly well.
Note that the TDA1170 is a very old circuit, modern IC's can certainly do better regarding power, bandwidth, and features like short-circuit or thermal protection

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3-band tone control preamp excessive hum

Hello,
I could use help with a preamp circuit I have built. This is a 12AU7 cathode follower buffer into a 3-band tone control from the Max Robinson website: https://www.angelfire.com/electronic/funwithtubes/Amp-Tone-A.html
The output of the tone circuit goes to a John Broskie CCDA line-stage: https://www.tubecad.com/2009/03/blog0161.htm

Functionally everything works, but with the tone control section in the circuit I am getting hum on the output. With the scope probe, the excessive hum appears starting at the plate of the first CCDA triode. There is hum on the grid, but it is very, very, small signal. If I disconnect the output of the 12AX7 on the tone circuit and connect the cathode follower out directly to the CCDA section, no more audible hum. (the green line on the schematic). The signal gain of the circuit remains relatively the same whether the tone section is connected or not.

The power supply is very well filtered. The 12.6VAC heater supply is elevated 48V.
The hum signal on the scope, when spread out to 20usec shows about a 50kHz signal.

Could it just be heater noise? Thank you for any ideas

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Mixing Sealed & Bass Reflex Subs, Phase Adjustments?

Hi all,

Is there an elegant simple solution to applying a filter of some kind in a DSP (such as MiniDSP HD 2x4) to adjust phase from a frequency range on a sealed sub so that its easier to integrate with bass reflex subs? I'm trying to flip the phase of a sealed sub in the 10hz to 25hz range (unless it would be easier to flip the frequencies above it, and then just invert it in the DSP). I was reading about all pass filters. And reading about other methods, but they're beyond complex. Surely there's a more simple way as this is not for a full range speaker but just for a very limited range such as with sub frequencies (10hz to 100hz or so total).

I have a Umik-1 USB microphone, MiniDSP HD 2x4 and some basic use of REW at disposal.

Edit: Surely it's not as simple as using RePhase plugin and using the parametric phase sliders to adjust phase -180 degrees in the ranges I want and just generate and import into miniDSP FIR on the channel output?

Very best,

Arcam A85 help needed please

Hi all, first post here and not one I wish to be making! I've owned an A85 since 2002, was in mint condition. In the last 10 years or so it hasn't been used very much for audio. It was paired with a CD72 and used to be the centre piece. More recently it was driving the fronts on a 9.1 atmos set up on a denon x4100w. About 6 months ago the denon blew up whilst just powered up (not driving) so I replaced it with a 9.1 powered Onkyo RZ50, now rendering the a85 kind of homeless...
And here's my point, I agreed to sell it and courier via parcelforce insured who managed to trash it the front panel within 24 hours.... fuming!!
I've refunded the buyer and they have returned the amp to me. I've stripped the front off and already messaged Arcam who no longer stock parts. I feel I can repair the broken parts almost fully, my only issue is 2 components on the front PCB - I've looked at the repair manual and got the following info
The power switch on the PCB doesn't work (SW17, TACT SWITCH BOURNS SDTX-644) and the volume control was broken away from its board attachment (SE18, EC16B2414)
I have managed to power the amp on by shorting the power switch on the PCB and controlling the volume via the remote. It all works fine, so if I can get a replacement for those 2 parts I can at least get another a85 into circulation.
Can anyone help identify replacements please?

Alpine CDA-7838 Issues

I have a truck I inherited from my late brother, and he was big into car audio, having worked in the 1980s as a professional installer for a while. He installed an Alpine CDA-7838 Receiver/CD player in it, but it's got some issues. I do electronics repair work, so I'm just hoping to get pointed in the right direction since I have little experience with these. This is a unit with a removable faceplate. In general, it works fine, but it has some glitches.

Sometimes if I go over a bump, the whole unit will reset, and I'll lose all the radio station and time settings. I've checked the power supply, ground, and plug connections, and they seem to be clean and tight.

The lighting of the control buttons is flaky. Sometimes the buttons are lit. Sometimes they aren't. According to the manual, there are no controls for the lighting, so it seems like they should always be on.

Are there any known issues with Alpine units of this era? I'm thinking cracked solder joints. Should I look first in the main unit, or in the faceplate?

ZCD based delay

I was recently diving down the "distortion" rabbit hole. I had an idea about getting rid of crossover distortion. But I hit a snag. The idea was to use an all-pass filter and have it be triggered by a ZCD. I was thinking of the all-pass being an elliptical as well (for the steep curve) but I'm not sure I would want all that ripple. I guess I have two questions then:
1) is this feasible in the analog domain?
2) what filter would give me a steep enough curve to be right on top of the zero crossing? For that matter, would I even need such a steep curve?

Dayton Audio DATS V3

Good evening everyone

I just bought a Dayton Audio DATS V3. I know there are several previous posts regarding the use of this device. However, I just wanted to share my experience using DATS V3, and hope to get some feedback from people with more experience than me. At arrival, I quickly set up the DATS to run some T/S measurements on a handful of drivers I have laying around. These drivers are:

Scan speak 21W/8555-01 (4 pieces)
Scan speak 21W/8555-00 (2 pieces)
BMS 15S320 (measured only one)
Fostex FF85K (2 pieces)

Some observations:
  • I have four 21W/8555-01, which all measures pretty much the same (very high degree of consistency). All four woofers new in boxes. I broke in two them by simply running a low a freq. signal in free air for three days before measuring them. The remaining two came right out of the box (no brake in). All four woofers pretty much measured the same (only some minor differences). For all of them, Fs is very close to factory spec (meas. Fs = 19-20 vs. fact. spec Fs = 19). However, Qts is way off (meas. Qts = 0.37-0.38 vs. fact. spec Qts = 0.26).
  • For the two Scan speak 21W/8555-00, observations were similar to the observations for 21W/8555-00. Right out of the box, the two woofers measured the same (meas. Fs = 21 vs. fact. spec Fs = 20) and (meas. Qts = 0.44 vs. fact. spec Qts = 0.31).
  • Measurements for the BMS 15S320 was bang on factory specs (meas. Fs = 42 vs. fact. spec Fs = 41) and (meas. Qts = 0.29 vs. fact. spec Qts = 0.28). Also, this woofer came right out of the box.
  • Measured specs. for the Fostex drivers were quite different from factory spec. Higher Fs and Qts. I got these drivers from Planet10 some years ago, and they came with measurements performed by Planet10. By comparison, I got slightly lower Fs and Qts, but differences were marginal. That makes me feel somewhat confident that the DATS V3 works as intended.
I intended to make a FAST speaker system using the W21/8555-01, but I got somewhat concerned using the factory specs. I would very much like to hear from someone else with more experience on this topic.

Cheers

Best way to wire this power supply?

Hello

i have a power supply board here (already bought) (https://www.thel-audioworld.de/module/NT-HQ/NT-HQ.htm#NT25) that is able to provide symmetric power and im wondering if i could power a +9V dac with -4V/+4V instead of +9V/GND and if it would bring any potential benefit ?

another thing im wondering since these can act as 2 seperate power supplys ... my dac needs 9V and my headphone amplifier 18V, could i series both sides for 18V and just use one side for 9V at the same time?

(also any comments on the power supply itself are appreciated, is it good? and can it be tweaked further? additional caps?)

yet another *cube* enclosure

Hi folks,

The plan to make yet another cube enclosure with full range speaker. I know it's not the best but please bear with me.

Outside cube will be 18cm3. Internal wall will be 2cm thick with 4-layer perimeter and a 50% infill. Material will be black PLA. Outside will be sanded and painted, I want a piano black finish.

The front face will be made of 1 inch birch plywood.

I plan to use the Mark Audio CHR70.3 but I’m flexible if a better match could be found in the same price range.

Box will be closed back with some fill material, but it could be ported if desirable. I will use the speaker only near field in a small room. Port distortion, scuffle or other noise is a big no-no for me. That’s why I plan to go with closed enclosure and maybe add a subwoofer if I find the bass is lacking.

The big part that I need help is with the internal geometry. Maximum possible internal volume would be a 4L cube. However, since I’m printing it, I could easily make a cylinder, a demi-sphere, rounded corner, etc.

I thought going with the golden ratio, but it would reduce the internal volume to only 1L which is not desirable.

3L internal enclosure is possible with a demi-sphere design or a cylinder design. I’m not sure if losing 1L internal volume for that is desirable.

I could go the rounded corner route, which will have a lesser impact on the internal volume, but I’m not sure if it would help with the resonance or not.

What do you think would be the best to avoid the cube resonance issue?

Should I brace the speaker in such small enclosure ?

Thanks for you help 🙂

Look inspiration: Login to view embedded media PLA 50% review : Login to view embedded media

For Sale Precision Preamplifier PCB

Had some PCB's made for a preamplifier project. It is the "Precision Preamplifier" by Douglas Self, as described in his book "Small signal audio design". And after finishing this project, I have some boards left. The amp worked great btw, and I've only discovered minor error in the design (swapped "Hi Gain" and HI freq" text in slikscreen and too big lead spacing for two caps, but nothing that really mattered).

So if anyone is interested; I've got 4 pcs which I offer for €4 each + shipping (about €7 for Europe). There's schematic and BOM with Mouser part no's available too.

DM me if you're interesting.

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"Cube" box design FR and subs help

I have a bunch of "cube" shelves. They are 16in wide x 16in high x 12in deep. I'd like to build matching boxes for a set of speakers and subs.

The plan is to have 2 full range speakers, and 2 subs. I think I've settled on 8" all around, but I could do differently if that would sound weird.
I think I'll build them to look like the MasterSounds speakers since they are symmetrical and fit my "design limitation" (the cube shape).

I need help with the internals though. I've looked at a few different layouts/blueprints around but I guess I need a sanity check. Mainly I can't figure out why I would pick 1 layout over another. Like bass reflex, maze, single fold L, 4th order, 6th order, so on. based on MasterSounds they just list the Speakers as Bass reflex, and the subs as 4th order. Should I do the same? CAN I do the same? Do I even need to worry that much? I was thinking just doing it like this for the Speakers. And like this for the subs.

Linn Lingo vs. Dr. Fuß or Square-Wave vs. Sine Wave Oscillator for Motor Control

There are several motor control units (in Germany sometimes called wrongly "Power Supply") for turntables/record players with synchronous motor.
Most of them uses therefore a quarz oscillator (square wave), divider and low pass filter so as one or two power amplifier sections (mostly totem pole topology) for the motor coils.

But now I have heard about an other approach for higher spectral purity of the sine wave signal (i. e. lower distortion):
The Dr. Fuss motor control. Check therefore this URLs (control device with Wien-Robinson-Bridge sine wave oscillator section):
Erfahrungen mit Netzteilen von Dr. Fuss - fairaudio - leserbericht (1)
Dr. Fuss Motornetzteil - Frank-Landmesser.de

The circuit describtions for such an oscillator circuit you will find there:

http://www.ti.com/lit/an/sloa060/sloa060.pdf (page 17/18)
Types of Quadrature Oscillator | eHow.co.uk
https://web.archive.org/web/20130615120921/http://www.mics.org/getDocum.pdf?docid=1332&docnum=1
Quadrature Oscillator (Part 1) | Liivatera: Diary from the workbench
Liivatera: Diary from the workbench
http://www.ti.com/lit/ds/symlink/lm4766.pdf

Dr. Fuss uses either a Wien-Robinson-Bridge or a Bubba Oscillator/Quadrature Oscillator/SVF ( according TI "Wein bridge Oscillator" resp. "Wien bridge Oscillator") and an audio chip amp, model LM4766TF from National Semiconductor.
Attached PDF in post #16 under
Homebrew Motor Control Linn LP12 - Circuit Description wanted for Sine Wave Osc.
shows a schematic of a clone from Dr. Fuß motor control - based on a state variable filter topology.

Linn uses fore the LINGO two quarz oszillators and two power amplifiers for 33 and 45 rpm and for the Valhalla PCB one oscillator and one power amplifier so as an additional capacitor for the second coil (phase shift) - go to the attached files.

What are the pros and cons of both ??

Thank you for your comments.

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A ON/OFF AXIS brain storming and influence on bass response

Hello, welcome! I'm back in that unhealthy vortex of dissatisfaction that is created when you try to listen to music and you are convinced that there is a way to listen to it better. I've had a pair of old but good MTM transmission line tower speakers for a while now, so far listened to in my living room in a 'calm' state, all ok, but perhaps 'too ok' in the sense that I a little bored, and I said to myself: why don't I do some measurements to see if I can liven something up? I then saw that in my opinion the Morel was cut too high, thus losing much of its expressiveness, and causing, again in my opinion, a bad off-axis response, because the mid-high range was entrusted to the midwoofers (Vifa 6.5'' ) up to too high; also the time alignment was good but could be improved. This is what I achieved with fairly simple changes (also perhaps understanding how to avoid the mistakes I made before which always led to sibilance problems):
original.JPG

ORIGINAL situation

modified.JPG

MODIFIED situation

overlay.JPG

OVERLAY of the two (I attach the simulations only for the convenience of superposition and because they are quite faithful to the measurements, which only have a more regular and flat trend)

A more open and emotionally engaging, dynamic, 'fresh' and pleasant mid-high range, which in my opinion exploits the expressive potential of the beautiful Morel much more.
Furthermore, I also installed 5 cm sound-absorbing panels on the rear wall behind the speakers, which are approximately 35 cm from it, for a height and width equal to the height of the speakers; the mouth of the transmission line is on that side about 35 cm from the ground.
The problem is that now I can hear the basses well everywhere in the room... except the listening spot on the sofa in the middle of the living room!! When I sit there, the bass disappear into thin air, if I walk around they reappear in all their beauty.

My question is: what could this phenomenon depend on?
  • from the fact that the off-axis response has worsened rather than improved as I expected (and therefore I haven't understood anything about it so far), and then I 'feel' the mid-high range stronger only on-axis?
  • from the fact that before I was more careful to 'look' for beauty in the medium-high range and didn't care about the low range? Yet I am convinced that there wasn't that much difference in bass between sitting and standing....!
  • from the installation of sound-absorbing panels?

Thank you to all!

For Sale Wood frames for tube amplifiers

These are "raw" High quality american walnut wood frames + 3 mm thickness aluminium top plate + perforated aluminium bottom plate.
Dimensions are:
390 mm x 450 mm x 80 mm.........3 units
260 mm x 580 mm x 80 mm.........1 units

Price per unit ......140 euro + shipping anywhere......
Thank you.
I use them into these projects:
https://www.diyaudio.com/community/threads/el34-baby-huey-amplifier.326920/post-7380662

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USB-output to USB-input?

I'm considering using a small machine (such as a thin client or SBC) to make a digital crossover with EQ. My source is a bigger desktop PC, providing sound from several different applications. Both machines would have USB connectors (and preferably be running Linux, if that matters). From my initial searching, it seems that most people would use audio interfaces/converters (e.g. toslink) to make the link; plugged into the USB ports at each machine.

Is there no easy way to stay with native USB for the link, or if not would the conversion to toslink (or similar) and back again make little/no audible difference?

Thanks,
Kev

Totem Mani-2 opening cabinet

Anyone have any experience opening up the rear of the cabinet? Several Screws removed but tolerance is so tight the panel does not want to come loose. Remarkable build quality but I need to access the tweeter which will not come loose either after removing its screws. Sure I can pry the tweeter or read panel but I risk damaging the veneer . Looking for suggestions . Thanks !

XA30.8 gate resistors

I own an XA30.8 and noticed that the gate resistors for the output devices seem quite high in value (don't recall the exact value - would need to open the amp up to check, but I remember they seemed high, maybe several 100 ohm each).

Since I have previous experience with lowering gate resistors which seemed high in value with favourable results (significantly more transparent sound), I was wondering if I should do this here too. Is there anybody here who has tried this. on any recent Pass amps?

Please share your experiences!

Thanks,
Alexander

Linn Switch Mode Power Supplies (SPS/SMPS) - Schematic Overview wanted

In the meantime there are a wide range of different SMPS versions from Linn - usually suited for various Linn (pre-) amplifier models.
Until now Linn don't support any repair efforts of the user's - even the delivery of any circuit diagram is refused, because Linn claim, the replace of the whole SMPS module provides best reliability.

The actual reason, however, is to avoid under all circumstances that the presence of deficiencies on circuit design - which are clearly present - come to the public.
The main deficiencies are thermal stress and/or inferior parts (mainly bad caps, both in leaded and SMD version).

It has been increasingly common for about 15 years to offer low-quality products with a very high-quality exterior very expensive (sometimes extremely expensive) - btw in all specialist areas of products and not only in the home audio area.

The question is, how many technical guys have create schematics (reverse engineering) of their Linn power supply versions - in the kind of this "Brilliant" schematic, which I have create several years ago - go to post #5 under
Linn "LINTO" MC Preamp Schematic without SMPS - Schematic for SMPS wanted

Maybe on Linfo so as the Linn Forum are own creatings of schematics (maybe of various power supplies), unfortunately this URL's don't longer exist because Linn closed both - additional also not to find by webarchive.

2-way horn system based on the MK3B2

My speaker quest started off with grand ambitions of finding a fancy 3 or 4-way active setup. Luckily, @vineethkumar01 , the voice of reason, drilled some sense into me, urging us to go for something simple enough with better chances of success (or lesser chances of screwing up). The goal was to create a high-efficiency horn-based design, complete with active crossovers and DSP.

Enter @vineethkumar01 , the bringer of audio enlightenment, who mentioned this brand spanking new redesign with improved directivity of Yuichi Arai's 290 called the mk3b2 Radial Fin Horn, a collaborative effort from the minds of @DonVK , @docali , and @fluid (The thread can be found here). We didn't waste any time, we quickly decided that this was the one we were going to attempt. And so, we begin this project with a dash of enthusiasm and a sprinkle of uncertainty!


Particulars:

Action sequence:
  1. Build the horn first. Which will include 3d printing and CNC.
  2. Design and execute the adaptor for the CD.
  3. Design and build the BR cab to suit the horn dimensions.
  4. Extensively measure the horn and create a full spinorama.
  5. Set the crossover and tune with DSP.

Are your heatsinks sold in the market really flat?

Good day everybody.

I have many heatsinks of various sizes. Some reclaimed from older equipment and many bought from reputed sellers and eBay etc.
I check the flatness of the heatsinks using a flat ruler or square prior to mounting components.
Those recovered heatsinks are truly flat and can be reused. Those purchased from online or local shops are not that flat.
I have to sand the uneven surfaces to make any use of them.
What is your experience? Do you check them also? How you make it flat?

Regards.

Path to noiseless Linux streamer...

After now 20 years of building different DACs and playing with different streamer, I finally ended up with Andrea Mori's Clocks/ FIfo and DAC...which brought a level of realimns and transparency to my system which called for reviewing the source. Until today the reference has been an Alix 1D, Sotm Usb-Card, linear PSU and MPDpup, a very lightweight Puppy derivate where many good people fine-tuned this to the best...second best distro I heard until today has been Archphile.

Unfortunately, my hardware is old and none of these distros are maintained any longer, so I started a new approach:

  • "Universal recipe": ARM, x86, Raspberry, NeoPi: Should serve any platform and any Linux "dialect"
  • Target to play PCM-files, so no real-time application like for a musician, this is about
  • Lowest Noise primarily and lowest jitter secondaraly, Lowest latency in itself is not a reqirement at all.

The Linux Kernel brought with 5.15 a new tracer (RTLA)to trace OS-Noise and Latency in a white box approach, so you can easily make your Kernel configurations or system modifications visible....this is a HUGE help to see what does what...and we can now truely measure OS-Noise, so it is not a philosophical believe question.

Have a look just as an example on the osnoise effects of binding tasks to CPU-core vs. isolating the core vs. true tickless operation:

NOISE.png
From H. Akkan, M. Lang, L.M. Liebrock, Stepping towards noiseless Linux environment)

The RTLA-Osnoise tool is very powerful, it shows you the noise effect of anything you try per CPU-core be it threadning IRQs, isolating core, assigning and priotizing task or running a tickless Kernel. I invite anyone to try it.

Sofar there is a very strong correlation in my experiments between lowering OS-Noise and better, more direct, more transparent SQ with richer tone at the same time, some people would call it "analog" sound...you can measure as well the effects on OS-Noise of

  • isolating cores for your player
  • Using flac vs. wave
  • hardware vs software volume mixer
  • resolution of your files
  • network vs. direct stored files

and many more.

Additionally, I found that in Linux Core 5.15 (and versions after), some very interesting topics have been added for the audiophile:
  • Rewritten USB-driver for lower latency
  • Dynamic switchable PremptionModel (none/voluntary/full) at runtime
  • ALSA using HR_TImer as default

For my experiments, I started with Debian Bullseye as I can easily generate my own kernels for nearly any platform using DIetpi or Armbian as a starting point. It is really very easy and I am happy to share the how to if there is interest.

There are two different categories of settings:

A. Those which are difficult to negelect or argue about their positive efffects, any Linux streamer would benefit from them
B. Those where it is a matter of personal taste, your way of listening

Lets start with A.
These are setting which not only from a theoretical and OS-Noise-Measurement PoV make a difference, but which I found easily audible (in my system suing a N5100 fanless MiniPC):
  • usage of a linear PSU is a a prerequisite for anything.
  • usage of log2Ram or equivalent recommended
  • use the USB-Port which comes from your SOC-Cpu and goes not through an extra third-part Controller (e.g. many MiniPC which a Z8350 have the one USB3.0 port from the Z8350 and some more USB2.0 from an external chip)
  • Isolation of CPUs giving Ethernet to one core, USB to another and MPD on the third, any other IRQ or App runs on CPU 0.
  • DIsable frequency scaling of the CPU, use userspace as governor with fixed frequency
  • 1000 Hz timer frequency (which is normally only use with the Realtime PAtch, but here we use it without the overhead of the RT-Patch)
  • nohz_full (full tickless mode)
  • ALSA using HR_TImer as default
  • echo 2048 > /sys/class/rtc/rtc0/max_user_freq
  • echo 2048 > /proc/sys/dev/hpet/max-user-freq
  • MPD running as a service with FIFO-RT scheduling Prio higher than 51 and higher IO-Prio (brings OSnoise down by 30%)
  • If you can, avoid using a network. HIgh OS-Noise and directly streaming from an SSD sounds much more direct.

And here is my current list of B. topics which are audible as well, but a matter of your personal taste:
  • Preemption Model: I prefer None to Premption voluntary or Full. None Sounds more direct more resolved, but a bit dirty. Full sound a bit damped and controlled, not as spontanious, in transients but very clean and warm.
  • TSC vs HPET clocksource: Difficult. HPET is more musical, more rounded but less resolved sounding. I prefer TSC and set the KErnel option tsc=perfect to avoid any overhead.
  • These two Prememtion Model and clocksource are interesting to combine in different ways as the crystalline sound of TSC goes nicely as well with a stronger preemption model...but I like it as resolved as possible,

Thats is where my efforts stand at the moment...next on the list:
  • Compare x86 with many ARM-devices with the same Kernel-Settings
  • Start to look into I2S directly like NanopiNEo3 seems to be supported directly by the Kernel, but each I2S implementation is board specific...there might be reasons why one boards sound better over USB than I2S and on another its the opposite.
  • Compare Audio-Linux and Archphile at some point of time to this generic Debian-based solution...to see where we stand.

I am happy to share any level of detail how I did what if there is interest. I am very eager to learn from others and their findings I could try myself in my setup additionally. I am not interested at all in generic discussion about "All Sources sound the same" or "A fifo will make any source sound the same" etcetc... that is a waste of time and energy.

So, the purpose of this thread is really to get to a modern alternative to archphile or MPDpup for any platform and for any Linux-version usable. And sharing knowledge, saving you time.

"Wandering" Distortion?

I am not sure if this is the correct forum / place to ask this question. If not, mods, please move.

I am just now starting to get my feet under me when it comes to understanding some facets of even the most basic measurements, how to do them properly / consistently, and most importantly how to understand and apply what I see in all the pretty, pretty, pretty charts and graphs.

I ran into something this morning that I haven't seen before, and it fascinated me. I was measuring my first "real" amplifier that I got back from a friend recently. It carries a lot of nostalgic value to me. So, I was going to put to use all my new and wonderful DIY knowledge, clean it up and possibly give it a refurbish. I am reluctant to even recap a commercial amp, so I wanted see how it was currently performing. Overall it seems fine. 0.00 VDC offset, stable for 24 hours, and basic performance measures seem in-line with expectations.

But...

Here are a few plots of the distortion residual. The reason they're not really tight across just a few waves is that when I was watching it "zoomed out", it was clearly varying in a pattern over time, while the signal remained stable. The software sets the scale for the residual vs. signal automatically. So, the residual isn't varying dramatically (relatively speaking) over time, but it does "swing up and down" in a pattern.

In all the vastness of experience (HA!) I have measuring 6 or 7 other things around the house for fun, I haven't seen this before. All the other amps are FW clones, a BA-3 variant, and my Sony VFet. For all those amplifiers, the voltage of the distortion stayed 'flat' for lack of a better description vs. the average across a few ms wandering up and down. I wish I could describe it in more accurate terms. I can take a video also if it helps. This amp may have a servo. I am wondering if this type of behavior is associated with a servo correcting for offset... (The amp is an Aragon 2004 from circa 1995).

Why do I care and ask? I'm curious, and I haven't seen this before. We tend to look at "static" images of the residual and/or the FFT. Since I typically go with the idea that what we distinguish in amplifiers is mostly due to the distortion (all other things remaining equal)... I was wondering if I was off my rocker. If we accept or consider that a few mV of distortion with a certain pattern against a much larger signal affects our perception of the sonics of an amplifier... I am wondering if the distortion profile "wandering" up and down is normal for some amplifiers and its potential effect.

Listening tests with my ears likely won't reveal too much, but I've become fascinated with "what the distortion looks like" in a manner of speaking. Possibly a rabbit hole not worth diving into, but ...

Thanks in advance for any insight and ideas for how to understand what's happening and why...

Cheers,
Patrick

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Discussing about 8channel DSP using Camilla DSP + Raspberry PI -- Interface options

Hello,
and sorry in advance for beginning just another topic about multichannel DSP using CamillaDSP. But the questions I have are not answered so far. My intention is the implement a Stereo 4 way active loudspeaker system (8 channels in total). Initially my plan was to use analog active crossovers. But the limitations doing this are there. Hence I searched about DSP solutions and came across CamillaDSP, a great project.
Besides using very expensive multichannel sound interfaces (RME for example) I considered other solutions and hopefully get a feedback to this from the forum. My most important concern is timing accuracy. The individual channels must not have a timing error >10us (assumption) to reduce phase shift in between.
My elementary need is to have 4 I2S / Toslink IOs for using 4 fully digital AMPs (1 for each frequency band).

* according to my understanding using 4x Raspberry Pi and 4x Digi+ IO boards should do the job. The audio signal is split using a 1x4 TOSLINK splitter, all Raspi get the same input data stream, each Raspi only has its own filter and some delay to make all ways equal. As I understand the situation timing should be no issue because input and output audio is synced by the (common to all) I2S input clock.

* using 1x Raspberry Pi and 4 USB sound devices: as far as I know JACKD can be used to implement a aggregate sound device. 4x 2 channel devices give 1x 8 channel virtual device which can be used in CamillaDSP. Of course USB timing and delay is not known because of its asynchronous nature. But what can be said about the relative timing in between the USB devices? Does JACKD guarantee that there is no timing error in between them? If this is not the case what is the sense of the aggregate device?

* using 1x Raspberry Pi and HDMI 7.1 audio extractor: this implementation should give no channel relative timing error at all.

Perhaps here are some experts who can comment on this. Most likely this is a questions many others are interested in as well.
Many thanks in advance...

Stephan

HexiBase 3D Printed mini subwoofer, redux

As many in this forum will know, a couple of years ago youtuber HexiBase built several 3D printed subs using the Tang Band W3-1876S 3" subwoofer. I was very interested at the time, but the lengthy 3D print was a problem for me. I got the idea to make a version of his design that was hollow on the inside, so it would print much faster and I could then fill it with concrete. I got as far as making the 3D model, but then life got in the way and I all but forgot about the project.

Fast forward 6 months, and youtuber DIY Perks posted a video where he did pretty much what I had in mind (Fill a 3D printed speaker enclosure with something pourable) only using his own speaker design. He also use concrete, but plaster of paris with PV glue mixed in. This gave me the kick in the butt I needed, and I bit the bullet.

I ordered the Tang Band driver, finished my HexiBase redux design:
HexiBase subwoofer redux Hollow filled with plaster - Imgur.png


And kicked off the 3D print:
IMG_7839.jpeg


It took quite a while (39 hours), and it's far from the cleanest 3D print I ever did, but eventually it was finished:
IMG_7847.jpeg


After a test-fit to see if the driver actually fit it was time to fill the enclosure it with modelling plaster. I used the orbital sander in the background to vibrate the enclosure while pouring to convince any air-bubbles to move to the top:
IMG_7855.jpeg


And here's the sub after curing in its natural habitat:
HexiBase subwoofer redux Hollow filled with plaster - Imgur.jpg


And this is what the plaster-side looks like:
IMG_7858.jpeg


This is a temporary setup, since I don't have a 2.1 amplifier, or something similar. Not entirely sure what I'll do yet, but right now I'm investigating a Teensy-based DSP + I2S DAC. I don't have any measurements or anything, but even with this temporary setup I subjectively get a much better bass sound than with just the KEF Q150's I have.

Feel free to ask any questions you like. If there's interest I'll happily publish the CAD and STL files on printables.

Help choosing the right resistors

hi.
can someone explain me what resistors do i need to get for my crossover? i have 2 questions

1. do i need a 10W or 20W resistors? this are the only W options i found (woofer=4 ohm 40W RMS. full-range woofer=8OHM 10W RMS)

1696970064467.png



2. what is the meaning of this 1OHM resistor (marked in the picture)? it dose not seem to change the output on VituixCAD, this crossover is based on a design from the software lib.

this is the layout
1696970180161.png


Peace and love

My MOTU UltraLite journey

I recently purchased a motu ultralite hoping to use it to replace a behringer DCX I use as a crossover for my 4 way speakers. This post I hope will be helpful to others who want to use the motu.
The cliff notes so far is if you want to use a pi, it must be a 4. Mine is a 2GB version running 6.1.21-v8+ #1642 64 bit and not sure if
it mattered, but the motu was plugged into a USB3 slot. I'll try to answer any questions about the config if you are trying to get an ultralite to work on a pi-4.

The motu is significantly better than DCX from all reports. My plan was to use a pi-3 to drive the motu using Camilla as the brains for the crossover. The plan took a many twists and turns. Using the pi-3, recording would either go dead or freeze with a tone. Playback also would freeze with no tone although the meters on the motu would show a constant signal. Not good. I dug some, changing O/S version, tweaking ALSA, and numerous other dead ends I don't remember anymore. Next up, I tried it on my linux laptop. Competely dead. The laptop is running 4.15 kernel with ubuntu. Next up, I picked up a windows laptop to try that. That worked, but with caveats. Capture worked for 2 channels, so I did some captures of my SACD's and may use it to capture some vinyl. Output is a no go though I think. Windows shows the device as multiple devices with 2 channels each. I think for Camilla that will not work. I thought Camilla wants to see all channels as one device. Its sort of ok, as my ability to stomach windows may have ended up making it a no go anyway. After some 40 years of *nix O/S's, windows is just not me. Anyway, as it happens, I'd ordered a pi-4 back in November and Digi-key sent me a notice saying it was shipping. I'd kind of forgotten about it. So I took the SD card out if the pi-3, put it in the 4, booted and plugged in the motu. To my delight, it just worked. I'd already discovered the motu using usb audio class compliant that linux uses forces sampling rates/channel counts. 44.1/48 is 22 channels, 88.2/96 is 18 channels and 192 is 10 channels. I'd already planned on using the 96/18 channel mode which gives me access to the headphones, all 10 analog outs and 6 digital outs which I won't be using. As a test, I drove the motu with a digital sine wave and captured it using a scarlett box. After an hour, perfect sine wave still coming out of the motu. So I think whatever it did not like in the pi-3, the pi-4 hardware fixes the problem.

My next step is to get Camilla setup to what my current DCX settings are set to and start moving to the new setup. This means I first must decide balanced or SE from the output of the motu. The motu is balanced out and all my amps are single ended. My understanding is the motu provides the best performance in balanced out, and I do see a very slight degradation using REW and looping back the signal fully balanced verses only using the +/shield to drive the motu input. The difference is very slight though, THD+N is .0030 balanced versus .0032 SE. The difference is larger when only considering distortion, .00095 balanced versus .0011 SE. And low frequency noise from 20-100Hz is about 10db worse at 20Hz and trends to the same by 100Hz. So not sure there is any value in building a balanced to SE converter that may well be worse than just using the + only output of the motu

Well that is it for now. Just happy I got the box working. I was really close to giving up and selling it.

JL AUDIO A1800

Hello
I have a jl A1800 amplifier. I had all the source fets and drivers damaged. I already changed the 3525 one new and one used.
The problem is that I do not have high voltage on the 63v 3300uf capacitors
UC3525
Pin 1: 0.002
Pin 2: 5.008
Pin 3: 0.260
Pin 4: 0.430
Pin 5: 1.863
Pin 6: 3.76
Pin 7: 1.778
Pin 8: 0.085
Pin 9: 5.75
Pin 10: 0.049
Pin 11: 0.003
Pin 12: 0.001
Pin 13: 13.21
Pin 14: 0.003
Pin 15: 13.21
Pin 16: 5.08

Event ASP-8 Problem / Diagnosis

My father-in-law has a set of Event ASP-8 monitors that I've fixed before. They get so hot that the board expansion caused some of the board-edge SMD caps to fracture. I soldered on new ones years ago and added a big heatsink to the back of it, and it seemed to work pretty well for several years. I'm not an expert, but I'm the family expert, if you know what I mean. I know these monitors are plagued with problems, but he likes them and I'd like to keep them working as long as possible.

However, one of the speakers is now acting up again. Behavior is as follows: Intermittently changes randomly between wayyy too loud, a more normal volume, and totally muted. Not sure if the levels are discrete or if it's an actual random volume. It happens on both the high and low end outputs. I've run a frequency sweep with input audio and I've also isolated the woofer and the tweeter and it definitely happens on both, so I doubt it's far down the signal path. I've checked the rails from the crappy power supply and they do seem pretty steadily at +-18VDC, even when the problem is occurring, so I doubt it's the problem.

Looking at the schematic, I'm guessing it's in the front-end/initial opamp/crossover area, since that's common to both low and high frequency. It does not seem to be the input connectors. I'm guessing it's not one of the ceramic SMD caps since, in my experience, they fail open, maybe closed, or somewhat predictably oscillate, and it's also not an issue that's sensitive to pushing or tapping on different parts of the board. Also, I checked the gain knob and it seems to be fine.

My guesses are that it's the front-end/crossover op-amp that got fried/out of spec, which I guess is common on these boards. Or, it's something in its feedback path, but that seemed unlikely to me based on what I know about SMD cap failures. The resistors all seem fine. My only other guess is that, despite the signal path being somewhat isolated from other parts, there could be something wonky still happening with ground, the power supply, or something else common to both channels. The fact that it is so intermittent and random in presentation makes me think it's something with silicon, though, but I don't know if that analysis is BS or maybe I'm missing something obvious. I haven't hooked up a scope to it yet, but maybe that's the next logical step.

I have ordered a drop-in replacement opamp for the original (which is a OPA2134) to figure out for sure if it's the culprit, but I wanted to just see if anyone else has any ideas.

Thank you.

Edit: Can find the service manual at https://dokumen.tips/documents/event-asp8-service-manualpdf.html or lots of other places

Steg K204 blinking protection

Good morning everyone.
In my hands I have a STEG K2.04 with green LED on and red flashing.

Let's get to the point: The amplifier had an entire channel shorted, so I removed all 14 FETs (FQP22P10 and FQP33N10).
I also removed some transistors of the preamps as they had values out of tolerance.

In theory, since nothing is shorted anymore it should turn on, but instead it doesn't.
I then tried removing everything from the second channel too, but the problem persists.
By measuring the microcontroller(ST7LITE1) that manages the protection, I noticed that in the PROTECTION pin, +4.9V is present even if I lift that pin from the PCB. (In the K2.02 diagram I should find +5V in input not output)
I assumed it wasn't exactly the pic that was generating the problem, but by separating the switching board and the amplification board (they can be divided using screw terminals and a flat) and keeping them connected only through the flat, as soon as I apply a voltage on the REM, the amplifier starts and is no longer in protection.

I have a feeling that some RAIL voltage checking is done and for some reason it fails.
Does anyone have any idea how to test the microcontroller?

ST.JPG

Help with BrianGT chipamp power supply

Hello, I am tearing my hair out trying to figure out the wiring on this LM3886 power supply. I think that AC1 and AC2 are each a pair (0v 18v) secondaries and that the lower board connection is 18v and the one above 0v on the left of the board and the same on the right but they are not marked and I don't know how to test the MUR860. I have been going round in circles and started relatively confidently but now am completely confused. I assembled the boards 12+ years ago so would like to finish them 🙂 Any help will be hugely appreciated.

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For free pickup in NYC Area SVS 20-39 sub and external plate amp

I hadn't turned it on in 10 years so I guess I don't need it anymore. This is the unpowered SVS 20-39 cylinder sub, with an external PartsExpress (or was it Partsconnexion?) generic plate amp. I have the original paperwork for the amp, but the speaker itself was bought used. The original owner didn't use it much either. I've vacuumed the accumulated dust from it so it is clean. The amp uses the original egg cardboard box it came in as a mount. One thing I don't seem to have is the power cord for the amp, but any IEC cord will work. I had to use my computer's cord as I just couldn't find one.

It is free for pickup only in Brooklyn, NY. I will NOT ship, I will NOT deliver it. I can barely lift the thing anymore. Don't kid yourself either, the thing is big. If you live in a typical Brooklyn apartment you're not going to want this, and neither do your neighbors. The 20-39 refers to the diameter and height, or maybe hz and height, but it is 20" in diameter too and goes down to 20hz.

Free, but a decent bottle of wine couple of beers are always appreciated in trade.

For Sale AD1865NK Analog Device Genuine

I have 4 Analog Device AD1865NK that I don't need. All four AD1865NK I have since 2007, so I have no reason to believe they are not genuine and in perfect order. Asking 45€ for one. The price does not include PayPal and shipping costs. PayPal is accepted, registered mail order buyer pays the additional 5% PayPal fees. With PayPal Freund the 5% does not apply and can be sent without tracking, the shipping costs within the EU are approx. 15€. Shipping within Germany €6 Outside the EU it may vary by country. I can combined shipping. If you are interested send me a PM.

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Tesla Coil " Speaker "

Hi, I built a few tesla coils for fun in the past, nothing big or fancy. But I want to try out a tesla that can play music, came across this schematic from a ebay kit .
I know it is a "slayer exciter " that oscillates by itself, the secoundary coil providing feedback into the Base of the transistor trough the " capacitance of top load trough air " and it is " modulated by the mosfet something similar to AM radio if Im not wrong, but I have a few questions about the schematic:
  • Shouldn't D1 be from base to ground instead ? ( in LTSpice it does not work with it as in the schematic, WHY ? )
  • what does C2 do ? from Q2 to ground. ( in LTSpice it reduces substantially the output voltage, or does not work at all,. depending on the self oscialltin freq ( increasing Cx would make it work. )
  • R5 and LED 2 are some sort of " protection " for the mosfet ? ( I added a zener diode from gate to source to keep the voltages in range and not kill the mosfet, I think I put it correctly there.
  • Coil I have built long time ago, around 1000 turns 0.12mm wire . 14x3 cm plastic tube.
it has 3.3mH

Please explain to me what these components really do, and maybe help me improve it ?.
Thank you , Bruno!.

Also sorry if I posted in the wrong section .

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Hifonics A3000.1D

Hi all,

I have a hifonics alpha 3000.1D that goes directly into protect upon connection of remote power. Nothing on the board looks bad to the naked eye. All of the FETs measure OK.

This amp uses the TL494 for the power supply. I've checked all the SOT23-3 transistors near it and they all "measure" OK. Of course they could've failed in an open state.

I'll check the voltages going into the 2 opamps of the TL494 tomorrow. Does anyone have a schematic for this guy?

Attached some pictures of the amp.

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Rotel 1412 troubleshooting

I had a working ra-1412 that ended up blowing three output transistors. I have gone over and over it and I feel like I am running in circles. Transistors, emitter resistors, and everything I test seems fine. Then I discovered if I unhook e1 wire from the right main amp board, no short. E1 is connected to the phono board. Should I be looking at the phono board?

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