The following circuit was developed for use in a power amplifier that utilizes lateral MOSFETs as output devices. This version is a slight update, since the originally spec'ed BUZ501D/901D devices are no longer available. Theory of operation is straightforward but requires careful selection of component values in order to guarantee stability for all load impedances. IC1 is a high speed current mode op amp that is powered via floating +/- 12V supplies. The use of floating supplies permits the use of an op amp with a reasonable rail voltages that are independent of the MOSFET rail voltages. The op amp inputs consist of In+ from a previous amp stage and V1, which is the unbuffered MOSFET output. As with all op amps, IC1 attempts to generate an output level such that there is (ideally) zero voltage difference between In+ and V1. Therefore the buffer voltage error is dependent upon only the op amp's non-ideal characteristics. Simulations show distortion, (primarily odd harmonics) in the -110 dB range, although such low numbers must be taken with a grain of salt.

The output MOSFETs require a DC bias voltage that sets the idle current. This voltage is derived from IC1 output and the resistor network R4-R8, Pot1, and Vref1. Constant current sources CR1/CR2 effectively isolate the op amp rail voltages from the bias voltage applied to the MOSFETs. Vref1 generates a precision 2.5V that is divided by R6-R8 and Pot1. The resulting drive voltages applied to the MOSFETs are therefore IC1 output +/- 0.5 * the voltage across Pot1. Capacitors C6-C8 provide an AC path between IC1 output and the MOSFET gates.

The cutoff frequencies of both IC1 and the MOSFETs are very high, in the 20-100 MHz range. As such, it is necessary to carefully consider compensation and load decoupling networks. C4/C5 and R1/R2 set the active stage roll-off in the 3 MHz range. Selecting a different op amp or changing the compensation network can result in a lower cutoff frequency at the cost of higher distortion. As is often the case, there exists a tradeoff between distortion and stability. If the load on the MOSFETs were always resistive the previous compensation network would suffice. However such is not usually the case, hence the need for the load decoupling network. Simulations show, for the values given, that the buffer is stable for all R/C output loads. For real-life loads the load decoupling network is mandatory.

MOSFET rail voltages are shown as +/- 50V, but higher or lower voltages can be supported. A single pair of MOSFETs is shown, but multiple pairs can be used, especially for higher rail voltages. For the amplifier I am designing each channel supports 6 n-channel and 6 p-channel devices and, for +/- 70V rails, is capable of approx 500W into 4 ohms.

I'm contemplating building a PP 2A3 amplifier and I have a good 5 VDC SMPS source. Is it possible to wire a pair of 2A3 filaments in series? I don't have experience with DHTs [obviously]. Thanks in advance!

I'm designing a crossover for a 3-way and initially I was targeting a flat response and have evaluated several designs. But then I started thinking about alternative voicings, and then recalled that the human ear has an auditory sensory function. So, if I design for truly flat response, the human ear would, for example, actually perceive 2k-5k much more than other frequencies, which may sound unpleasant. Now, if I design for the human sensitivity function then it should "apparently" sound flat to an ear. Now, my question is, do sound engineers already bake in the human auditory function into their mix?

All well supported arguments appreciated.

It looks like after a bit of an absence, I am getting back into the hobby. Talk to you all soon enough.

I was very lucky getting a Valkyrie Line state preamp very cheap

The preamp came in the early 90S, it uses a combination of op amps AD744JN, AD811N, AD712JN.
Sounds really good the way it is, but I always like more performance

Here are the schematics

https://studylib.net/doc/8685097/valkyrie--a-line-stage-preamplifier.


Any recommendations ?

Hi all
Have 4 18 inch Maelstrom sealed subwoofers with 4 Hypex DS 8.0 plate amps about 12-14 years old. frequently I run a test tone to each sub via minidsp 2 by 4 Hd via RCA cables and find a sub is 5 db-10 less output my target volume of 85 db .

Sometimes the test tone is the is at 50-60 db and I think it is not working at all until I am standing next to that sub ready to power on off again.

temporary solution for the sub that is low on test tone , I have found over time that if I toggle the sub power on off rocker switch usually after 2-3 on off the sub test tone volume comes back to the 85 db. Sometimes that does not work and I have to pull out the power cable from the sub and replug, the sub end, not the wall socket. And that works to return sub to my set level.

This happens on several of the subs on a random basis, however I would say that It happens to the same sub very often, sometimes several times a week. a second sub has a similar issues and the other 2 subs very rarely.

I have raised this with Hypex service support and they are not familiar with this problem and have no suggestions.

My son is Electronics Engineer who builds and repairs amps as a hobby and he has tested plate amp on his bench and found no issues. He has a range of test equipment if there are other suggestions as to what he may check?

Thanks for any suggestions?

Regards
Kevin Australia

attempting to repair it I rubbed the pad off of the tl074 pin 2 inverted input on the second board. Where can I tap into to get the signal? Is it even used thanks

My current setup is 4 ceiling hung 2x2 styrofoam DML panels + a sub + heavy EQ. It sounds good, but is very inefficient, and with the panels' LFE ending at about 200Hz the bass is uneven throughout the space.

I want to replace the panels with conventional drivers slim enclosures. I want to keep it simple and cheap. For the tweeter I am debating using a compression driver without a horn as those seem to have great dispersion. But I imagine the overall sound is harsh. So any recommendations are welcome there. Then for the woofer I think a 6.5-7" driver will be fine. I'm just unsure of whether to keep it simple and go sealed or add a passive radiator. I mainly listen to music from the 60s-80s so rarely anything super bass heavy. I'm looking more for clarity, response and better efficiency. I'm OK with doing a little EQ as well but nothing crazy. TIA

So testing my new tube amp. I'll post schematic if this goes further. Was chasing down some hum and ended up with twisted pair going to input jacks on back panel. With a shorting jack loaded it is dead quiet. Same condition with Mogami coaxial with shield and would hum to much. Anyway everything good but then I pulled the shorting jack and went to squealing terribly. So all the guitar amps I work with use a 68k loading resistor soldered right to the input jack. Do I need to add something like that?

Hey there all,

Hoping to part ways with a few (mostly) populated PCBs from Neurochrome!

I planned to make a four-channel amp for some LX Minis but never ended up finishing the project sadly.

The boards are fully populated aside from the ICs (which are all included here) and built using good quality parts from Mouser.

If you've been looking for an easy project to finish here you go! 🙂

Just looking for the price I paid for all the parts + shipping (located in Calgary, Canada): $960cad / $700usd + shipping to you

List Movies or TV shows where you've seen high end separates or where audiophile audio equipment is displayed. This will probably be a short list. 😀

Here's my list.

The Conversation- Gene Hackman, tube gear, vinyl

Love Potion #9- Sandra Bullock, Krell gear

Indecent Proposal- Robert Redford, Not sure of gear but nice looking stuff.

Californication- David Ducovney, Avid TT and Krell amps, I think.

The Mechanic- Jason Statham, tube and vinyl gear

Daredevil (Netflix)- Vincent D'Onofrio B&W speakers

9/12 Weeks- Micky Rourke and Kim Basinger, Nakamichi Dragon? tape deck. Actually shows tape flipping sides. I rewound that part a few times. Priorities. 😉

American Psycho- Christian Bale, Looks like NAD or Rotel

Pulp Fiction- John Travolta and Uma Thurman, reel to reel tape.

Dressed to Kill- John Travolta, There was some DIY in there. 😀 I 'm pretty sure Brian DePalma was into electronics and attended MIT, if I remember correctly.

Blow Out- John Travolta, More DIY. More Brian DePalma writing and directing

Hi all,

I'm trying to put together a recording system where I can simultaneously record a mic input and the signal generated by a vibration transducer Knowles BU-27135-141, which is attached to the speaker's neck. This has been done in the voice research field to estimate the aerodynamic pressure close to the vocal folds. Anyway, I've been researching for the simplest way to implement this and converging on using a Scarlett Solo as the A/D solution. However, I'm not 100% sure if it's as simple as I'm making it and wondering if anybody here could comment on the validity before I spend money on it.

From the linked datasheet above, the signal range is +/-1.5 V with the spectral response:



The vibration is lowpass and is expected to roll off steadily from the first harmonic of the voice. So the >3 kHz amplification shouldn't hurt.

The circuit shown on the datasheet is pretty darn simple, the sensor is driven by one AA battery, and the vibration (with dc bias) is measured across the sensor output.



Because the Scarlett Solo's input appears to be AC coupled (freq range 20-20kHz), can I simply hook up these output terminal directly to the Scarlett Solo's instrument input? I think I want to use the instrument mode to keep the battery drainage minimal. But, I really don't know a whole lot about for anything analog circuit or audio recording.

I'd greatly appreciate any inputs. Thanks! -Kesh

Mostly need the four - 5 mH inductors
and
80 uF non-polar caps
Can also be close to the values not exact and I can add them together.

Might also need the other values.

If I use a Wavecor tweeter it has --
3 uF
.47-.5 mH

hi i "designed" this little headphone buffer amplifier im intrested if its of any use and should i lower the voltage, also if i could add an extra gain stage that i could turn on/pass the signal through with a switch. and also dac recommendations.

I apologize if I missed a post about this if there is one, but are there any complete kits available anymore? I have built an Audiosector LM3875 and would like to build another chipamp. I've looked but can't find anything. I love my Audiosector kit and might purchase the LM4780, but I'm just checking to see if there are other quality kits available.

If not, any recommendations on a chipamp project that doesn't come as a complete kit?

Thanks,
Mike

Remember when the radio chassis were live? You’d probably go to jail today for trying to market a product like this:-




My son bought it from a 2nd hand shop and asked me to repair it.

Into the bin it goes.

I was experimenting with an interstage driver using symmetrical CFPs. It was working pretty well so I thought I would see how it performs using the old JLH auto-bias output-stage topology. The simulations look encouraging:



I like the monotonic distribution of harmonics. THD calcs out to around .002%@1KHz, despite the circuit's relatively low OLG. The input stage has a LOT of local NFB, which helps in that regard.

I believe the other helpful factor is that the two CFP blocks use the same NPN' and PNP's so the halves are better matched. At least as far as the simulation is concerned. In a real-life application you'd probably want to match all the NPN's and PNP's: but I don't think the NPN-PNP pairs would need to be matched.

BTW the input capacitor is a relic of some earlier experiments. It really isn't needed in this case.

I tried further increasing the input circuit's transconductance by reducing R11 to 10 ohms but it didn't work out too well. Substantially increasing the OLG would mean that the 50 ohm resistors also would need to be substantially reduced in value, making the idle current less stable w/regard to temperature. And at some point the feedback network's resistance would be lower than the load presented by headphones (or speakers). Given the already-good THD, it just doesn't seem to be a worthwhile tradeoff.

Pair of Motorola branded Power Transformers. App 117V:275V with filament supply. App 2.5 X 2.75 X 3.5. Show some general wear but work fine. $75 or closest offer. $25 fixed shipping within CONUS.

In box, as new.

Never used.

£800 for the pair, or swap for either a subwoofer or a pair of TD15x drivers?

Reasonable offers considered.

HI there.

I have been a KEF fan for over 40 years starting with Calindas. After a few years of KEF 104/2 ( I also happen to be moderator on the KEF 104/2 Owner group on FB) and missing a few deals on 107/2s, I finally ended buying a good set of 107/2 this summer. After 16 hours of undoing what previous tech did to the speakers (including finding shims stuck on the voice-coil after a refoam job), they are now finally clean. Sadly, they didn't come with the KUBE 107/2, and when I found them for sale, they are at outrageous prices. High prices driven by way more demand than offer, plus a little KEF fetichism, I suppose. So, I decided to make clones. It started with a conversation with SpeakerGuru at www.hifiloudspeakers.info/speakertalk, ex KEF engineer, and I ended up with a prototype KUBE 200 that KEF engineers used for testing a new Sub and satellite system in the US around 1991. That KUBE 200 was a left over from when KEF got bought out by the Chinese group. Many good equipment found their way into the trash bin. Sometimes salvaged like this KUBE... That KUBE is no longer a 200 as the BEC (Bass Extension Curve) and input and output gain stages have been modified 33 years ago. But that's a good start to study and make clones. And it works quite good on my 107/2, and better fits them than the KUBE 104/2.

I have put my hand on brand new KUBE 200 PCBs, revision 2, and I am making a 100% clone of the KUBE 200 for myself, as well as a hybrid version of the KUBE 107/2. The later will have the BEC of the 107/2, but the tonal correction of the KUBE 200. I also have all the white papers and data for both devices sourced from SpeakerGuru. Not the limited pdf stuff that can be found on Hifi Engine.

Fascia will be CNC machined. But the box will be 3D printed since purchasing a minimum order of 20ft of 2"x6"x1/8" extruded aluminium won't work for me at that stage of the project. 18 gauge stainless steel for front and rear framework. I have modified RCA quad connectors so that the R channels get the Red plug (bought 100 of them ;-) ). That explains why on original KUBEs, the connector is red plastic. That RCA connecting block is flipped upside down. I also found 12 and 18-pin ALPS latching push button switches. Supposedly "imported from Japan". But I doubt that. Most likely knock-offs.

I don't intend to offer complete units. But will most likely sell bare boards with mechanical components, fascias and front/rear stainless steel framework. 3D print your own casing or make it out of 1/8" precious wood. DIY enthusiasts can then populate the boards with the electronics of their choice. And experts will be able to experiment. Which is why I am using sockets for the OpAmps. Some will stick to the original ones, others will feel the urge to upgrade to newer ones and recalibrate the circuits if needed. Pretty sure that tech savy members will offer to assemble complete boards for others.

I am posting a few pictures here to show progress. Any question or comment? Don't be shy.

Checking a T400-4 and it seems that some resistors were removed.
Does anyone have the schematic or photos of the area near the RCA connectors?
R1
R2
R47
R48
R49
R50
R51
R52

Hi everyone, and greets from Sweden! I have a little-but-spicy soundsystem over here that is mostly used for outdoor parties.

Here's what it looks like, looking forward to expanding it slowly 🙂

If you are planning on building this preamp, please read the Errata list on Post 777. Also, read at least Post 1 of the GB Thread.


I had been complaining to Hugh Dean (AKSA) a few weeks ago about the dearth of simple, but nice sounding preamps with muscle to swing 40 volts peak-peak. Why would we need a preamp to have so much muscle? There are quite a few unity gain current amps out there in want of a high quality preamp that can swing their outputs beyond the usual voltage limitations of a +/-15v rail supply. The Pass F4 is one that comes to mind. For the person who likes the sound of opamps (and nothing wrong with that), I recently discovered (a well-known fact by many, invented in the eaarly 1970's) that you can bootstrap an opamp with a pair of TO92 BJT's and 4 resistors to get it to work from +/-44v rails for 70v p-p swing. That was an eye opener for me, but alas, the sound of an opamp to me, just isn't enthralling. After all, my time to listen to music is limited, and when I do have time, it's SE Class A or nuthin' 🙂

So, Hugh was kind enough to make up a new preamp for me to address this need for a 40v swing SE Class A amp with guts and a nice SE Class A signature, but this time, SE Class A and very low THD. Hugh says the design is inspired by a Lender balanced input stage, but with a few tweaks added by Hugh. It is simple: just 4 BJT's and a CCS, or 5 actives if you count the BSP129 depletion mode MOSFET I employed for the CCS. Although simple, the predicted performance is quite phenomenal. Hugh provided me the basic schematic in LTSpice with a generic 12mA CCS. I had some BSP129's on hand so decided to use them. A DN2540 would also work as well and not require SMT soldering. I could not find readily LTSPice usable models for the BSP129, although a very comprehensive manufacture model is available - I asked for help over in the Software Tools Forum, and Keantoken was kind enough to take the factory model and port it to a easy to use model and symbol for me to plug into Hugh's circuit. So I owe a huge thanks to Hugh for the design, and Keantoken for the BSP129 model porting.

Here is the schematic of Hugh's Lender Preamp. As you can see, it is quite simple and I was able to veroboard prototype it quickly:


Before we go into the implementation, here are some predictions of the performance. I am assuming that the power amplifier input impedance is 27kohm, although 10k still looks very good. First at an output of 4v peak-peak and this is representative of typical very high volume levels one would ever need in a line-level preamp connected to a power amplifier with circa 20dB to 30dB gain. The distortion is exceedingly low in the ppm's and the nice thing is that it doesn't rise with frequency:


Here is the predicted FFT for 20v peak-peak output, a serious level and if driving a buffer power amp, only gets you about 6w rms with 8ohm speaakers, meh... but look at the low distortion figures (0.005%THD, the figures in parenthesis is the one I am using and think it includes THD+N?) and the nice descending higher orders from dominant 2nd order:


So, what if you need to drive 25w rms into 8ohms? Well, you need to swing 40v peak-peak. The predicted THD is still quite low and the harmonic profile is still beautiful with no higher orders going nuts. So this preamp can do it all: works well with normal high gain amps, and with unity gain current buffer amps like MOSFET followers:


For completeness, here is the predicted frequency response and phase. Nice and flat from 10Hz to 1MHz. Phase is very flat too.


Here is my quick veroboard proof of concept build. I am using a simple DC-DC step up converter followed by a CRCRC to get 44.5v PSU rail that is pretty clean, shown is the 12.5mA (125mV over 10ohms) bias setting that I was adjusting using the pot (R15) on the BSP129:


Here is the obligatory scope shot to show that indeed to makes 40v p-p. This was with an initial gain of 4.3x, which I later boosted to about 9.2x for later with still excellent results:


Here is the measured FFT for 1kHz input and 20v p-p output into a 25.3kohm load (27k plus a 10:1 voltage divider comprised of a 22k & 2k2 resistors):


Here is the measured FFT for 5kHz input and 20v p-p output into same load. As you can see, there is no rise in the THD:


The FFT signal looks low because I was using a 10:1 voltage divider to drop the signal to a safe level for the Focusrite 2i4 2nd gen audio interface.
So how does it sound? Well, I only had one channel, so I connected it to both left and right inputs a low-distortion, DC-coupled, current feedback topology(read, neutral in character), commercial headphone amp (Schiit Magni 3) and listened with some DT880-250's and also Status OB-1's. The preamp sounds amazing. Very transparent, no hint of noise, hiss, or hum. Dynamics were outstanding, and in part, the transparency of te M3 allows the nature of the preamp to come through. Cannot comment on soundstage etc as only in pseudo stereo. But I expect that it will not disappoint based on phase margin predictions and THD and noise floor.

Edit Nov 12, 2017: I found a software glitch that caused the 5th and 9th harmonic. New measurements here:

http://www.diyaudio.com/forums/soli...lender-preamp-40vpp-output-4.html#post5242220

4vpp into 25.3kohm:


20vpp into 25.3kohm:


30vpp into 25.3kohm:


I think the measurements speak for themselves, this is a superb measuring and sounding preamp.

Edit Nov 13, 2017: Looking at performance into a lower 3.4k ohm impedance load.

http://www.diyaudio.com/forums/soli...lender-preamp-40vpp-output-5.html#post5242398

Here is FFT for 4v p-p into 3.4kohm load:


Here is FFT for 20v p-p into 3.4kohm load, still very admirable performance:


Here is FFT for 36v p-p into 3.4kohm load, here the H2 and H3 are about the same and any higher in amplitude results in clip:


Here is IMD (DIN standard 250Hz and 8kHz 4:1) for 20vpp into 3.4kohm yields 0.02% IMD:


Here is IMD ((DIN standard 250Hz and 8kHz 4:1) for 20vpp into 25kohm yields 0.01% IMD:


Edit Nov 17, 2017: Here is the final design laid out in SMT with exception of 1 resistor (R10) is a carbon composition and you can find out why here. The feedback resistor is 6k8 and the LTP emitter degeneration resistors are 47R. I am now getting -85dB for H2 and -100dB for H3, and nothing else. It is a superb sounding profile - engaging, great dynamics, full of energy yet natural sounding. Here is 20vpp driving 7kohms. Noise floor is deep deep black with no hint of any mains peaks.



Preamp looks like this:


Other side:


I love how it sounds. Sublime.

Edit Nov 17, 2017 - Bimo requested boosting gain to about 12x and show distortion profile and THD. With 10R degeneration resistors on LTP and 12k carbon resistor on R10, we get this for 20vpp into 7kohms - 0.002%THD and nice profile with only H2 and H3:


For line level driving of typical amps that have some gain, no more than 4vpp will ever be needed and here the FFT and THD for 4vpp into 7kohms we get 0.0011% THD and H2 is only5.7ppm:


Edit Nov 18, 2017 - perhaps the title should be modified to 70vpp? Here is a recent simulation with 100v Vcc showing a nice result for 70vpp output into 10kohm load with a 20dB gain setting on the amp. THD is predicted to be 0.0077% with a nice profile.


To do this requires changing the input LTP devices to MMBT5401 (or 2N5401 for through hole builds) and adjust R4b to 17k to keep 2.5mA current in the LTP. Change the degen resistors R5/R6 to 3.3ohms, and R9 to 100k. Make sure all electrolytics are revised to handle max voltages of 100v on rail and output node and 63v elsewhere.

Here is schematic v2.0 for the HV amp:


Group Buy here:
http://www.diyaudio.com/forums/group-buys/315521-aksas-lender-preamp-40vpp-ouput-gb.html#post5261067

Edit Dec 10, 2017: GB boards verified to be working. Here is MB with TH and SMT daugterboards installed.



Cloesup of SMT board:


Here are measurements from the TH and SMT boards:

TH has 0.001% THD for 20vpp into kohm load:


SMT has 0.0012% THD for 20Vpp into 7kohms:


Edit Mar 3, 2018. BOM and Schematics for the GB PCBs here
http://www.diyaudio.com/forums/anal...ender-preamp-40vpp-output-48.html#post5360015

Hello all,

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More details and prices at https://reverb.com/shop/audiofile

Cheers, Ralph

https://audiofile.net.au/

To follow up with the work I did on the big sister amp, the Boyuurange/Reison A50, a viewer had a new in the box A12 dropped shipped for me to review and possibly modify it etc.

I'm going to post this spoiler, it's a hot mess.

Massive distortion was visible on an analog scope and running a THD vs power sweep showed me that what I was seeing on the analog scope was reality. It goes into hard stop, square wave producing clipping at 1/2 the rated watts and crosses the 1% THD mark at 0.015W. It is making 6-7%+ distortion at 1W and it goes past 10% before 2W. The cheap little Nobsound 6P1 out of the box performs better than this. With my DIY >$20 mods it destroys the A12 at almost 1/2 the price.

I go into more detail in the review video and I'm hopeful in future videos I can find a resolution to whatever is causing these problems. It's a shame, as I had high hopes for this little amp, and it has what you would think it takes be a decent performer, 12AX7 driver into a SEUL EL34. But it's clearly a swing and a miss out of the box.

TL;DR at this point I would not recommend buying one of these with the expectation of goodness out of the box or just basic tube rolling. I'm sure this will be controversial, but the numbers don't lie.

Boyuurange Reisong A12: Technical Review - YouTube

After more than 25 years of faithful service, it seemed that it might be time to redo my phono system. After all, I like to think that I've picked up a few tricks in the intervening years... The old system consisted of a VPI HW17-II, a Linn Ittok LVII tonearm, and a Troika cartridge. The Troika was a substitute for my previous cartridge, a Technics MM that I sorely miss. Given that the Troika cannot be retipped for anything less than a California mortgage payment, new MCs are priced like the fashion accessories that they are, and that the deck end of the phono system still worked reasonably, my attention focused toward upgrading the phono preamp and accommodating it to whatever cartridge I could dig up. That ended up being a vintage Technics MC, which was a nice find- I'm a big fan of Technics' high-end cartridges from the late '70s to mid-'80s. Second priority went to bringing the VPI up to a higher standard, but that's a story for a different time.

To celebrate the acquisition of the Technics, I decided to just chuck the old phono preamp and build a new one from scratch. The old one, a hybrid tube-FET cascode design, suffered from less-than-optimal noise and distortion, insane power supply sensitivity, and OK-not-great RIAA conformance, so the new one is designed to be better in those respects. And it turns out to be simpler than my old unit, which is a bonus.
This article will describe the preamp design and construction. An intermediate builder can duplicate this circuit in a point-to-point or perfboard build, but if you're an advanced builder, you've probably got your own way of doing things. Even if you don’t build this circuit, there are elements of the design process and result that you might find useful in considering your own designs. Beginners will probably want to use a PCB just to make sure that all is stable and quiet.

And, as usual, I will make some very unpopular design choices. Let’s go rile up the villagers!

Background

May I let you in on a little secret? All electronics are not created equal. Some parts are more critical and difficult than others. In any analog system (and music is originally and ultimately analog), there's some sort of input transducer, some sort of output transducer, and a bunch of wires and circuit elements in between. The difficult bits, and the most critical, are where the output of the electronics meets a transducer (speaker or headphones) and where a transducer meets the input of the electronics (mike amps, phono amps, tape amps). And (confession) most of the in-between is pretty trivial to do well; levels and bandwidths are generally well-defined, and there's not much “real-world” messiness.

The output part of the electronics chain has been well-handled by many fine power amp designs. In the tube world, the phono end has been much less successful. So-called classic preamps have terrible distortion and noise performance, poor drive capability, lousy headroom, and unmentionable overload recovery. Too many of the more recent designs are just lipstick on a pig; someone takes a circuit out of a tube manual, designed for the cheapest 1950s department store record consoles, slaps in a few designer caps, and proclaims it a wonder. Or a straightforward circuit is badly crippled by stuffing it full of fashion statements without regard to basic engineering. There are a few good ones out there, but not many.

I think we can do better. If you're going to do something as irrational as build a piece of tube gear, it's worth engineering it properly. And there are actually good, sound, rational reasons to use tubes for this application in the first place: headroom, overload recovery, and linearity.

Requirements

Accurate shooting requires a target. We will start by defining requirements:

1. Low noise. The cartridge chosen has chillingly low output (0.2mV/5cm/s) making this a nontrivial point. It's unreasonable to expect 100dB signal-to-noise at these levels, but it would be nice to not add much to the cartridge’s intrinsic noise.

2. Tight RIAA conformance. I don't think that 0.01dB absolute accuracy is necessary, but 0.2dB is a reasonable target. Channel-to-channel should be at least twice as close (if not better), and this conformance ought to be robust toward change in tube characteristics with age.

3. Low distortion. This is where so-called classic phono amps really do a face-plant. One of the most popular "classics" drove a heavy capacitive load off the anode of a cathode degenerated 12AX7, a very high source impedance. And because of the resultant marginal gain and the economic constraints of a two tube design, the designer threw in a bit of positive feedback. The mediocre performance, then, comes as no surprise.

4. Freedom from overload and blocking. This is another point where "classic" circuits fall flat. Let the cartridge mistrack or rattle slightly, whack the rock and lever that we laughingly call a "stylus and cantilever," and that 50kHz ding-dong resonance kicks the input stage in the groin- remember, magnetic cartridges are velocity sensitive, and though the mistrack is only momentary, that tip resonance is a lot of velocity. A minor disturbance (a swift kick) can turn into hundreds of milliseconds of overbias (rolling on the ground, gasping for breath).

5. Capability of driving a 10k load and reasonable lengths of interconnect cables. The line stage with which this phono preamp is paired has a floating 10k input. No reason that a circuit should wuss out at this load just because it’s tubes.

6. All-tube amplification. There are some decided engineering advantages to using tubes as the voltage amplification devices. At the top of the list are high linearity and overload immunity. My old unit was a hybrid using FETs in cascode with tubes (not unlike Allen Wright's excellent designs, though not claiming the same performance), and while it was good enough to live with for a quarter century, I still feel that its performance can be surpassed with an all-tube amplification lineup. Of course, I will still not hesitate to use semiconductors where they do best, i.e., for providing power, increasing power supply immunity, and controlling operating points- for constant voltages and constant currents, silicon is the way to go.

Design Requirement 1 is the toughest and most subtle one, and will drive some of the most painful design decisions.

A Short Digression on Noise

If you know your e_n from your i_n, you can pass over this section lightly.
There are basically two types of noise that electronic circuits can contribute, deterministic and random. Deterministic noise would include things like hums, buzzes, and rattles. These need to be fixed at the root-cause level, but they are fixable. Random noise is just that- random noise. Often white, sometimes pink, this is the “shhhhhhhh” background that pollutes many phono systems. Unfortunately, Nature sticks us with an irreducible minimum of noise from any device or source; the amount of noise voltage that a device contributes is always greater than or equal to

(1) V_n = √(4KTRΔf)

where R is the real part (resistive) of the source impedance, Δf is the bandwidth over which the voltage measurement is taken, K is Boltzmann’s constant, and T is the temperature or equivalent temperature. The derivation of this equation from first principles is shown in the Feynman Lectures on Physics, and is one of the most beautiful results in all of science. Admire it. Worship it. This equation will be with us until the end of the Universe because it is a fundamental consequence of the structure of matter.

Likewise, any random noise can be represented as an equivalent resistance, that is, for a random noise voltage of V measured over a range Δf, there is an equivalent ideal resistor R at a temperature T that is given by

(2) R_eq = V²/(4KTΔf)

The takeaway: noise and resistance are correlated and can be inter-converted as equivalents. A random noise source can be expressed as an equivalent resistance and vice versa.

Don’t get too fat and happy. This theoretical value is the minimum noise that a real-world component can give. Real components add both random and systematic noise to the ideal resistance. This noise is often called (logically enough) “excess noise.” For example, if you measure the noise from carbon, metal film, and wirewound resistors of the same value, you’ll find that the wirewound resistor will have noise very close to ideal, the metal film will be slightly worse, and the carbon resistor to be a bit worse yet (sometimes a LOT worse). And to add to the worries, that excess noise is dependent on current… Well, we have our eyes open.

Because the source noise is random and the various sources of noise are uncorrelated, generators in series add as noise power rather than noise voltage, i.e., square root of the sum of the squared noise voltages. So for example, suppose one has three uncorrelated noise sources in series of 1mV, 0.5mV, and 0.1mV. The total equivalent noise voltage is then √(1² + 0.5² + 0.1²) = 1.12mV. It’s notable and convenient that the noise power addition does make the largest voltage noise source overwhelmingly dominant, hence one doesn‘t need to do terribly precise calculations to get a good estimate of noise in complex circuits- simplifications will help.

A nice little calculational shortcut: if we plug room temperature and a 20k bandwidth into the noise equations, we can express the noise voltage-resistance relation as

(3) V_n = 1.8 x 10^-8 √R

Or

(4) R = V_n² (3.1x 10¹⁷)

And because of the RIAA equalization, any pre-equalization noise is reduced by the downward slope of the transfer function. The math is handled very nicely in a National Semiconductor application manual (Audio Radio 1980, Appendix), but the bottom line is that for a white noise source (characteristic of resistors), the effective bandwidth is reduced from 20kHz to 120Hz. So since V is proportional to the square root of bandwidth,

(5) VRIAA = V_n √(120/20k) = 1.4 x 10^-9 √R

This approximation is surprisingly accurate.

So a noise source can be expressed as an equivalent resistance R or as a generator voltage V_n. There’s one more little twist- sometimes, noise is expressed as “noise voltage density,” usually symbolized by e_n. Usually given in units of voltage per square root Hertz, it’s a way of comparing noise sources without putting bandwidth into the equation. For example, an ideal 1k resistor will (by equation 3) have a noise voltage from 20Hz-20kHz of 5.7nV. Another way to characterize the noise is to use equation 2, and rearrange:

(6) e_n = V/(√Δf) = √(4KTR)

Using this formulation, the 1k resistor is said to have a noise density of 4.1nV/√Hz. One can multiply this value by the bandwidth to get a noise voltage. Using voltage noise density is useful for situations where the noise source is not a simple resistance and where the noise voltage density changes with frequency.

As an interesting exercise, we will compute the signal to noise of the cartridge itself, just to see how quiet the preamp needs to be in order that it does not add any significant noise. The cartridge resistance is 15R. Plugging that value into equation 1, we find that the cartridge has 72nV of thermal noise at room temperature over the usual 20Hz-20kHz audio bandwidth. The cartridge’s nominal output is 0.2mV. Doing a quick decibel check, signal to noise works out to -69dB. Remember that number, it will come up again.
*

Overall Topology

There are several approaches to low noise tube RIAA stages. One is the classic feedback circuit popularized by Dynaco, Marantz, and Audio Research. These circuits are… not great. The limited open loop gain means that RIAA conformance changes with tube aging or replacement, the first stage will require a noisy cathode resistor, and the output stage will drive a heavily capacitive load at high frequencies. Distortion rises at the low end, where the feedback isn’t very effective, and at the high end, where the capacitive load reduces the available open loop gain and in some cases causes slewing distortion. What was it that Morgan Jones said about classic phono stages…?

Since we’re working with tubes, which are intrinsically more linear devices than transistors and can swing considerably more voltage, there’s no reason not to use a passive equalization scheme. One quite simple implementation is the topology shown in Figure 1. In this topology, a so-called “all-in-one” RIAA equalization network is sandwiched between two gain stages. A buffer isolates the second gain stage from the load (interconnects and whatever you’re using for a line preamplifier). The all-in-one RIAA network achieves the 3180us, 318us, and 75us time constants in the RIAA standard; working equations for this circuit are given in a useful paper by Lipshitz, or you can use a handy-dandy on-line calculator which may be found at KAB ELECTRO ACOUSTICS to get the values of R1, R2, C1, and C2.

A common variation of this topology is splitting the 75us time constant off from the other two, then sandwiching it between the second gain stage and the buffer. In many cases, this can help with noise and headroom, but I have a sneaking suspicion that it’s done that way just to make the math easy. No matter, if the first stage has LOTS of gain and reasonably low source impedance, the all-in-one network will carry no noise disadvantage and will certainly improve the headroom of the second gain stage.


Let’s estimate the gain we’ll need overall, taking into account the RIAA equalization. At 1kHz, RIAA is down 20dB from 50Hz. So if we want (nominally) 0.5-1V out from the phono preamp at 0.2mV and 1kHz, we’ll need about 85-90 dB of gain (taking into account the loss from RIAA). No wonder so many circuits fall flat! Now, because the 0.2mV spec is at 5cm/s groove velocity and many records have modulation higher than that, we can relax the gain a few dB and make it up in the line stage, if necessary.

First Stage

OK, I'll begin by turning off half my readers- the design will use an input transformer. This "impure" approach carries too many advantages to ignore, including nearly noise-free gain, galvanic isolation, and the opportunity to run the phono cartridge in a balanced mode without doubling the tube count. Running balanced inputs is an amazingly powerful technique to reject common-mode noise and is common studio practice for low-level mike signals. This does require a nonstandard cable, but some twisted pairs and shields never hurt anyone.

The question begged is, “Can I run an MC in directly?” Mmmm, perhaps, but at a noise penalty. Let’s say that you choose a particularly quiet specimen of tube; the very best have input-referred equivalent noise resistances in the 60-100R range. Cherry pick to get a 60R. Compared to the cartridge thermal resistance (15R), that represents a 7dB degradation of the already-marginal signal-to-noise ratio.

Massive paralleling of input tubes might prove efficacious, at a stunning penalty. Four tubes in parallel will drop the equivalent resistance to 15R, reducing the degradation to 3dB, which is still not quite there. Transformer starting to look better?

Things get worse: to get that low noise out of a tube, the transconductance will be quite high, and high transconductance means oscillation is but a moment away. The solution is generally grid-stopper resistors, but they have to be 100R-1k before doing any good, providing yet another noise source, one big enough to swamp the tube’s noise. Maybe you can wave the magic ferrite beads on the grid leads and keep things stable (I can’t). But that just takes us back to the tube noise problem. Nope, if we want tubes, we want MC, and we want quiet, we need a transformer.

Once we have crossed that Rubicon, we find that the transformer brings some intrinsic advantages of its own. One advantage that do-it-yourselfers have over appliance operators is the ability to use non-standard interfaces if that will get the job done. We can design and construct for performance and not worry about some strange item that might get plugged into either end. This can pay dividends if we consider the phono cartridge, arm, cable, and preamp as a system instead of worrying that each part be universal. Out in the Real World, where billions are made or lost depending on the engineering, balanced circuits are de rigueur for transporting low level signals from Point A to Point B. So where do audiophiles use them? Why, between preamp and power amp, where signals are large and hum is easy to prevent. And high enders love doing it inside preamps, where there’s shielding, short wire runs, and controlled grounding. The weak link, the cartridge to preamp transfer of microvolts of precious signal is of necessity outside and exposed to the cruel world. Yet it is almost invariably done single ended, making a mockery of all further effort. It becomes doubly incredible when one considers that a cartridge (the odd Decca excepted) is not inherently single-ended; it can have both ends float just as well as it can have a leg staked to the ground.

The phono cartridge unquestionably needs to be run balanced. The signals are tiny, the opportunities for hum and noise pickup are huge. As a practical matter, that involves changing two hunks of wire and designing a phono stage with a balanced input. The first hunk of wire is the tonearm wiring. In a many cases, it might be thin, twisted wire, which would be ideal for balanced connection. If not, the intrepid constructor will then need to replace the arm wiring. Looking at what that would involve for my rare Linn arm, I decided that I was not intrepid, and refused to take the arm apart to check the wiring. But please do as I say, not as I do. As it happens, the short length of uncontrolled wiring in the arm didn’t cause me any undue noise problems, but every listening room is different.

The second hunk of wire is the tonearm-to-preamp cable. This is a critical hunk, since signals are small and noise lurks around every corner. One excellent suggestion from Morgan Jones is to sheath silver wire in Teflon sleeve, twist it, then slip a shield braid over the whole shebang. One of these days, I’ll do that, at least if I win the lottery and get over my fear of triboelectricity. In the meantime, I picked up a Good Trick, and found that some shielded CAT5 cable worked well. That stuff is perfect, several sets of twisted pairs, good quality wire and insulation. The pairs can be paralleled to reduce the cable resistance. And I can spend the difference between that and the silver/Teflon approach on beer and lottery tickets.

Once that balanced signal is delivered to the preamp’s input, the last remaining link is the Common Mode Rejection Ratio (CMRR) of the input. In order to do any good, a balanced signal needs a balanced input on the preamp. That input ought to have as high a common mode rejection as possible. The classic single-ended tube input will contribute nothing. A balanced, differential input will be better, but doubles the input stage noise. The transformer still seems to be our best choice.

My old preamp used vintage Peerless/Altec 4722 mike transformers, which gave a very healthy 1:32 stepup. Unfortunately, this came with a bandwidth penalty because of the interaction of the secondary with its own capacitance and the input capacitance of the first stage. Thus my previously-mentioned Hobson's choice of a FET cascode, since first stage input capacitance for any high mu triode is too high for this transformer to handle without some impressive ringing of its own, right around where the cartridge is misbehaving the same way. That's not good.

Better transformers are available, and one excellent choice is the 1:10 Sowter 8055X, which was designed to have excellent input balance. The big disadvantage for the American builder is the price - after bending over for the unfavorable dollar/pound exchange rate, then paying shipping and duty, the Jensen equivalents start looking much better. Serendipitously, a friend of mine in the process of moving ran across a spare pair of the Sowters, which he sold to me for a bargain price.

The Sowters seem much happier with capacitive loads than the Peerless units, and on the test bench, I found that from a source impedance of 15R driving the primary, a 6k8 secondary load resistor paralleled with 200pF across the secondary gave me rather beautiful square waves, free of ringing, and with a 4us rise time. That gives me quite a bit of flexibility as regards input stage capacitance- the FET can be dispensed with, and I don't need to suffer the disadvantages of cascodes (like their essentially zero power supply rejection). And the common-mode rejection was measured to be in excess of 100dB at 60Hz. All right! To make things even nicer, the secondary resistance (which determines the noise contribution from the transformer) is quite a bit lower than the competition- about 100R compared to nearly 1k in the Jensen 1:10 step-ups. This results in about a dB of lower noise, but every dB counts, doesn’t it? The nifty little attenuator I built for transformer testing is shown in Figure 2. It’s driven from the balanced output of my M-Audio Audiophile 192 soundcard, knocks down the signal level 74dB to avoid stunning the transformer, and has close to 15R source impedance like the cartridge. If you’re using a different cartridge (likely) or a different transformer, you can transform the values I used accordingly.


The 6k8 load does knock down the cartridge's output a bit, about -1.5dB to 0.15mV, but the signal to noise degrades less than that since the effective Thevenin resistance of the input system also drops from 1k6 to 1k3. But the loading still costs us a decibel of S/N.

With the transformer present and accounted for, the next most critical decision is the nature of the active part of the first stage. If we don't get it right here, we won't be able to recover later. And a good choice at this point will ease the overall design. The basic requirements here are low noise, low noise, and low noise. Secondarily, we would wish the highest gain possible- the input transformer has been kind enough to give us 20dB and we wouldn't want to let it down. We also recognize that the necessary RIAA equalization will knock down the level at 1kHz by about 20dB, so we're starting out on the wrong end of the lever. Every bit of gain will help.

The cartridge has a source resistance of 15R. This is transformed up by a factor of 100 to 1k5 by the 1:10 ratio of the input transformer. The input tube then should have an equivalent noise resistance that’s low compared to 1k5 so as not to add significant noise. It should also have a mu as high as possible to get the still-tiny signal out of the muck. That turns out to be 70-100 for practical triodes. One can get higher gain from a cascode, but an all-tube cascade that’s linear and stable is not a trivial exercise; worse yet, cascodes are superb at passing along every last bit of power supply noise, making the design exercise in ultra-low noise supplies a complex one. I think we can do as well with a classic grounded cathode triode input stage- not tricky, not glamorous, but it works.

A popular tube for this position is the 12AX7/ECC83. That can be a decent choice but requires quite a bit of design thought because of the high plate resistance (60-80k). If the passive RIAA build-out resistor isn’t large compared to the tube’s source resistance, the RIAA conformance will be dependent on the tube- not a good thing as tubes warm up, age, and change. A large build-out resistor means noise. Also not good. The 12AX7 often gets a bad rap for linearity, but the reality is that, with a very high plate load resistance (or better yet, a CCS), it has stunningly good linearity. But it does need that high plate load. Even with CCS loading, at high frequencies, the tube’s effective plate load becomes the RIAA build-out resistor, one more reason that the resistor has to be large and noisy.

One other tube traditionally used in the first hole is the 417A/5842. This tube has a very low plate resistance (1k6) and high transconductance (25mA/V), but the mu is marginal (43). If only we could combine the low plate resistance of the 5842 with the high mu of the 12AX7…

And of course, we can. The D3a is a European pentode that has become much better known in the past several years. It’s easily available, not terribly expensive ($10 is average), and can be connected in triode mode to give us a tube that’s ideal for this application. Mu in triode connection is 73 (not quite as good as a 12AX7, but 4.6dB better than a 5842), plate resistance is slightly over 2k, and the transconductance is an impressive 35mA/V. Equivalent noise resistance is 65-100R, certainly well below the 1k5 transformed source noise. It is slightly tricky to use- the high transconductance means that layout is critical and stopper resistors must be used to prevent oscillation. And there’s a lot of unit-to-unit variation, so it’s worth getting extras and doing some selection.

For the grid-stopper, I found that 100R kept things calm. And the noise contribution (compared to the 1k5 resistance of the transformed cartridge) is negligible.

Let’s choose the operating point. Looking at the D3a datasheet, we see that both plate resistance and mu vary rather steeply with current at low currents. By 20mA, they are beginning to level off, so let’s use that as a starting point. That leaves plate voltage and grid-cathode bias as the remaining interdependent variables. The bias voltage will determine the overload characteristics of the stage, so we want that to be high. But that also forces the plate voltage to be high and threatens excessive dissipation. For reasons that will soon be clear, 1.2V grid-to-cathode will work well. This results in about 140V on the plate- at 20mA, that’s a bit under 3W dissipation, so the tube is run well under its 4.5W limit and should be reliable.
Overload at the grid will probably start about 1V peak, or 0.7VRMS. Will that be sufficient? The cartridge has a 0.2mVRMS nominal output at 5cm/s. That ends up 1.5mV at the grid. Mistracking, dirt, and other vicissitudes of the LP Life will whack the stylus like little hammers, which is a lovely way to excite the high Q ultrasonic resonances to which all MCs succumb. The cartridge is a velocity transducer, so the high frequencies generate proportionately higher voltages. It’s not unreasonable to want at least 20dB of headroom above that. With our chosen operating points, we have 53dB before any input problems. I think that will do.

Now the question of bias method. Cathode bias using a resistor is a common method. But it carries the penalty of reduced gain and increased effective plate resistance. Worse, the cathode resistor contributes its own noise- the cathode is, after all, an input terminal. Bypassing can help, but requires a large cap, almost certainly an electrolytic. That can’t be good. And finally, although it’s unlikely this stage will overload, bypassed cathode resistors turn brief overloads into severe and clearly audible “choking” of a stage by extending the recovery time.

Another solution is battery bias. This is quite a good one, but I just don’t trust the stability and reliability of batteries, especially wrapped up in hot boxes full of glowing tubes.

My favorite solution, as anyone who has seen my earlier projects will know, is LED bias. The origins of this clever idea are obscure (I first saw it in the late ‘70s, proposed by Ike Eisenson), but it works like a charm. Forward biased LEDs have low dynamic impedance, low noise, high bandwidth, and provide essentially instantaneous recovery from overloads. They’re also nice troubleshooting devices- an LED that’s lit means the tube is conducting. The dynamic resistance is a function of current, so the magnitude of that and its effect needs to be considered in the design. The noise of the IR LED was too low for me to measure, probably somewhere around the Johnson noise of its dynamic impedance; even if the equivalent noise resistance were ten times the dynamic impedance, it would be negligible.


OK, we have our 1.2V bias, we’re running 20mA through it, we know from the datasheet that the plate voltage is going to be 140V, the last bit is the plate load. We can determine a good one by putting a dot on the D3a plate curves and pivoting a ruler around to find a good load line. But the most linear of all is a perfectly horizontal line, that is, a constant current. Constant current also maximizes the gain to near mu.

Constant current sources as plate loads bring some other advantages to the party. Constant current means that the variation in LED dynamic impedance with signal can be safely ignored. The plate voltage is automatically adjusted tube to tube and as tubes age to maintain the correct operating point. And power supply rejection increases dramatically- the CCS acts like an enormous (100M or more) series resistance, and that resistance forms a voltage divider with the tube’s plate resistance. So any noise from the CCS or the power supply rail is knocked down another 80-90dB or more.

The best bang-for-buck CCS also happens to have exceptionally high performance. The DN2540 depletion mode MOSFET is perfectly suited to make a simple cascode CCS with output impedances north of 100M and exceptionally low noise (14pA/√Hz, an insanely minuscule amount when multiplied and integrated with the D3a’s low plate resistance). So our basic gain block will look like the circuit in Figure 3.

We will check the input capacitance. The gain of the stage will be about 73. The grid to cathode capacitance is 7pF, the grid to plate capacitance is 2.7. Plugging gain and the latter capacitance into the Miller equation, then adding in the former capacitance and another 5pF for strays, we end up with about 210pF. Say, wasn’t that what we wanted to load the input transformer’s secondary? Hmmmm, quite a coincidence…

This exegesis on the first stage may seem overly long, but it’s the single most critical part of the design- anything wrong here will be faithfully passed down the chain and can’t be fixed.

Let’s see where we are: the input-referred equivalent resistance of the tube is 65R. Source resistance of the transformed cartridge plus transformer secondary resistance plus grid stopper is 1k5 + 100R + 100R = 65R = 1R765. From equation 1, this is equivalent to 0.76uV. Our 0.2mV signal has been transformed to a 1.5mV signal by the transformer, so our signal to noise is -68 dB. Remembering that the cartridge’s thermal noise limits the maximum obtainable signal to noise to -69dB, we’ve gotten through the first stage relatively unscathed! Our signal is now approximately 0.15mV * 10 * 73 = 113mV, which is well out of the muck and something we can deal with. At that swing, the distortion from the D3a drops to below any reasonable measurement limit. In the spirit of democracy, let’s move on and equalize!

The EQ Network

This is the easiest part. Once we make one basic decision, it’s all rote calculation. The D3a stage has an output impedance of about 2k2 or so. Referring back to Figure 1, R1 comprises the source impedance of the D3a. Adding a series resistance will make the loading on the tube kinder, limit the drift of EQ accuracy with tube aging, and provide a convenient way to trim the network for accuracy. If we choose the value of R1 to be about 10 times the plate resistance, the gain will hardly budge and neither will the distortion. A 10% drift in the tube’s plate resistance will only cause a 1% change in the effective value of R1. The only penalty is slightly more noise. We can take comfort in the observation that many well-regarded designs use much bigger (and hence noisier) resistors in this position, but we’ll quantify that momentarily. By jiggering things around a bit, we can try to get as many standard values as possible. With the total R1 (resistor plus tube) of 21k7, C1 works out to be 0u1, R2 to be 3k15, and C2 to be 33n5. Given the tube’s output resistance, R1 will be somewhere north of 18k- I used a 20k, then used large resistors (>200k) in parallel to trim it. Likewise, R2 can be 3k3, with a large resistor (100k) in parallel to trim it into place. C2 came very close to 33n, a standard value, and could also be trimmed with a few hundred pF.


Now, what will be the noise contribution of this network? In this case, because of the shunting effect of the various capacitors, the voltage noise density will have a strong frequency dependence; it won’t be just be R1, it will be somewhat smaller because of the parallel reactances of C1 and C2. There’s at least two ways to handle the complication of the shunt caps. If you’re a He Man, you’ll follow the procedure that National Semiconductor outlined in the early versions of their Audio and Radio Applications handbook. This involves dividing up the audio spectrum into frequency bands, computing noise from each band, then power summing the contributions.

The wuss way (but far more accurate) is to send a schematic to someone who can actually use SPICE (I am totally inept) and have the computer run a much more accurate simulation. Being a wuss, I pawned the task off onto a Dutch elf and ended up getting the graph of Figure 4, which shows e_n as a function of frequency.

To determine how much effect this noise source has on our circuit’s signal to noise, we unfortunately have to do some He Man math (though in truth the computer could have done this for us). First, note that because this is a log frequency plot, the rise at low frequencies has very little effect- Δf is pretty small. No matter, let’s do our sums. Divide the plot into three segments, 10-100 Hz, 100-1000 Hz, and 1000-20,000 Hz. For the first band, e_n runs between 10 and 25 nV/√Hz. Let’s approximate it by 15 nV/√Hz. Δf is 100 Hz, so the voltage contribution from that band is 0.15uV. The next band has e_n at about 6 nV/√Hz. Δf is 900 Hz, so the noise voltage contribution is 0.18uV. And finally, the last band averages out to something like 2 nV/√Hz with a Δf of 19,000 Hz, for a noise voltage contribution of 0.28 uV. Total noise from the network is then the power summation of the three sources, or 0.37uV. We have 113mV of signal which is knocked down 20dB at 1kHz by the RIAA EQ, or 11.3mV. The noise voltage from the RIAA network is then seen to be better than 90dB down from the signal. I think we need not worry about the RIAA network’s contribution to the noise!

There’s one little twiddle to this that we’ll implement in the final circuit; tune in later for The Case of the Missing Zero.

The Second Gain Stage

The hard part is done- the signal is of decent size, the equalization has been implemented, and now all we need to do is get the signal just enough bigger that we can hand off duties to the line amp. If we wish the nominal 0.15mV signal at the input to give us something like 0.5V on the output (leaving room for “hot” cut records to go significantly higher), then the 11.3mV signal at 1kHz will require the next stage to have a gain of about 35-40. This stage will be handling the biggest input signals and have to swing the most volts at its output. Simply because it will give a gain of 35 and I have had a lot of positive experience with it, the next tube in the chain will be a 6DJ8/ECC88 or one of its variants. These tubes are relatively inexpensive ($10), quite linear, and have an equivalent noise resistance of 200R-250R, well below the input noise.

Being the lazy sort, I will use the same topology as the first stage, though the bias voltage will be larger in order to give a bit more headroom. From previous work, I found that with 10mA of current and 1.7V of bias, the ECC88 was at a very nice linear point, with about 90V on the plate. So the LED in the cathode becomes a red one (1.7Vf) and the plate CCS is adjusted for 10mA. And that’s pretty much it.

A few minor details, though. First, because the transconductance is fairly high and the interelectrode capacitances are fairly low, the ECC88 will oscillate if you give it half a chance. So, don’t go without protection- use a grid stopper, preferably as tight as possible to the tube’s grid pin. Second, there’s the question of coupling the first stage, the RIAA network, and the second stage together. One popular method is RC coupling right after the first gain stage and before the RIAA network. This has the advantage of keeping high DC voltages off of the RIAA components. The disadvantage is that the network is now driven by a source that changes impedance with frequency at the low end. This can be overcome by using a HUGE coupling cap, but why bother? Let’s put the RC coupling after the RIAA network and spend the (perhaps) extra dime using 400V caps in the RIAA.

In order that the RC coupling not load down the RIAA network and attenuate the signal, we want a nice, large grid leak resistor. 1M is a safe value and barely disturbs things. The usual 0u1 coupling cap completes the picture, and is large enough that the 1M resistor is shunted by the much lower impedance of the RIAA network at frequencies above 100Hz, so contributes pretty much diddly squat to the noise.

Finally, a key requirement of this stage is overload immunity- we don’t want blocking. Fortunately, the RIAA network really hammers down the treble frequencies most likely to cause an issue. And in the midband, the overload margin is ridiculously high: 11.3mV versus slightly more than 1.6V to cause overload- that’s 40dB of cushion. You won’t see anything like 0.15mV from the cartridge at low frequencies, but even if you did, that translates to 120mV input to the second stage, well below the overload point.

Well, that was easy, wasn’t it?

The Output Buffer

There’s a temptation to take signal right off the second gain stage plate. After all, the ECC88 has a plate resistance of about 3k… Remembering Design Requirement 5 (ability to drive interconnect cables and a 10k load) should give us pause. Putting a reactance in the plate load and swinging the load line vertical will significantly reduce the gain (from 35 to 26) and increase the distortion. We don’t want that, do we? Of course not, so it’s probably a good thing to insulate that tube from the vicissitudes of the Real World by attaching a buffer.

I’ve made no secret of my affection for a properly designed cathode follower. Though there is a quasi-religious objection to this simple and wonderful circuit, the fact remains that not only does a cathode follower measure nearly perfectly, but no one (and that means NO ONE) has ever demonstrated that they could hear the effects of a competently designed cathode follower inserted into a signal chain. The objections are either from experience with badly designed followers (and there are many of those, sadly) or theoretical philosophy, which gives me heartburn. I’ve tried fancier circuits (bootstrap, mu follower, White follower) and never found one that actually worked better for the requirements of an audio preamp.

So with simplicity our key, I’ll refer you to my article on The Heretical Preamp for an overly detailed analysis of the Right Way to design a follower.
Part of the Right Way is using a high transconductance tube, since follower source impedance is inversely related to transconductance. Another part of the Right Way is using a current sink to set operating points and maximize the follower’s load. Let’s consider each of these in turn.

First, tube choice. We already used an ECC88 as the second voltage amplifier- we have a section left over, so why not use that as the follower? Transconductance is good, and the Heretical showed that with a CCS in the cathode, the distortion performance of the ECC88 is impeccable. How convenient!

Now, we have two good choices on how to set up the follower. One method is to direct couple from the plate of the second voltage amplifier to the grid of the cathode follower. This puts the cathode some 90V above ground, plenty of room for the CCS load to operate. The output is then capacitively coupled. An alternate method is to capacitively couple from the plate of the second voltage amplifier to the grid of the cathode follower, then return the CCS load to a negative voltage rail. This allows the output of the preamp to be direct-coupled and servoed, a la the Heretical. The advantage of the second method is that it replaces a large coupling capacitor on the output (on the order of 1 uF to keep the LF rolloff below 2 Hz) with a much smaller capacitor on the follower input (on the order of 0u01 for the same rolloff). The disadvantage is the need for a negative rail and considerably more complication in the circuitry.

I've opted for the first method due to simplicity, but would not argue with anyone who wanted to implement the second method instead. Either way, the stage will have a source impedance in the low hundreds of ohms, and a set of interconnects and 10k at the far end will not faze it a bit.

The blocks are arranged and outlined. Time to look at the whole package and put together a finished design.

Design Integration and Details

The schematic of the signal section is shown in Figure 5. The blocks will look completely familiar, but there's a few details that need explaining.

The implementation of the CCS loads is identical to that of the ImPasse preamp (I'm lazy and prefer to re-use good circuit blocks in all kinds of places). And like the ImPasse, each CCS is held at arms length from its tube by a resistor (R3 and R12). R5, R14, and R18 set the CCS currents, 20mA for the first stage, 10mA each for the next stage and the buffer. The values may have to be adjusted slightly, depending on the particulars of your batch of MOSFETs; this is most easily done by attaching the + end of the CCS to a power supply (24V or more) and the – end to a 100R dummy load resistor connected to ground. The 20mA CCS should be adjusted to get 2V across the dummy load, and (naturally) the 10mA CCS adjusted to get 1V. Exactitude isn't critical.


Next question- what's R9 doing in there? This part is optional- some claim that there's a missing zero at 3.18us due to the rolloff of the Neumann cutter heads used to cut many records. Huge sonic advantages are claimed by adding this Missing Zero. Others disagree. Frankly, I can't hear a difference with the resistor in circuit or shorted out (though admittedly my old ears would be lucky to hit 15kHz on a good day), but am leaving it in the schematic as an option for those who believe that they CAN hear it.

The neon bulb between cathode and grid of the follower protects the tube at turn-on by limiting the voltage between grid and cathode to about 70-80V. This isn't quite as safe as the usual reverse-biased diode, but it's probably below the flashover point and doesn't have the nonlinear capacitance of a reverse-biased semiconductor diode. The neon will fire at turn-on, then go out as things start to warm up.

The output features a cathode stopper which helps ensure stability at the expense of slightly higher output impedance.

Q2 dissipates the most power of any of the devices so should get its own little heatsink. You can do the same for Q4 and Q5, but it's not quite as critical.

Pretty straightforward!

We move to the regulator circuits, shown in Figure 6. For the HV regulators, four are needed, two at 260V, two at 160V. This allows each channel to have separate regulation. Don't separate the feeds for the second gain stage and the cathode follower - things are most stable if they share a supply. These regulators are nothing fancy, but Good Enough considering the power supply rejection afforded by the extensive use of CCS loading. They're variations of the classic two transistor regulator I used for the Heretical and, in fact, I pressed some leftover Heretical circuit boards into service. Noise is low, stability is high, and the simplicity is appealing. One might substitute a MOSFET for the pass transistor, but the cheap plain-vanilla TIP50 works very well indeed.


The regulator for the heaters is shown in Figure 7. As before, this is quite straightforward, an exercise in three pin regulation. But there are a few little twists. First among them is the seemingly odd choice of using two 6V regulators instead of one 12V regulator. This choice greatly increases common mode rejection, an Achilles' Heel of most heater power supplies; a common-mode choke will make things even better, but I just didn't have one. If you do, feel free to use it.

The more well-known twist is the use of R30 and R31 to elevate the heaters 65V above ground. This has two salutary effects - first, it reduces the heater-to-cathode strain of the cathode follower. For that tube, the cathode is roughly 95 volts above ground. Without this heater elevation, the heater-to-cathode voltage well exceeds the 50V limit for triode section one and strains the limits of triode section two. With 65 volts of elevation, the heater-cathode potential for the follower is a balmy 30V.


The second salutary effect is that heater supply noise is less likely to be injected into the first or second stage cathodes. One thing that helps is that the cathodes are pretty well tied to AC ground (not degenerated). Nonetheless, noise can couple efficiently between heater and cathode via a diode-like interaction. Elevating the heaters seems to ameliorate that coupling. C15 ties the center of it all to AC ground. In the Raw Supply section, I'll mention one more trick to rid us of heater circuit noise.

In any event, the required heater current is about 600-700mA (depending on which ECC88 variant is used). With a raw H supply of about +/-10V, each regulator dissipates about 2.6W, so should be heatsinked.

The External Power Supply

The raw supply is even more embarrassingly straightforward. It's built into a separate box using only the finest Radio Shack $10 cabinet (figure 8). I used mostly on-hand and surplus parts, but have given part numbers for some currently-available units. The HV power transformer was a lovely surplus item, potted and shielded. The one in the parts list is amazingly well-priced and should work fine. I'd avoid toroids since they are superbly efficient at coupling mains noise into the circuit.

A nice trick- use a separate transformer for the heater supply rather than the usual extra winding on the HV transformer. The reason for this is that, even using high speed soft recovery diodes, rectifier hash from the HV supply will be coupled to the heater winding, providing unwanted rattle. This is a minor effect, but we're handling microvolt signals…

If you're extremely lazy (who, me?), there's a nice cheat that works great and saves money and effort- there are lots of wall-wart supplies available surplus. Find a couple of nice heavy ones (conventional supply rather than switching brick) rated at 9V/1A; I dug mine up for $2 each. They can be fit inside the power supply box in place of the discrete circuits. Nonetheless, I have indicated more generally available parts, but don't hesitate to visit a surplus shop and improvise. Again, laziness compelled me to use a power entry module (PEM), a chunk of plastic containing an IEC inlet, power switch, fuse, and RF line filter. I dug a pile of them out of a surplus bin for maybe $3 each- you should be able to do as well. The earth ground should be securely attached to any exposed metal parts (in my case, the top plate and HV transformer).

The sharp-eyed might have noticed that the transformer in the photo only has a 200V secondary. This was boosted to 230V by connecting the mains to the lowest primary tap. The raw B+ comes in at 285-300V, depending on the transformer, which allows a nice 25-40V cushion for the regulator (the higher the better).

The schematic for the raw supply is shown in Figure 9.

Building the Signal Circuit

I freely admit that my first prototypes are usually a bit… rough. This one was no exception. The next build will be lovelier. But it does work. I found a Collins S-Line-style cabinet in the back of a surplus shop so grabbed it immediately. The front received a 5 pin DIN connector for the balanced phono, the rear took a surplus Amphenol connector for the power and RCA plugs for the single-ended output (Figure 10).

The internals are shown in Figure 11. Some digging around and filing got me a place to mount the perfboard. All signal and ground wiring was done with solid silver wire with Teflon sleeving. The schematics give a clue about the grounding- I used a combination of star and bus, with all grounds returning to a large solder tag right next to the input DIN jack.

Since the photos were taken, I've added heatsinks to the "upper" CCS transistors in the phono stage, twisted the gray and white transformer secondary leads on the lower right (I think forgetting that was one of my late-night moves), and added a 1k8/5W resistor in series with the collectors of the pass transistors for the 160V rail. I've also moved the earth ground lead from the raw supply from the chassis next to the power input plug (the blue wire) to the single point earth ground to the chassis next to the DIN input connector. This made a major difference in the noise floor, with a sharp reduction in odd harmonics of 60 Hz.

Tubes are, sadly, microphonic. This is one case where fanatic antiresonance measures and shock mounting really pay off. Me, I just put a piece of foam rubber on an old VPI turntable platform and stuck the preamp on there. No huge ringing problems, but a nice build will mechanically isolate the signal circuit so that I don't hear the constant whining about how trailer-trash the whole setup looks.

Designer components may be nice, but I just don't see the point. Use good-quality polypropylene coupling caps, carbon resistors for the gate and grid-stoppers, and 105°C-rated electrolytic caps in the power supply and regulators. As long as you don't look inside, the sound will be every bit as good as the fancy spread.

Trimming the RIAA network isn't too hard. Adjust the buildout resistor R7 to get the 1kHz response to be 20 dB below the 50Hz response. Then adjust C4 to give response at 20kHz at about -19dB with respect to 1kHz (if you're using the Missing Zero) or -19.6dB if you're not.


The cabling between power supply and signal circuit is likewise straightforward, but make sure you include a separate earth ground lead between the boxes so that all exposed metal parts are at safety (third pin) ground!

Proof of the Pudding

A noise spectrum of this box of gain, with the cartridge connected, is shown in Figure 12. It clearly follows the RIAA curve indicating that the primary noise source is indeed the cartridge and first stage. Total noise was measured over the range 20Hz to 20kHz with a 60Hz notch filter (the hum seems to be a function of how I run the cartridge-to-preamp interconnect) and found to be (unweighted) 0.16mV, which is very close to the noise prediction of -68dB with respect to 0.5V.


The 60Hz spike was bothersome and, in fact, when I turned the volume to 11 and stood very near the speakers, I could barely hear it. But I could hear it… This was one of a continuing set of lessons in Small Things Count- I discovered that I had neglected to twist together the leads between the input transformer secondary and the D3a; once that was done, the tiny bit of residual hum disappeared.

The spectrum of a 1kHz tone at 3VRMS output is shown in Figure 13; this was generated by feeding a signal through the attenuator in Figure 2, not from the actual cartridge. The second harmonic dominates at -70dB (0.03%). The THD is a strong function of the tube choice for the second voltage amplifier. For example, substitution of a 6KN8 raised the distortion to 0.06% (still predominantly second harmonic). An Amperex Bugle Boy clocked in at slightly under 0.05%, and the champion was the Siemens CCa at 0.03%. Swapping input tubes made almost no difference.


Speaking of input tubes, most D3a that I've checked needed some time to burn in before their grid current stabilized. One way to do this is to bake the tubes as suggested in "Valve Amplifiers," 3rd edition. I'm a bit leery of that method, worrying about compromising the glass-to-metal seal where the pins enter the envelope; my own choice is to burn in the tube in situ, grinning and bearing the sound until all has reached equilibrium.

Enhancements

This preamplifier has several compromises for practicality. Chief among them is the coupling methods mentioned before- an all-out version would have a direct coupled output with a servo. I won't argue with anyone who'd go in that direction.

For more gain, the second stage tube could also be a D3a. Set its CCS load at 10mA and use a 1.7V red LED in the cathode. True studs will also use a D3a for the cathode follower- the transconductance and gain give it a theoretical advantage. In the Real World, the follower's performance is already quite impeccable, so this is really more for show than for go.

Sonics

I'm not one who is much for the purple prose of audio writers. Nor did I do a rigorous double-blind level matched test comparing this preamplifier to my old one. But, my uncontrolled subjective impressions were consistent with the measurements- quiet, clean, and unobtrusive. The shattering mistracking and noise on some of my older, lousier records seems to be much less noticeable. No oddball blats, buzzes, or shrieks. It's really pretty delightful.

Acknowledgments

So many discussions with so many people about phono stages! But I really should point to John Curl, Morgan Jones, and Allen Wright (Vacuum State - High End Hifi Equipment) for many long and involved arguments that really shaped my thinking here. Many of the critical parts came out of Mr. Jones’s Locker. Tim de Paravicini declared this a "terrible" design, and I appreciate his input. Dave Dlugos (planet_10 hifi) scrounged up the Technics MC, an amazing feat (can you find an EPC100C Mk4 that’s been in stasis?). Without Cynthia Wenslow, the mechanics of this article would have been impossible; all the good photos were hers (and her copyright, used with her kind permission), lousy ones were mine. Jan Didden (Jan Didden audio diy and other human frailties place) was kind enough to run the SPICE simulation of the RIAA network and provide noise graphs, and suggested the name of this project. Following massive consumption of alcohol, I convinced Morgan Jones to read through the manuscript and make many helpful (and some downright insulting) suggestions, and for this I most sincerely thank him.

And as usual, great discussions amongst the denizens of diyAudio.com were a constant inspiration.

References and Further Reading:

Jim Hagerman, "On Reference RIAA Networks,," available at www.hagtech.com/pdf/riaa.pdf
Morgan Jones, "Valve Amplifiers," 3rd edition, Newnes 2003.
Walt Jung, audioXpress 4/09 "High Performance Current Regulators Revised"
R. Landee, D. Davis, and A. Albrecht, "Electronic Designers' Handbook," McGraw-Hill 1957.
Stanley Lipshitz, "On RIAA Equalization networks," JAES 27:6, 458, 1979.
National Semiconductor, "Audio Radio Handbook," 1980.
Allen Wright, "The Tube Preamp Cookbook," 2nd Edition, Vacuum State Electronics, 1997.
Stuart Yaniger, audioXpress 2/09 "The ImPasse Preamplifier"
Stuart Yaniger, "The Heretical preamplifier," available at SYclotron Audio The Heretical Preamp

PARTS LIST: SIGNAL CIRCUIT (one channel, two needed)

C1, C3 0u1 400V
C2 36n* 400V (33n plus trim for RIAA)
C4 2u2 250V
C5 1u 400V
D1 IR LED (1.25V)
D2 Red LED (1.7V)
D3-D4 LM329
NE Neon bulb (NE-2 or equivalent)
Q1-Q6 DN2540A
R1 6k8 0.5W
R2 100R 0.5W
R3, R11,
R12, R16 1k 0.5W
R4, R6, R13,
R14, R15, R 17,
R18, R19 300R 0.25W
R5 120R 0.25W
R7 20k* 0.5W (trim for RIAA)
R8 3k3 0.5W
R9 91R 0.5W
R10, R21 1M 0.5W
R20 150R 0.25W
T1 Sowter 8055X, 1:10 step-up transformer
V1 D3a pentode
V2 ECC88 or equivalent dual triode

PARTS LIST: HV REGULATORS (One channel, two needed)

C6, C9 0u22/400V
C7, C10 47u/400V
C8, C11 0u1/400V
Q7, Q9 TIP50A
Q8, Q10 MPSA42
R22 51k/2W
R23 160k/1W
R24, R27 8k2
R25 68k/2W
R26 220k/1W


PARTS LIST: HEATER REGULATOR (both channels, one needed)

C12, C16 0u1/100V
C13, C14,
C17, C18 220u, 25V
D5, D6
D7, D8 1N4007
R28, R32 470R
R29, R33 120R
R30 100k/2W
R31 33k/1W
U1 LM317
U2 LM337

PARTS LIST: RAW SUPPLIES (both channels, one needed)

C19, C20 4700uF/35V electrolytic capacitor
C21 47uF 450V electrolytic capacitor
C22 470uF 450V electrolytic capacitor
D9, D10 MUR4100EG fast recovery diodes (4A, 1000V)
D11-D15 5A, 100PIV (or more) rectifiers or bridge
L1 Choke, 5H/100mA or greater (Triode Electronics M100D)
R34 470k/2W
T2 Power Transformer, 250-0-250V, 100mA (Edcor XPWR001)
T3 Power transformer, 18-20VCT, 1A (Xicon 41FJ010 or Edcor PWRC20V1A-1)
PEM IEC Power entry module, 3A, with line filter, fuse, and switch

FIGURE CAPTIONS:

Figure 1. Passive EQ RIAA stage topology
Figure 2. Attenuator for measurements
Figure 3. First stage topology
Figure 4. Thermal noise from RIAA network
Figure 5. Signal circuitry
Figure 6. HV regulators
Figure 7. Heater supply regulators
Figure 8. The external raw supply
Figure 9. The raw supplies, high voltage and heater
Figure 10. Low-rent casework; a) DIN balanced in; b) RCA single-ended out
Figure 11: Low-rent innards. See text for changes since this photo.
Figure 12. Noise spectrum with cartridge attached.
Figure 13. Spectrum of 1kHz tone at 3VRMS out.

This is a complete set by Ryanj

• Dac board Distinction-1541 V3 "D3"
• I2S to Simultaneous mode board V2
• Capacitance multiplier CMx

in addition, I include

• TDA1541A that is original chip which I have removed from a CD player myself, so I know it is original.
• Toroidal transformer from TOROIDY in Audio Grade version - epoxy filled center + covered on the outside with a black Mylar tape which gives it an aesthetic appearance.

It took me some time to get the transformer with the correct parameters, and I consulted Ryanj. Eventually, I got it right. I guess this will be helpful for the next owner.

590 EUR plus shipment cost. DHL in EU should be around 17 EUR

Please see the attached pictures.

Regards
Marcin

After a long time as a spectator, I'm starting a line source project. The driver parameters measured in REW are the following:


They are Sony almost oval TV drivers and measures approximately 105mm x 50mm or 4 1/8” x 2”. With the parameters, I defined a volume of 2 liters per driver because increasing the volume has little impact on the response. As illustrated in the above image, the active area of the speaker measures 82mm x 39mm.

In VituixCAD, should I simulate it as a circular driver with Dd=82mm (largest vertical dimension) or as a rectangular driver with Height=82mm and Width=39mm?




Update:

Later, in post #50, I demonstrate that these first simulations were wrong.
In short, in the diffraction tool (F4), place only one driver. From the data obtained from this driver, the program simulates the results for the arrangement of the various drivers that you place in the driver layout tool.

Hi,

My name is bsipahi and I'm new to the technical part of audio. I've always been interested in music but now that I've got my own place I would like to get into the ins and outs of sound.

DIY Audio is very appealing to me because of the combination of theoretical and creative aspects it presents. This is why I would like to get into the topic and what better way to start than to make my own speakers.

I'm here to hopefully get some feedback on the choices I will make during this project, and in the future do the same to people starting out like me now!

Cheers

Hello,
Newly registered on the site. I worked as a production sound mixer for 25 years (on movies and TV), then I was a R&D engineer for a sound gear rental company, in Paris, France, for another 20 years. I've been retired for a couple of years, now, but still fixing audio gear to keep my neurones busy. And DIYAudio is really great to find infos and help!
Thanks for letting me join in.

It's a shame that Windows Bluetooth support is lacking, when Android carries all the popular codecs.

I've had some success with A2DP alternative software (not free), but 990Kbps doesn't run smoothly. Even 660Kbps gives breakup with PCM signals.

Does anyone have advice how to get reliable full quality LDAC Bluetooth?

For sale

A gently used pair of JMLC Azura Horn AH425. Made by Martin Seddon for me, the original owner. Throat size 1.4 inch. (SOLD)
400 USD or 560 CAD.
The cost of shipping the two horns to the Continental US is around 145 USD.

Also, a gently used pair of custom aluminum stands designed by me for the above horns. Made by Front Pannel Express, these are very precise and sturdy. Two sets of holes on the vertical posts allow positioning the horn lower or higher according to taste.

700 USD or 990 CAD. (half the original cost)

Buyer pays shipping. CAN or US only please

Thanks for looking!

Pierre

Hello good sirs,

I come to you for wisdom and advice in regards to understanding what is the tightest bass I can possibly achieve in a folded horn bass cabinet.

The main objective would be a low or even very low volume, but extremely tight ("kicky") bass that hopefully, if possible REMAINS "kicky" all the way down to 20Hz although I am definitely willing to trade some depth for additional speed and authority which is the main purpose as long as it goes pretty damn low, not necessarily all the way down. But I would like to achieve an extremely authoritative bass, I'm not sure what the possible trade offs are, so I'm not sure what I'm willing to sacrifice more - authority down low or overall speed over the entire low frequency range. I will be using digital crossover from the 20Hz to 120 at the lowest, but likely 200 maybe even as high as 400 if the cabinet happens to extend effortlessly that high. From which point I intend to take over with my tower speakers which do have pretty nice upper bass, so I will likely determine the exact crossover through experimentation.

1. What size drivers would I be best served by for the purpose? 10"-15" or something closer to 15"-18" or even larger? Somehow my experience so far makes me distrust large drivers in that they can be sluggish, but of course they do couple more strongly at the very low end and I'm not yet sure which exact frequency is most important to me, but hopefully I can get just big enough to handle the sub bass frequencies gracefully while still being extremely fast, 'get up and go' and responsive at the 80-120Hz too. Maybe it's all about the cabinet and I shouldn't focus so much on the driver diameter, please enlighten me.
2. What requirements for the cabinet should I pay most attention to? length of the horn?, less sharp corners, smoother path? Bracing and rigidity? Bigger mouth? longer throat? Sealed volume where the woofer negative wave is contained? Something else, like flare curve maybe?
3. For amplification I want to use either low powered class A single ended BJT based DIY amp with 10 to around 50 watts (which believe it or not can sound extremely immediate and alive and kicky) or some of the Crown class D amps with pretty much unlimited power, so here comes the question again if for low volume aliveness I want to use something very efficient with the strongest magnets, square profile wire, possibly low excursion etc or high power handling stuff that I can just power with more or less unlimited class D wattage and achieve the ultimate immediacy that way? Exact driver recommendations at various price points are very welcome. I'm EU based. What should I spend more on - the drivers or the amplification (or the cabinet? although that's mostly a time investment) to get actual results that deliver? Low powered amps I already have to be fair, but good class D don't cost an arm and a leg nowadays either and delivers wonderfully controlled high power to the bass.. may be a bit problematic at low volumes, but absolutely shines at high volume.

If I'm gonna be playing above 120Hz, musicality, fidelity, low distortion becomes important too. With all of those considerations, what have been the most popular designs in recent years that I should look into and learn about? Ease of fabrication is of course a great plus, but only if the sacrifices are relatively small. Are there any other questions that I should be asking instead of the above that I have not realized I should ask? I intend to play outside in an open terrain at which point I will surely be tempted to blast them pretty loud... At least that's the intended purpose - outside, but to be fair they will play 98 percent of the time - indoors in a yet undefined, but likely not huge venue at all. Hence the focus on low volume fidelity, but more than low distortion - ability to couple to air and deliver authoritativeness, 'tightness'... at frequencies 20 to 120 or 20 to 200 as the main focus. Although 100Hz crossover works as well if that's a limiting factor. Very curious about you's guys opinions and expertise.

Hi
I am Raj from Pune, India.
Although my profession is IT but my passion is electronics

Since childhood I would open all the electronic items in my home and look inside wondering how they work
I am an audio enthusiast and have many gears in audio and video.

I hope to learn from the experience of the members

Cheers

Raj

I know this is a school boy mathematics problem but I'm lazy - has anyone come across the formula to locate a scratch on a CD?

I have a CD that skips about 1 minute in on track 5 and can see 1 or 2 scratches. I would like to polish a small area where the scratch is.

Since CD data starts at the centre so measuring the distance from the start of the data to where it ends, indicated by plain silvering, and knowing total play time and time where it skips, it should possible to calculate the distance from the start of data where the scratch is. The only complication is that unlike a vinyl record that plays at constant RPM, time is not constant with distance, and that's the part that needs thought.

Anyone up for the challenge?

Hi everyone

This is my first post.
I have done a couple of speakers designed by others.
Now i decided to build my own 2way from scratch.
Drivers:
https://www.sbaudience.com/index.php/products/woofers/bianco-12mw200/

https://www.sbaudience.com/index.php/products/compression-drivers/rosso-65cdn-t/

The CD is to be mounted on a Yuichi 480 (ø35)
Crossover point is not decided yet, but around 600-1000hz

And here is my question:
Will it make sense to build a backloaded horn cabinet for the woofer, and is it even suitable for backloading?
I hope the woofer Will hit 35hz at F8 in vented cab, so thats fine for me.

Maybe the BLH could bring the woofer closer to the CD in sensitivity.

Thanks in advance!

Thomas

Hey!

I stumbled upon a software called neural-amp-modeler that can can train a model of amplifier or distortion pedal so that guitar players can use it without having the pedal/amp itself. The procedure requires recording a processed input signal, providing the software with un-processed and processed versions of the file which is about ~3 min of length and after some time it outputs a .nam file which is used by a VST plugin. One can capture the cab as well using microphone.
So I was thinking if it is possible to use it by creating a model of an active subwoofer which adds distortion to the signal and then, when one has the model, derive an inverted error signal by subtracting original signal from the one processed by the plugin (sorta predicted) and mixing in this inverted error signal into the signal chain and hoping to reduce distortion in the subwoofer?
I understand that a lot of properties will change on both long and short term basis like heating voice coil and cone sag, for example, there is still a chance that having a model of loudspeaker may improve sound. It can probably be used as smart EQ, so that we deal with frequency response and distortion. Well, in my perfect imaginary world that is 🙂

I know that Klippel does something like this:
ACTIVE REDUCTION OF NONLINEAR LOUDSPEAKER DISTORTION

...However, recent activities in loudspeaker research have developed physical models for the nonlinear mechanisms. They are the basis for digital controllers which compensate actively for loudspeaker distortion by preprocessing the electric input signal inversely. This paper gives a summary of this work and shows possible applications to active noise control.

There are a lot of papers available where Volterra kernels are used and they date back a few decades, so the idea to model a non-linear system and use a kind of pre-distorted signal is not new. I am just thinking whether todays advances in AI/NN can be utilized in fruitful ways for us, hobbyists.

Thoughts?

Hi,

I have an old amplifier from somewhere in the 1980s, a Montarbo model 164.
It used to work fine as a bass guitar amplifier years ago. Then it got moved from one house to the next and sat on the attic for many years. The attic is insulated, so although there will be some temperature variations, it will never be freezing cold nor scorching hot.

The amplifier powers up, but produces no sound.
I found the circuit diagram in a huge document of Montarbo circuit diagrams on archive.org [1], the correct diagram attached for your convenience.

I was able to identify that the NPN output transistor T4 is short-circuited. I measured it out-of-circuit and it is short across all three pin combinations: base-emitter, base-collector, emitter-collector. This transistor is of type TIP 33c, which is no longer produced. Its PNP complement T3 (which measures fine) is TIP 34c.

The power rail was still correct (+42V, -42V and +15V, -15V). I also measured the diodes, the T1 and T2 transistors (in-circuit) and a number of capacitors and resistors, although not exhaustively. There are no visual problem signs like burn marks or similar.

My questions are:

1. With which transistor should I replace the TIP 33c?

2. Should the complement TIP 34c also be replaced, if so with what?

3. Is it reasonable to expect the faulty T4 to be broken due to age? Or do you have a suggestion of what to check in this circuit that could explain T4 to go short-circuit?

4. There is one potentiometer on the output circuit, between pins 6 and 7 (both marked 'bias') of the LM391 IC (marked 'x2'). The LM391 datasheet says "Adjust to set bias current in the output stage." . The manual annotation on the circuit diagram indicates the bias current (Courant Repos, in French) is 6mA, which more or less corresponds to the official indication on the first page of the full pdf [1], which specifies 5.5mA for this circuit.
Am I correct that I should measure the voltage drop across one of the 0.18 ohm resistors, then tune the potentiometer until I read a voltage of about 1mV (0.18 ohm x 5.5 mA) ? (or measure across both 0.18 ohm resistors and tune to 2mV).

5. Finally, is there anything to do with respect to maintaining 0V on the speaker terminal, or will this automatically be the case once the bias current is set correctly?

Thanks a lot for your advice!

Patrick


[1] https://archive.org/details/montarbo-service-1970-2000/mode/1up (page 70)

Love this place, love big amps and big speakers!

Uk based

Below is my evaluation:
PIC12F629
pin 1 connected to VDD
pin 2 connected to PMD200 pin 18 SPI/I2C
pin 3 tied HIGH connected to PMD200 pin 26 SCL
pin 4 seems no connection
pin 5 connected to GND
pin 6 tied HIGH connected toPMD200 pin 25 SDA
pin 7 tied LOW connected to PMD200 pin 29 RESET
pin 8 connected to GND

Pulse waveforms can be observed on pin 3, 4, 6 and 7.
PIC12F629 is always working and does not go to sleep.
So that PMD200 is not properly set up at all.

Can anybody help me fix this problem?

Ok...I need help. Yes I've read and read and just got more confused.

So a little while ago I tried a basic passive with a single input, and a single output and a khozmo shunt attenuator. It worked beautifully and I was in love with the sound, so I bought a case (mine was in a cookie tin), some fancy RCA (for fun, to try) and an Elma 2 pole 6 way rotory selector.

I put this all together and...ground loops + the sound comes through independently of the volume control. As in, it comes the same if volume is at zero or max.

My explorations with multimeter seemed to make sense to what I would expect.

So I guess issue 1 is, my grounding obviously isn't working. Worked fine in my cookie tin, didn't work with exactly same configuration in my aluminum case. Confusion.

Issue 2 is, the fact that the volume control isn't working except as a pass through. I can't see any other way to wire the selector switch except what I have done. Everything I find says I did it correctly.

Image 1 is the original cookie tin that worked perfectly.

The others are how I think it should go, with some guesses as to grounding...I'm hoping for advice on this grounding. Should it be grounded to the case? I tried doing so with zero response - hum didn't change.

I feel like I've tried every configuration today and am flailing. Ie, I went back to exact original configuration and it worked again. Tried same thing but with new RCA, and it hummed. And so on and so forth...

Image 1 was original, that worked. Image 2 is another option and maybe the only I haven't tried yet. Image 3 I've tried, and failed.

It's a slow day. Time for a little entertainment - a micro project!

Someone sent me a request for a simple 1 watt amplifier.

How could I resist?

My answer:

Assuming that your preamp can put out 2 volts (and it should),
a 1 watt design would be the easiest thing in the world.

Attached is an example.

I will put it up on the Pass Labs forum at diyaudio and see
what kind of suggestions it gets for construction and improvement.

Feel free to post questions.

"Zen Mod" will likely be the first to take the bait.

The following draws 1.5A off a 19V computer desktop supply and gets
1 watt into 8 ohms at about .5% THD.

My Stereo setup includes Rose 150B as streamer and preamp running via parasound JC1 monoblocks feeding Monitor Audio Platinum 300 tower speakers
2 days back I was listening to this setup for about 30 minutes. Midway I restarted my Rose150 B while the amps were already powered up.
I heard a spike in the speakers with small blast. The bass driver coils in both the speakers burnt. There are 2 base drivers in each speaker so 4 base drivers were damaged.
The problem more likely emanated from Rose 150b as it is unlikely that 2 independent amplifiers will develop same problem at the same time.
I can rule out the power supply issue as the system is being fed from a 5KVA APC UPS.
Has any one faced similar issue. What checks I need to perform before I restart my system

I also want to know if Parasound Jc1 monoblocks have speaker protection circuit or this is something I will have to mod.

I changed the speakers and the system works ok. So at present the damage is only to the drivers.

Any help will be appreciated.

I Just stumbled upon this:

https://heissmann-acoustics.de/produkt/waveguide-ha-wg-169/

I wanted to share a quick and permanent repair for the nylon collar that always splits inside the tonearm lift mechanism on the internal cueing/anti skate asy on the SL1200 family of turntables.

As many of you already know, the part in question is part of the entire plastic plate that houses the cueing and antiskate. Its NLA and the small pin with nylon collar thats pushes on the arm lift platform was never sold separately as a service part. This piece usually cracks and splits, reducing the lift on the cueing mechanism so that the arm won't fully lift from the record.

I've come up with an easy fix for this problem. As long as the collar is only split in one spot (usually the case), it can be reattached and pinned, so that it won't slip, even if the collar is split.

What I do is drill a .037" hole into the flat side of the collar, going half way into the brass shaft. I then pin it with a cut off piece of drill bit with a similar diameter as the hole. I finally secure the pin with loctite R638 retaining compound, making sure the pin sits slightly below the surface.

I've repaired a whole bunch of these parts this way and so far haven't had any issues with the repair lasting. Usually people will break the upper cueing pad before the lift pin collar gives way.

Hope someone can benefit from this. Alot of the complete SL1200 lower arm mechanisms I see being sold on ebay have the part mentioned above already failed, so you're going to pay a big price for one that is intact. The part will however fail soon after you install it due to the way nylon ages and becomes brittle.

Hello, I am an EE/robotics student that hobbys a lot. Currently I am designing a class D amplifier PCB where I can use some valuable insights for!

Yet another soft start circuit with power zero cross detecting and switching relays in zero.

Hello,
My name is Theo from greece and i am a want to be drummer,so my main theme is audio equipment for instruments.
But i also have some Hi-Fi equipment wich i always tinder with (old school stuff) 🙂

I got some 2sk2087c vfets on TaoBao last year. But cannot get its parameters. I have to give it a try.

I solder it on my 2sk182 follower test platform, results is:

Vds about 30v;

auto bias about -2.8v;

Ids about 1.5A;

Sounds very like sk182 on my speaker clone ProAc D15.

But I haven’t found any parameter about this fet except some test pictures on the web :bbs.hifidiy.net. It’s very cheaper than sk182/180.

Have anyone known some informations about it?

Thanks a lot!

I'm a critic, but when the house smells like fire, it's too much power.
AN explosion scared me, but life goes on...

Good morning everyone 😁. I know Mark Audio Drivers have been talked about their measurements and debated on this forum and other forums as well about this subject. I did have a few questions that I need clearing up before I buy a set for a project? So here goes? (Pardon my repeated questions if they have ready been asked?).

1. The new Magnetic Arc Oxidized or (MAOP-5 Driver’s). Do they Really sound that much better next to the Alpair 5 Gen. 3 drivers? Or the 7MS drivers? I‘Ve read it’s a very expensive process to make the cones? What do they sound like? And are they worth the money? 😬.

2. So I see that some of the design for the lower end Mark Audio drivers are following suite to the more expensive drivers. Like a dimple in the middle of the dust cap? Or two round circle lines in the middle of the Pluvia drivers? Does this really make a difference in sound? Or the Dispersion of the distortion coming off the drivers dust cap? (If I wrote this right and am understanding it right?). I would enjoy seeing the before and after measuring of these two dust caps against regular Mark Audio Dust caps?

3. Last question? Or questions? When you up the line with Mark Audio are the metals better that he is using for the cones? I know they upgrade something’s and the shallow profile of the cones curved shape.

I will say that I have never heard a metal driver sound more towards the warmer side like the Mark Audios I heard. ( So use to the Fostex sound). So I do give Mark and his team credit for always trying to meet the challenges of the past full range driver issues and addressing them. Thanks for your help. Jeff

!!!!!The available Components are listed in the FIRST POST of this thread!!!!!!


Trying to clean my drawers! PRICE TO SELL!!!!! PLz help me clean up! Make me offer!

NO MORE Mills MRA-05 0.56ohm/4w Non-magnetic/non-inductive audio resistors!



Make me an offer for the boards.

Hi All,

Looking desperately for Sanyo RD-4450 cassette deck service manual. This deck is also known as Otto RD-4450. I posted message at Tapeheads forum as well but no responces yet. Ebay has plenty Sanyo cassette deck service manuals but not for this one 🙁
Deck is really gorgeous, got it for looks but I want it to be 100% functional as well. Can't stand nonfunctional equipment in my house.
Thanks a lot in advance.

Hello everyone,

I have a custom CRGOSS OPT, with Kraft paper insulation and Kraft bobbin/winding former.

It sounds pretty good, however I'm wondering if it's worth waxing it?

Specifically, what will happen to the HF, and dampening. How will it sound after waxing?

Thanks 🙏

I have an EAR834p clone, and the power supply died (burnt resistors, failed zener). I am rebuilding it. I am good with the standard PSU filtering, but this is my first time working with a silicon voltage regulator Transistor and a zener stack.

I am getting a situation where the regulated voltage ramps up to 285 pretty fast, but then continues to creep up past 310V. Zener stack is hot too. With 72V each across the 68v rated zeners. I suspect that is what led to failure.

I am getting about 360V b+ unloaded going to the Transistor, an FJP13009.

The FJP13009 data sheet says a minimum of 400v. Does that mean my input B+ is actually too low to be properly regulated to 385?

Saw this ad for Pete Millett's LR Phono as built by Beezar Audio:
https://www.ebay.com/itm/1164137451...2P18NB&hash=item1b1accc802:g:txAAAOSwlC9nVF66

The chassis is split case by Landfall audio, anodized and engraved which alone is $200+
No affiliation but looked like a good deal and is likely a very solid build

Bought a 4 amp power supply off ebay.
When I tested it the supply would reset around 3 amps.
So did a return on it, item defective.
The seller then supposedly sent me a returns label which turned out to be just an address to send to and not a prepaid return label.
So gave up and just paid for return myself.
Tried to input courier and tracking number but neither edit box would allow me to input anything !

Tried to get in touch with ebay and chat box just says try later and there is no option to get a call back.
Poor show.

Anybody know what fails on the attached driver board
Amp turns on but no audio, preamp stage all good have audio signal into the driver board but no audio out this is the 1st failure of this particular driver board I've come across

Yesterday I made my first k pipe tweeter from some old piece of 1.5" PVC pipe and a cheap p-audio 2" compression driver. This sounded so promising that I kept on tinkering. I found a larger 2.5" pipe and mounted a 4.7" full range driver with a quickly hand cut foam adapter.

The result was quite bad. When I slowly did a manual frequency sweep there I could hear lots of little resonances and crazy didgeridoo noises..

I almost abandoned the experiment but last minute decided to try stuffing. I put some polyfill in the bottom 1/3 of the pipe the result was stunning. No more resonances and only very smooth sound. It almost seems like you could use stuffing as a crossover so you could forgo a lowpass on your midrange pipe.

I also tried some stuffing in the smaller tube but that killed the high end.

If I have some more time I want to measure to verify what I hear. I also like to experiment with different materials. I have a feeling cardboard tubes may sound better than PVC.

Also I will at some point try a very large pipe just to see what happens. Maybe a 6" 5' pipe with a 8" driver and stuffing.........three pipes with a minimal crossover....only highpass filters to protect the drivers.

Have for sale:

BASH 500S - Subwoofer Amplifier - $150+shipping

Dayton SPA-500 - Subwoofer Amplifier - $150+shipping

I'll discount another $15 if you pick up locally (20817)

Thanks,
D

I am building this 3 way for my desktop. I am sure some of you have seen my previous posts.

The main build thread is here: https://www.diyaudio.com/community/...ffice-build-thread.421776/page-2#post-7890664

Can someone take a whack at notching out this 340 hz peak for me? I will attach the full data packet

I am having zero success notching out this 340 hz peak in the woofer response. This is a cause of the port resonance which is, itself, the cause of me crossing the woofer at 800 hz.

I have tried 4 different port designs and about 20 different Heimholtz resonators. I am able to kill it with heimholtz resonators but I would need 6-7 of them attached to the port for it to actually work. The Heims cause their own issues. Details of that in the thread.

My notch filter skills are minimal at best. I honestly haven't run into any super problematic issues like this to require me to develop said skills. I know how to make a basic one with a cap, an inductor, and a resistor. I tired both in series and in parallel. Neither seemed to work. I tried some notch filter calculators that used more elaborate notches but those did not work either. 3 hours in. I am asking for help.

I very much appreciate any help as always. Thank you in advance for trying, even if you do not succeed.

Short Story (kind of)

I have a Marshall JCM 900 4100 100W Dual Reverb head from aprx 1990/1991 and am trying to troubleshoot why the reverb on this amp head is not working.
See the long story for more verbose details including link to the preamp schematic, but unless I am missing something just about everything in this amp works including the reverb tanks, so my current troubleshooting assumption is that there is something wrong with the "Reverb Drive" portion of the preamp, hence why I am here.



Please correct any of these assumptions, but my top level review of the "Reverb Drive" circuit is that the three op amps used before the reverb tank are voltage followers that are there to prepare both the impedance and the current to drive the actual reverb tank.

And if I were to probe connections with an oscope in that "Reverb Drive" portion of the circuit I would be able to see a sine wave on the scope. I am aware that the components of the drive portion of the circuit traditionally are the least likely to fail (in my opinion) being that they are just run of the mill resistors, caps, and common IC for opamps, and while none of them are blackened etc a PCB that is 30+ years old could have some issues or sometime things go bye bye w/o looking fried. I also acknowledge that the more common failures to reverb are tubes, tank, and rca connections, but I view I have got past that (see Long Story below), so I will happily eat crow if my assumption this opamp area is the culprit is inaccurate.

I "tried" to keep the short story short, so many other info about what I have done and got to this point may be answered in the long story. I very much appreciate you reading any part of this and pre thanks for any replies including "you are way off be careful and learn way more before you try again". This stuff is fun!


The plan, testing, and how I'm stuck
With the amplifier on and a 1khz sine wave playing through it and heard through the speaker cab, I placed a scope ground on the chassis and the other lead to the end of C16 where C16, C17, R24, and pin3 of the TL071 join assuming that a) this was the input to the reverb drive circuit and b) that the ac signal of the sine wave would be low voltage because either C14 or C16 decoupled the high voltage DC. Unlike a successful sine wave appearing on the scope when I tried the scope leads on the input jack and from reading the "send" from the fx loop, the end result on the scope for this start of the "Reverb Drive" was what I will refer to as low mv noise and not a sine wave. I had a similar noise reading on the black lead that goes to the reverb tank, but I was unclear if the scope leads should be pin/shield or pin/ground since the BK lead to the reverb tank doesn't go to ground it goes back to R25/R26. These tests had no reverb tank connected and the BK and RD leads were loose in the event that makes these tests invalid.

My plan going in was that the connection at C16/C16/R24 was going to work OK on the scope and then the problem lay ahead in the circuit, but since it did not work I'm here seeking clarification on some additional items.

My reading of the schematic, the connection from R20 to R22 that goes around the reverb drive/recovery is for the dry signal to go to and why the amp works w/o reverb and that connection between R20 to R22 goes only in that direction and its not a "negative feedback loop".
❔ I should be able to put the scope between R20 and R22 and see an ac only (since DC decoupled) on the scope and should see a clean sine wave right?
❔ I should also be able to see a clean sine wave on any of these points too right?
Intersection of C14/R20/R23 with scope ground lead to chassis
Intersection of C14/R20/R23 with scope ground lead to chassis
Either side of R27 with scope ground lead to chassis
Either side of R28 with scope ground lead to chassis
Either side of R29 with scope ground lead to chassis
Unsure R25/R26 intersection should have a sine wav.
It is my understanding that I "could" see a sine wave at the intersection of C14 and R3, but that would have high DC voltage on it so I am preferring to stay clear of that (I know I can change the DC coupling, but avoiding for safety lower risk and not blowing up my old scope).

❔ Does the reverb tank have to be connected to test the "Reverb Drive" portion of the circuit?

❔ Please confirm that all three op amps are preparing current and that the two MC1458 aren't setup to do a push pull or inversion thing where one is handling one side of the polarity. I am basically just asking for ways to prove what works and doesn't and not necessarily a reason why this circuit is a bad design as outlined in other posts here on diyAudio.

❔ Please confirm that my secondary reverb tank that does have a different input impedance would actually be a better tank to try since it would require the drive part of the circuit to produce less current. It doesn't work either, so I am looking for confirmation that it should/could work if the amp worked.

❔ Please provide any suggestions on places to check with DMM or scope to help diagnose this issue.

❔ Please provide any background as my concern on the tip shield being isolated and feeding back to R25/R26 and not ground on the BK going to the reverb tank and how that makes it less easy for me to understand how to test that connection.


Long Story

I have a JCM 900 4100 100W Dual Reverb from aprx 1990/1991.
Everything about it is in working order except the reverb (both channel gain, both channel volume, bass, mid, treble, presense, fx loop, footswitch are all OK).

I have an seldom used background in electronics, but no specialty in guitar amp (and am aware of the safety concerns of the high voltages in such amps).

Below are the steps I used to troubleshoot it.

Troubleshooting steps...

1. There is zero reverb sound with the amp. None. Footswitch or not. Reverb 10 or 0. Both channels have their own reverb setting. Zero. If I twang the springs in the reverb tank with a chopstick they can be heard.
2. The springs in the tank are in tact (four supporting floaters and four main plate springs)
3. The connections to the RCA jacks are in tact using a beep on multimeter (tip and ring on both red and black cables)
4. The connections to the transducers in the reverb tank seem reasonable in tact (no known way to test those)
5. If I remove the tank and then connect it to my behringer umc1820 audio interface headphone output to the tank input and tank output to an input on my audio interface, a previously recorded guitar DI "sounds like it has reverb", this implies the tank is likely OK.
6. Playing the same guitar DI through the headphone audio interface output and connected to the red RCA (what would be the out from the reverb tank) on the amp itself plays and can be heard clean through the amp. Both reverb pots on the amp then act like a volume knob for this external sourced input, including turning on/off when using the reverb footswitch. This makes me assume the return connections, footswitch, etc are OK.
7. Playing the actual guitar through the amp input and then taking the black RCA (what would be the input to the reverb tank) and then connecting that to my audio interface does get some guitar input albeit needing to really crank the inputs on the audio interface and gain on the amp to hear/detect it. I barely call this a pass.
8. I don't have an impedance meter, but the DC resistance of the tank is about 50 on the input and 200 on the output.
9. The fx loop is operational
10. Everything is the same after doing all permutations for the positions of the three 12AX7 pre amp tubes.
11. The behavior is the same with another reverb tank I acquired. The tank is not an exact match however. My 8DB2C1D spec is 310/2575 input output impedance and the second tank I acquired is 600/2575. Is that really enough of a difference to hear "nothing"?

The numbers on the reverb tank are Accutronics 8DB2C1D
The numbers on the back of the amp are Y37594
The sticker on the side of the amp chassis implies March 1990 creation.

Looking at the schematic everything before the "Reverb Drive" section including the tone stack, presence, fx loop, V1a, V1b, V2a and everything after the "Reverb Drive" including the V2b, V3, power amp and putting an external audio input into the red return cable where it can be heard, adheres to the per channel attenuation from each reverb pot, it leaves me to believe that the reverb drive circuit is whacky since the BK input cables beeps pin and shield with a continuity DMM. There isn't much too the "Reverb Drive" section as its a couple of op amps TL071 and MC1458. Visually inspecting the top of of the board, none of the components look fried or problematic.

All of the components related to the reverb drive circuit pass a continuity test making the assumption that there may not be a PCB trace issue.
All of the resistors related to the reverb drive get expected ohm values within range noting the check was done with DMM on an in circuit reading.
I didn't check the caps as I am unaware of how to test those in circuit and was not about to yank those off the PCB yet. I didn't check any voltages to the IC pins while the amp was on yet (see above for the one test I tried and failed, stopped, then posted this post).

Testing was done by playing a 1khz sine wav file through computer umc 1820 audio interface out of its headphone output into the amp. The tone can be heard through the amp.
A scope successfully "sees" the 1khz sine wave at the input jack to the amp and if the scope reads the values from the FX loop Send.

I did not receive a sine wave at the C16/C16/R24 connection, so I'm here to get some clarification on how to test things further.

The preamp and power amp tubes are not original but they are 30+ years old as are the filter caps, etc. They do not appear to be microphonic.
Since I can "hear" the other aspects of the amp and swapping order of the preamp tubes makes no difference, it appears the tubes are in working order for their age.
Once I solve this reverb problem, I have additional questions/threads related to other aspects of this relic (for example there is either a very low transformer or filter cap hum can be heard not through the speaker).

The oscope I have access to probably is a dinosaur. It is an analog BK Precision 2160. It works minus any waveform I see moves to the left like an old school typewriter so I haven't found a way to have it lock in place.


References that got me here

JCM 900 Preamp Schematic from 1990
(I feel this is the schematic for my JCM900 based on date)

DIY Audio forum post from Gonecat that really helped me so far
(This forum post from this group here was very informative)
Gonecat,JMFahey,wg_ski,wahab,dotneck335 all were very helpful.


More Marshall Schematics

Marshall Guitar Amp Forum topic related to the same reverb tank I have

You Tube from Uncle Doug that helped me remember how to use an oscope

You Tube Uncle Doug another good video

You Tube Headfirst Amps showing examples of using an oscope on a tube guitar amp

Rob Elliot - Spring Reverb Unit For Guitar or Keyboards

Rob Elliot - Care and Feeding of Spring Reverb Tanks

Hi all,
I have built a pair of a Super Pensils 12, on Alpair 12p. But I`am not happy with the sound. They sounds very bright, pain sparky high range, with poor mids and average low end. That is after about 200h of playing. So I have a question to the pensils owners/users: are you got the same experience? What is best aplifier to pair with Alpair 12p? Do you have in you`re pensils any kind of baffle correction?
Thx

This Tuba PCB contains both an LC filter, to get rid of >50 kHz SMPS hash noise; AND also a of pair linear voltage regulators, to get rid of 100 Hz / 120 Hz hum. It's 2X filters. And just as 2X4 lumber is called two-by-four, this board is two-by-filters. Common mis-pronunciation is "two buh four" aka "tuba four". This board is thus "two buh filter" aka "tuba filter". Yes I know: too much information.

Each amplifier channel gets its own linear regulator, so Tuba provides an almost-dual-mono power supply arrangement. The left channel has no idea what the right channel is doing, or how loudly, so you get better stereo separation. Among other dual mono benefits.

The key component which makes Tuba possible, is the LD1084 voltage regulator IC from ST Microelectronics. It's a 5 ampere, low dropout regulator, which gives excellent hum reduction (>60 dB @ 100 Hz) without dropping much supply voltage. That's the second, downstream filter.

The first, upstream, filter is a critically damped LC network, using an inductor whose DC resistance is only 0.067 ohms. Even at the SMPS max rated current (4.4 amps), it drops less than 0.3 volts. This is paired with a "DC Link" polypropylene film capacitor, for lowest possible ESR. I suggest you actively consider the Epcos B32653A4105 , or if that's not available, the Kemet R75GI41005 . Both have truly mind boggling specs for dV/dt. The LC resonant circuit is critically damped by C2 and R1, reducing its Q well below 0.5, thereby eliminating any possibility of ringing.

Gerbers are attached below; I very strongly recommend that you order PCBs with double thickness copper ("2 ounce copper"). There's nearly 5 amperes flowing in these traces! Make them thick and let the board run cool, run safe.

There is also a thump prevention relay circuit, on page 2 of the schematics. It is just a slightly fancier implementation of the thump preventer on the N-channel VFET supply filter board, designed by Nelson Pass. The same two-pole switch ("DPST") from the Nch VFET filter board is used. One pole turns power on and off, the other pole gives the thump preventer circuit a warning that the supply is about to collapse Right This Moment.

The speakers are muted "instantly" (with no delay) at power off. And the speakers are only un-muted after a long and leisurely delay from power on. PCB jumper options J9 thru J12 let you, the builder, choose the un-mute delay after power on: either 1, 5, 10, 15, or 20 seconds. Both of the beta testers set their PCBs for 10 seconds. And both were so happy with the results, they didn't bother to try the other delay options.

Tuba's choice of relay is mostly dictated by what parts are on the shelf today and not backordered for 24 months. That means: avoid the high demand relays (with 12 volt coils) and use the low demand relays (with 24 volt coils). Also avoid the small and cute, low profile relays; instead, use the big clunky ones that take up too much PCB realestate and are too tall. So I chose 653-G2R-24-DC24. Other relays that fit the board and work well are 655-RTE24024F and 651-2961192 .

The board fits inside the Modushop VFET chassis and its mounting holes align with the ventilation slots on the bottom cover, so you don't need to do any drilling. Its mounting holes are also aligned on the same 10 x 10 mm grid as the standard steel baseplate of the Modushop non-VFET chassis such as the Deluxe.

CAUTION

The Tuba filter is definitely NOT an appropriate project for a beginning or junior-intermediate builder. There's quite a few amperes of current flowing, and a very real possibility of destroying components by setting the board up incorrectly. It's also possible to ruin the board merely by adjusting it incorrectly (!) So unless you are a confident, experienced, knowledgeable builder -- don't mess with Tuba. Let someone else blow up their lab and their one of a kind VFET amp.

Because Tuba is a project for advanced hobbyists only, there isn't much more documentation than you see here; no "training wheels" build guides or Mouser shipping carts or troubleshooting tutorials. If you think you need those, maybe Tuba isn't the right project for you just yet.

ADJUSTING THE TRIMMERS: IT'S ALL ABOUT THE IN-TO-OUT VOLTAGE DROP

Find the three test points "TP1" , "TP2" , and "TP3" on both the schematic and the PCB. Adjusting the trimmers depends critically upon the voltages between these points. Solder a piece of 22AWG solid-core wire, sticking up about 1cm above the PCB, in each of those three places. Or use explicit "test pin" parts made by Keystone or Harwin. And get out your most accurate digital voltmeter.

Before you ever apply power to the Tuba board, turn both trimmer knobs ("VR1" and "VR2") all the way clockwise, to the right. This commands the LD1084 regulators to set their output voltage to the maximum possible value. Aha! Maximum Vout gives minimum (in-to-out) voltage drop.

With no load attached, and certainly with no front end boards or amp channel boards attached, connect Tuba to the on-off switch and to the Mean Well SMPS. Turn it on and wait 60 seconds. While you're waiting you'll hear the muting relay click; that's the un-mute delay. If you don't see any flames or smell any smoking parts, proceed.

Connect your DVM positive lead to TP1 using crocodile clips. Connect DVM negative lead to TP2 using crocodile clips. Now slowly turn trimmer VR1 counterclockwise while watching the voltmeter. Keep turning until the meter reads 2.0 volts from TP1 to TP2. That's the in-to-out voltage drop of the CH1 regulator. Congratulations.

Now move the DVM negative lead to TP3 instead of TP2. Slowly turn trimmer VR2 counterclockwise while watching the voltmeter. Keep turning until the meter reads 2.0 volts from TP1 to TP3. That's the in-to-out voltage drop of the CH2 regulator. Congratulations. You've completed phase 1 of the adjustment. Now you can finish wiring up Tuba's DC outputs

PHASE 2 OF ADJUSTING THE TRIMMERS: WITH ALL BOARDS POWERED

After you've connected all of the wiring among all of the PCBs, it's time to perform the final adjustments on VR1 and VR2. Start by attaching your DVM to TP1 and TP2, and turning the amplifier on. Wait 60 seconds. If you don't see any flames or smell any smoking parts, proceed.

Check the in-to-out voltage drop from TP1 to TP2. Connecting the high power, class-A amplifier boards to the SMPS, has probably caused a change in this voltage. It's probably not 2.0 volts now, and that's normal. Slooooowly turn VR1's knob to return the in-to-out voltage back to 2.0 volts.

If turning the knob a little bit (1/4 turn) has no effect: STOP!! Your setup is somehow wrong. Your meter is connected to the wrong test points, or you are dialling the wrong trimmer, (VR2 instead of VR1), or you've made some other mistake. Don't keep turning the knob, you'll probably destroy the Tuba board. Turn power off, unplug the power cord, and take a 15 minute break. Then double and triple check your setup.

Once CH1 has got 2.0 volts from in-to-out (i.e. from TP1 to TP2), move the meter lead from TP2 to TP3 and work on CH2. Slooooowly turn VR2's knob to return the in-to-out voltage back to 2.0 volts. Done! Victory!

TUBA CONTAINS TWO FILTERS . . . BUT NOTICE WHAT IT DOESN'T CONTAIN

Tuba doesn't include great big electrolytic capacitors. All it's got is 33 microfarads on the input and 33 microfarads on the output, of each regulator IC.

There simply isn't room.

In my opinion, thanks to the Class-A load current, Tuba doesn't actually _need_ great big electrolytic capacitors on either input or output. In my opinion, Tuba will work quite well with the Class-A VFET amps and deliver great sound. And Tuba will also annihilate that 100 Hz peak on your FFT plots, whether it was audible or not.

There are probably diyA members who don't agree; folks who firmly believe that Tuba's lack of 33,000 microfarad capacitors is a fatal flaw. Which is fine; simply accept that Tuba is definitely not what you want, and move on. Or maybe perform some experiments which attach additional supply capacitance somewhere else inside the chassis, but not on the Tuba filter PCB.

Or maybe, create a new and bigger filter PCB of your own, which does contain all of the electrolytic capacitors that you believe are necessary. Perhaps including some of the elements on Tuba (damped LC filter for >50 kHz noise // LD1084 voltage regulator for 100 Hz noise // thump prevention relay) or none of the elements on Tuba. It's your drafting table, draw up whatever appeals to you.



_

The new layout looks really nice! I used to visit here often before the redesign for my speaker build. Back again to ask a few things on the electronics side this time. Happy to be back. -Kesh

I am restoring an Akai GX 255 reel to reel. It's my first attempt at restoration so I am a nubi. The left channel did not work when I started and the first thing I replace where all the C 458 transistors with the suggested replacements. When I tested it out I had the left channel sounding good but in a few seconds I could hear pops and other noises and then the it went dead. Can anyone suggest where I might start looking for the problem?

Hi everyone,

I'm Patrick from Belgium. I like to repair stuff, when I can.
At this moment, my hands are on an old (guitar) amplifier, which used to work but doesn't anymore.
I'm looking for expert advice on some specific questions I have for this repair, and looking at the forum history I'm sure to be in the right place.

~Patrick

Hi! Tobias here! I am a beginner but want to build simple pedals for my guitar. Looking for assistance and information for that.

Hi everyone, my name is Caio, i've been into electronics and mainly audio since kid, and i recently graduated as an electronics technician due to my love for this area. I've build some amps and now i'm looking forward to begin in speaker build, having studied some topics for this.

I really enjoyed discovering this forum and now i pretend to ask and share audio informations here to find help and help the others!

Hi Everyone,

My name is Tony. I've recently registered with the forum but have been doing DIY audio for a number of years. Most recently I've completed some open baffle speakers and an EAR834 phono clone. And a lot of turntable maintenance.... Right now I'm interested in DIY of quiet power supplies for tube and 5v electronics like raspi and usb dacs.

I have for sale this listed parts from National Semicondutor:
5 units - Lme49713 12€ each
5 units - LM4780 22€ each
3 units - LM4765 5€ each
5 units - LM4702C 25€ each
I have more parts from National that you could check here:
https://www.ebay.de/usr/rocksandsound
Regards

Hello guys,

I want to revive and combine two of my passions (DIY HiFi and making YT videos).

As part of this revival, I would like to contact a few audio equipment manufacturers, but unfortunately both my English and my lack of experience, especially with foreign companies, are standing in the way of this idea.

Are there any members here who have experience with both and can give me a little help?

Regards
Nik

Hi!


I got 489x537x253cm living/listening/cinema room, which means my 2 floorstanders on the longer wall stand pretty close to it (~20cm), and the L shaped couch stands pretty close to opposite wall (Probably that's why the ~35Hz rom mode is so huge - but this one is not my biggest concern of course)

https://amcoustics.com/tools/amroc?l=489&w=537&h=253&r60=0.6






With some luck I will squeeze around 7 pcs of 100x60cm/ (16cm/6.3" thick) absorbers in some corners of the room (for biggest air gap)

As I don't have a lot of space. I'm thinking about making some more 100x60cm absorbers for the ceiling (I fill fit only 10cm/4" thick absorbers + 8-10/3-4"cm air gap behind due to projector light)

Does it make sense to make the skinny ones for the ceiling?

If I make some. Whats the priority of ceiling locations? Should I focus on first reflections, or places over the speakers/ listener?


I'm thinking (have perfect space) about 2 Helmholtz Resonators - but I'll come back to it later, after the first part will be done.

Is it a good idea/plan at all?

Hello,

My first post after many years of lurking. I have been sketching up a pair of speakers with the drivers and come pretty far on how I would like to have it setup. Purifi 10" low(s) and bliesma mid / hi, sealed cab. However I would like to have the crossover part fully analog and no dsp on the signal path. Basicly I seem to have re-sketched MUM-10 speaker but without the dsp and hypex amps...
Both the skill and learning time needed to have proper crossover is way over my resources so I was wondering if there are any services available to design an crossover for a specific set of drivers in a specific cabinet.

Hi, I am enjoying my little caesar 300b but one of the D3a driver valves flashes bright at start up. Can I use a thermistor on the heater circuit? Two D3A would draw .63A at 6.3v. How do I calculate which thermistor to use? Should I put in a bypass? I know there are better solutions out there but I have very little space to work with and a switch with a thermistor is very compact! Thanks.

Hi all,
As I am doing my experiments with valves, many times I found my amplifiers oscillating.
To get a better insight of what is contributing to this, I have decided to simulate the circuit in LTSPICE, however the results I get make no sense at all.

Just to show an example, I designed a basic 2 stage 12AY7 pre-amp with the usual values for the resistors, capacitors, etc.
To do the analysis, I used the normal approach:
1 - Connected the amp in unitary feedback
2 - Broke the feedback puting a large resistor and capacitor, followed by a DC voltage source with DC=0 and AC=1
3 - AC analysis with 50 points per decade from 100Hz to 10MHz

Simulating the circuit gives me non-sense results. The poles should contribute with a 20dB/dec roll-off and 90deg/dec, which doesn't happen. It can be seen that the magnitude curve never crosses 0dB.

What am I doing wrong?
Do the models of the valves behave correctly for small signal?

Kind regards,
Pedro

Hello friends, sorry for the inconvenience. I am trying to repair this Planet Audio brand amplifier, Model AC3000.1D. I had already repaired some of this model, but this one in particular I could not get the output driver board to activate, I detected some shorted and open high and low channel transistors as well as an open 100 ohm resistor, but I cannot get it to activate. I appreciate your support, regards.

The air inductor in my speaker unraveled. Looks like it came out of the plastic cover and was glued to the bottom of the speaker.
Any ideas what happened?
Any suggestions on fixing this?

It all started nearly 20 years ago, when Gary Pimm introduced his push-pull 47 DHP amp to the audiophile community. I must've entered a mid-life crisis (which still shows no sign of abating) and decided that I had to build that.

I've assembled most of what was needed, except OPTs; PTs and chokes came from Peter Dahl of Texas (yes, I decided on monoblocks), and I talked Gary, who displayed an angelic patience in answering all of my idiotic questions, into building the CCS and servo boards for me.

Then I developed cold feet. Or, perhaps, some thing have their own timing and are guided by their own stars. I remember getting involved with vintage amps and doubting my talents to build something of equal quality. Also, I didn't have a reasonably efficient pair of speakers to match with the PP47, but now having built 3 pairs of Coral Holy Basket based speakers (one bookshelf and two Frugel horns), I told myself 'now is the time'.

Gary's CCSs still have all the juice, and I'm following his advice to run it for a month before wiring in the servos. I'm using power switching supplies to feed DC to the 47s and finemet 14K OPT from Audiofeast/Terramoto which fit nicely under the chassis.

What can I say? My heavily modified Ampex 6973 monoblocks, the creme de la creme of vintage gear do not hold a candle to these PP47s. Which fills me with daring thoughts to sell the Ampexes along with a bunch of other gear and put the money toward a stereo version of PP47 ... Tabor perhaps ... SS Tabor (see the brazenness, and no, it shan't take two decades).

Anyway, here are the pics. And let's hope that they may trickle down to Gary's attention because his old email and another that I found are not delivering.

Hi,
I found your forum when looking to build a new power supply for my Arcam rDAC.
I am definately at the beginner level in electronics though I've built a few things in the past (a bass boost for a pair of Linn Kans, a scratch filter for a record deck, a new main board for my NVA amplifier when the wife pulled the speaker cables out without turning the amp off).
I am a draughtsman by trade working in the electronics industry, and latterly a layout engineer for Wolfson Microelectronics specialising in high end audio chips.
best regards
Tom Laird