As shown here it is also missing an important component: the compensation capacitor required for high frequency stability. It goes between Q2's collector and base, and swamps miller capacitance which is parasitic to all devices, not just transistors. Leave that out and it oscillates and won't behave very nicely.

Above I showed you a preamp from a constructional project. Now I'm going to show you a very similar circuit from a commercial integrated amp...

You should be able to see the shunt-series pair of transistors (Q101, Q103) again driving a tone control stage (Q105), and this was the arrangement in numerous commercial products throughout the 1970s.

When dedicated hi-fi products started to emerge, stages like Q101, Q103 above started to be served up as modular boards within specialised line stages or preamplifiers, or incorporated in amplifiers without the tone controls as the new trend in hi-fi swept them away.

Here the stage is used much like an input valve would be used, to switch sensitivity, and equalisation for the phono input, and it dealt with all inputs. The difference being that with transistors having a more well defined and predictable gain, the switching network components proved more precise.

With an input valve which didn't have much gain, and gain differed from one valve to the next, RIAA equalisation was approximate and not very accurate. RIAA was also a new thing in the days of valves as it came to be phased in from 1953. Any deviation was compensated for by tone controls, provided the user had a clue as to how it should sound.

The shunt series pair changed all that because of its high gain and more importantly its repeatability. The negative feedback network (same as valves) could have its accuracy tightened up to give greater precision.

Because of this precision stereo was more viable because of improved left right channel matching, and stereo also brought an end to the stuffy old way of pushing speakers into room corners, and people started to arrange them into neat equilateral triangle positions to hear the stereo effect.

So thanks to transistors hi-fi stereo became much more of a reality for many more people (lest we forget).

I'm quite fascinated by this, brings back many memories.

If I might join the nostalgia, that reminds me very much of an integrated (though we didn't use that terminology then) amplifier I built in the mid 60s, from an article in good old Wireless World, perhaps by Linsley-Hood? All built point to point wired on paxolin sheet with "Vero" pins. the volume and tone controls were all stepped attenuator built on Elna wafer switches. Power amp stage was if I recall BD139/140?? PP, all housed in an aluminium U channel and perforated steel enclosure measuring about 8"x 4"x 3" with outboard power supply. Speakers were Wharfedale Kit 3, the DIY version of the Dovedale. It all went very loud and served me well until the early 80s, when a couple of caps and transistors (by then obsolete) failed, the speakers lasted well into the 90s.

Happy Days!

Ian

Ian, you are welcome to join the nostalgia, and so are others with an interest in transistor audio circuit configurations.

The above circuit was part of a Practical Wireless project, actually their 100 WPC disco:

https://www.americanradiohistory.com/Archive-Practical/Wireless/70s/PW-1975-12.pdf" rel="nofollow - https://www.americanradiohistory.com/Archive-Practical/Wireless/70s/PW-1975-12.pdf

A.C. Ainslie told me it was based on a Sinclair preamp and TR1 was added as summing amp for the mixer. Whether that is true or not I couldn't say.

The BD139/140 of your power amp would be output drivers, which could have been 2N3055 TO3 outline, which were quite popular at the time? The BD139/140's have stood the test of time, and I couldn't imagine making a driver stage without them.

We all know in low noise high performance circuitry to keep wiring as short as possible...

But every circuit has to be tried before committing to expense!

Hooked up to the AP it did 67dB S/N (CCIR 20Hz - 20kHz) and 0.01% THD and 50mV max input on a 30V lab power supply.

In reality hooked up to an Ortofon 2M Bronze it's a bit buzzy due to the less than ideal S-Dec lash-up and zero shielding.

Still, it sounds quite nice. In fact I'd best not say too much

I replaced the simple BC337's with BC184C's (discontinued these days) to obtain more bandwidth, but it didn't fully succeed. If it ever reached production I would have to use BC237's.

More on the theory as promised next up.

By the way, this is neither of the designs recently discussed, but close. It uses 4 transistors per channel.

Actually, I have to say it does sound shockingly good

It also reduces open-loop gain which has to be made up for elsewhere, and this is why my circuit has an extra transistor.

More to follow.

Here's my test circuit...

Part 2: (what shall we call part 2?)

Part 3 to follow.

How about "Real Input Impedance" and "Feedback concerns"

OK, we can call it that.

For the few who are following this topic I will continue by commenting on circuitry of real products in the 70s, in the hope we can learn how we can use these early circuits to good effect in perhaps an amplifier project?

One of the big problems with some 70s products was the use of "brilliant ideas" and how they impacted on the sound.

Often it was the case of using technology for technology's sake rather than for sound quality (in my opinion). And at least one British firm was guilty of doing so. I shall not mention who, but they have grown into a massive multi-million pound turnover retail and manufacturing outfit. Which just goes to show the general ignorance of the hi-fi buyer and those who recommend by way of review.

An excellent exposé site on the subject is by Paul Kemble: http://www.angelfire.com/sd/paulkemble/soundindex.html" rel="nofollow - http://www.angelfire.com/sd/paulkemble/soundindex.html

I also "blame" "the adventures of Doug Self" or was he simply regurgitating the ideas of the era for the delectation of Wireless World readers?

All I can say, equipped as I am today with my copy of Number One Systems "EasySPICE", is some of what we bought in the 70s (and into the 80s) was to a degree, a result of "anal retentiveness"...

No wonder the transistor has taken such a battering. But if used properly it can trounce all which valves can throw at it. But the damage has been done and valvee people will have none of it.

Yes, I think we need to make a transistor amplifier "the proper way" to enable "the few" to truly appreciate the transistor's capabilities.

So let us now begin to consider what's required.

You say you need to make a transistor amp the proper way (really make one?? For sale??) - so how would this differ from the Proprius?

Solid state can refer to transistors as well as integrated circuits such as op-amps (operational amplifiers). Transistors are therefore solid state. And integrated circuits contain a quantity of transistors all etched/deposited onto the same piece of silicon, along with any small passive components necessary, as well as diodes and zener diodes required for reference voltages.

morris_minor wrote: You say you need to make a transistor amp the proper way (really make one?? For sale??) - so how would this differ from the Proprius?

Why not? After taking a look under the lid of other 70s amps it shouldn't present much difficulty. And I have also produced one amplifier kit in the late 70s; a studio power amp which actually got used in quite a few studios; and I was R&D dogs body for an amplifier manufacturer (in fact two).

The Proprius is a bit of a unique design (erm... Bruce showed me a Dynaco who'd already done a similar voltage amplifier stage - it was news to me, honest!)

A 70s power amp could take many forms and is something I may look into in this topic, illustrating as I go.

Could even do the entire thing as a kit...

I thought it may be of interest to show you the circuit of my late 70s kit amp...

Savvy readers will note the output capacitor and also the lack of a differential input. As it uses an output capacitor it doesn't really need such precision control over its output mid point, as long as it gives the full expected output it doesn't matter if there's a few millivolts offset. Some may argue the offset increases distortion, but by how much? Often such argument is about distortion we would never hear even using high class headphones instead of a loudspeaker.

The output capacitor can cause distortion however, but only if it acts on the audio signal. A value of 1000uF was quite common in the early days, which had a turnover at 20Hz using 8 Ohms. For 4 Ohm use it was doubled up to 2200uF. It is debatable whether distortion can be heard at 20Hz because 20Hz is below what most speakers can do, and if they can, we have a job hearing it.

But phase shift relative to other frequencies might fudge timing, and 20Hz only comes back into phase at 200Hz. Perhaps nowadays we should use 10,000uF (or two 4700uF) and that would take the capacitor's effect way out of range.

DC coupling removes this problem, but adds the possibility of extreme loudspeaker damage should the amplifier "tip over" (if the output offset goes wrong). You would think that servos could overcome this problem, but I've never known it!

Just a little bit of instability seems to affect some amplifiers, blowing a pre-driver, and setting off a chain reaction ending in speaker drive unit burn out.

Therefore my preference is for output capacitors and I never heard anything untoward on a friend's Cambridge P50 driving Tangent RS4's on some really low bass. The P50 used output capacitors.

Thank you Graham for another illuminating piece about using technology the right way. As well as the Accessions I have a lot of old transistor circuits that still work fine, some built from electronics hobby magazines, and it's great to understand properly how they how they operate. For example I have a pair of 1970's active speakers with Motional Feedback, scheduled for a winter '18 refurb project and it is really good to look in the service manual at stages like the ones you have explained above.

Such periodicals as Wireless World, Practical Wireless and Practical Electronics were essential reading, and later Vance Dickason's Loudspeaker Cookbook, to name but a few.

Like Jon I have a selection of the better amps stacked on a shelf in my workshop, just can't bring myself to throw them out.

My appetite for a properly designed and engineered "modern" transistor amplifier kit, that won't reside on the aforementioned shelf, is truly whetted.

Ian

So we need to add a resistor across L1 in the diagram below (same diagram as prevoius page).

 

Given the starting point was designs from the 1970s it was my impression (as a teenager) that some of the power amps in those days were not overly load tolerant for certain speakers.  Whilst I fully appreciate you wont have a detailed working or practical knowledge of each or any of them, I am thinking here Quad 303, maybe 405, Naim's from the 70s didn't appear to like driving certain speakers and cannot remember about things like the Sugden A48 or Radfords?  I wasn't looking to draw (more powerful) later 80's amps in to this (Japanese or American for example).

 

Do you feel my impression some of these designs were not as universally compatible is a valid impression and if so were these flawed in some areas either by way of design or component limitation?

 

Thanks

I suppose I should have started this from 1966 when silicon transistors took over from germanium and prices became sensible (2/- shillings each!). But I only began to understand how to design using transistors in 1976.

Maybe I'm a bit too hard on the designers of that day? After all, they didn't have transistors that were really capable of doing the power amplifier job. In fact I'm not being hard on them, just those who followed.

The Quad 303; Sugden A48 and Radford transistor amps needed "output triples" or "doubles" with emitter-follower drive to get the required current gain. Quad and Radford knew about output inductors, requiring minimal compensation; but not Sugden which uses numerous fixes which a simple inductor would have taken care of.

The Quad 405 was a step beyond... a class B with a little class A feeding into its output node via a resistor. I remember trying an op-amp version for the headphone amp of the CCM3 mixer... it sounded awful, and I ended up using a properly biased class AB version instead. Quad reckoned the negative feedback would take care of the "abrupt" step as the signal rose to class B turn-on level, but IMO it didn't work.

As for Naim, I don't want to be sued, but will say I think they should know better.

As you mentioned the Japanese, I will say they realised our transistors were the problem in obtaining stable current gain, and developed their own 2SA/2SC range of NPN/PNP transistors, which some in the UK industry quickly adopted. Motorola (American) swiftly responded with better power output devices; some of which entered the Philips/European catalogue.

Even so, some UK engineers worked wonders with such as the 2N3055; at one time the only transistor available if you wanted to do 100 watts... but into 4 Ohms. It featured in latter Quad 303's I am told. Back in the 70s I found that epitaxial versions were unstable, but homotaxial versions worked a treat, and today researching this answer found this: http://www.amplabs.co.uk/quad%20303%20parts.htm" rel="nofollow - http://www.amplabs.co.uk/quad%20303%20parts.htm

When instability strikes the music suffers, usually the bass, and otherwise the sound is bright or brittle. The page linked confirms my feelings during the past 42 years. Glad there's not only me.

Correction: a 2N3055-like transistor was used in the 303, not a 2N3055 as I said.

3.5kg, or it will be with mounting kit. Does the transformer really need to be this big?

I decided I wanted a 68V nominal supply, and this is because I want to use my old favourite BD139/BD140 pre/driver transistors. They are rated 80V (Vceo) so I never want the supply to rise above 80V. Can it?

Transformer - rectifier relationships with a capacitive filter work thus: Vout = 1.414 x Vin, and Iout = 0.61 Iin.

So if I choose a 48 volt transformer for 230V mains, and being close to a larger than usual sub-station (a relic from an industrial age), and being in the UK where they generate 240V instead of 230V, I measured my mains voltage at 254 volts.

The transformer data also gives me off-load voltage of 51.6 volts, so at 254/230 volts that can increase by 10% to 57 volts.

If I then multiply it by 1.414 (root 2) I get 80.5 volts DC. Too much? Well, from the circuit diagram...

...we see at least one base-emitter junction in series with one of them, and that drops 0.6V, so I've just squeezed it to 0.1 volts inside its absolute maximum rating (see T7: if saturated it's collector is at 0V, and so is emitter T6 - no serious current through the emitter resistors - and so T4 emitter is T6's VBE drop above 0V. T4 is a BD139).

The chances of it actually reaching 79.9 volts are very slim indeed - it would have to be doing a full output squarewave into an open circuit, but it could happen.

What about current? I want to obtain something like 100 watts into 4 Ohms, and at least 50 watts into 8 Ohms. This is rms and so is the full power supply power...

100 watts into 4 Ohms is 20 volts rms into 4 Ohms which gives 5 amps of rms current (check: 4 Ohms times 5 amps = 20 volts). Watts is rms volts times rms current, and 20 x 5 = 100 watts.

And the transformer rectifier relationship for a full wave bridge tells me I need the inverse of 0.61 times the current required, or 1.61 x 5 = 8 amps.

Multiply the voltage by the current and it says I need a 386 VA transformer, so I bought a 400 VA transformer to be on the safe side.

But isn't there something wrong here? This is stereo so I should need double??

OK then, show me any commercial amplifier that uses an 800 VA transformer to do 100 watts per channel stereo... do they weigh in at well over 10 kilos?

The one I'm looking at spec-wise weighs 5.1kg (all-in including enclosure, and all other parts), and does 90WPC into 4 Ohms (well known British brand).

They will usually use a 300 VA transformer and allow it to run saturated, or pulled down, or overloaded or whatever language you'd like to use.

And this is because when used to play music the full power will (normally) never be exploited, even on 100 watt peaks, which it cannot exceed - think about it.

It just needs to keep up for long enough to meet measured specification. Any longer and the transformer temperature rise will begin to cause damage.

So here I'm using 400 VA (ac watts) for hopefully 100 watts per channel, 200 watts total.

Next up: protection.

If you look at a schematic for a power amplifier you might see a fuse rating which doesn't make sense.

For example the power rail fuse might read 2A but on a 64 watt into 4 Ohm amp, which is 4 amps, you'd expect it to be 4 amps...

However, on test it does 64 watts into a 4 Ohm dummy load indefinitely... with a 2 amp quick blow fuse!

What you need to know is average time current curves for fuses.

Pictured below is an example of what I will need for this amplifier. Maximum current to do 100 watts into 4 Ohms was established in my last post to be 5 amps.

The right hand edge of the dotted line is drawn up from the required current of 5 amps and it intersects the curve for a 2.5 amp fuse at 10000 seconds (2 and 3/4 hours - it may as well be indefinitely), and so if I fit a 2.5 amp quick blow fuse in the power rail (F1) it will do the required 100 watts into 4 Ohms.

I have always assumed that we should not short outputs and have always taken great care when doing anything with speaker connections/cables etc by switching off the amp.

When did it become ok to short out your outputs though lack of care?

Mind you I also have an amp that gets upset at high capacitance speaker cables so it puts me on edge.

Even with home designed/built equipment where you know the consequences of shorting the output accidents can happen, so a degree of protection is reassuring.

Yeah , it was more a comment people not taking responsibility for their actions these days and having to protect them from themselves.

Though it does seem that users are absolved of responsibility these days.

We understand.

So looking at the fuse again there is a glaring error. See where it is? In the positive supply?

This means that T7 has to rely on T6 fault current opening the fuse in which time T7 could be destroyed, etc.

A second fuse cannot go in the ground supply because its resistance increases and falls with load current, so would insert ground current which causes ghastly noises.

The one place it will open for both output devices is the output itself, but what's this about its resistance growing and falling?

Bob Stuart of Meridian realised this and did something about it. He took the NFB after the output fuse so it would reduce any distortion it caused. And Leak did a similar trick to reduce output capacitor distortion several years earlier. And in a way it's a bit like current sensing motional feedback. I did the same in the 1992 MA230 studio amp, and it isn't as straight forward as it first appears.

Should the fuse blow the negative feedback, which being part of the DC circuit, becomes no more, and the amplifier would not be biased.

The only thing which can be done is to divide R8 into two resistors and ground the mid-point via a capacitor, and then the amp will remain biased at DC, but there will be no ac negative feedback at that point.

The fuse can be conveniently placed in series with the output capacitor followed by the negative feedback take off point just before the output inductor.

It then not only helps reduce fuse distortion, but also any capacitor distortion, doing both the Meridian and Leak trick in one go.

The ac "duplicate" of R8 being taken to the output side of the output capacitor will therefore have the wrong voltage - ground instead of half rail, and as it is only now dealing with ac, we can block DC using another capacitor in series with R8.

Time for a re-draw...

I took a day off work to err reread this (honest..) , didn't help , may need another day off, now wish I had not thrown away my Art of Electronics when I started working in computers.

Also perhaps wrongly prejudiced by the title of the thread

Perhaps a view based on listening to old instances of '70s' amplifier designs rather than at the time when they were new but I have always had perception of a less open top end than later newer designs. When I heard my current amp against my then amp a Nad 3030, which I always like for its warm pleasant nature, I was shocked about how much it was rolled off and it was promptly consigned to the sound system at my father's garage.

Is this a trait of 70s amplifiers in their design or component capabilities  or merely the aging components in the examples I get to listen to?

 

Just saying .......

BAK wrote: Would a resettable PTC fuse work for F2 ?

It would work...

...but I don't think it would do the sound any favours.

Too resistive near it's breaking point.

Possibly, but I think you know me better (in-fact I know you do).

And sorry to hear you used up a day off work, but I do feel honoured

Steve and Richard, it was good of you to bring up how these products sounded, and I assume such questions beg an answer...

I think it would have been the introduction of audio-quality op-amps circa 1975 (Signetics division of Philips and the NE5532) which opened up the clue to how amplifiers could be made to sound (dare I say it?) faster.

In fact, a certain Mr Self took something like 48 NE5534's I believe and put them in parallel to make a power amplifier. Quite crazy but I believe it was to prove a point, but my own conclusions might not be as Mr Self imagined.

Steve's copy of the Art of Electronics might have helped and I'm referring to the item on "Slew rate, an in-depth look".

It is OK making the first stage current charge the second stage integrator capacitor quickly and claiming the slew-rate the mathematics suggests, but if the input can only take 20mV rms this is of little use. And power amplifier inputs are more like 500 - 1000 mV.

The Art of Electronics looks at input latitude in relation to slew rate in its study of op-amps, and op-amps are really only a miniaturised power amplifier.

In a transconductance amp the base can only rise so far before saturation and then the input signal is actually bypassing it and driving the next stage - the integrator capacitor - which is very silly, but nevertheless it happens.

The NE5532/5534 have been claimed by onlookers to feature emitter degeneration resistors in the differential inputs, which using the Art of Electronics equation seems plausible, but open loop gain might suggest this is not the case.

It could however be the 100pf capacitor across the collector load which is imitating the existence of such a resistor. You will need to understand intrinsic emitter resistance and how transistor gain can be adjusted by adding to it, to see that the same sort of thing is happening using the capacitor.

Anyway, to cut a long story short, emitter degeneration increases slew rate in real terms (or allows it to happen when the input is large), it is a form of negative feedback but it doesn't reduce distortion and it increases noise, so it is not well liked by specification chasers.

In the two designs commented on above, as far as I can tell, this is not used.

And in my proposed design I omitted it so thanks for raising the question which jogged my memory.

Some will have it that R7 in the latest diagram (previous page) is the additional emitter resistance and I don't need to bother because negative feedback sort of bootstraps this point. But it sort of bootstraps in a differential input without (or with) emitter degeneration resistors, so I don't see the emitter degeneration as being R7.

Yes, what it does by bootstrapping is increase input impedance, and input latitude but that latitude is wholly reliant on the negative feedback.

What it needs is some series resistance in T1s emitter, such that it is a useful number larger than its intrinsic resistance (IMO). At present it is around 38 Ohms, and playing with simulation I found the addition of 75 Ohms did not upset stability with the existing integrator capacitor. By making it bigger the integrator capacitor would have to follow suit, defeating the object.

Using the Art of Electronics' deductions slew rate will be increased by new resistance over old resistance which is 113/38 or nearly 3. That factor being plugged in as the value for "m". We could calculate slew rate for the amplifier using the op-amp formula given, but this isn't a differential input so will probably not hold true.

But if we did, and the amp's bandwidth modelled suggests 3MHz, then by 0.3mFt we would have 2.7V/uS... not good, but if we used the same formula on the amps asked about we would see that much lower, and slow slew-rate doesn't sound good at all.

Calculated the old way, the amp as presented will have a slew rate of 20V/uS. Confusing isn't it?

Next:

How consumer buying habits compromise design...

All the arrogance in hi-fi made those on the sideline go somewhere else, so who's going to be crying soon? Things that go around have a nasty habit of coming around and hitting one where it hurts.

The industry had plenty of sales prospects but instead it trashed its future existence, and Michael Heseltine saw it as he offloaded What HiFi to T3 recently.

For Bob and Ifor: I am taking an in-depth look at each part that makes up a "70s integrated", and in doing so I'm examining the possibility of whether it is viable to manufacture one - or offer one as a kit.

Much has changed over the intervening years which presents many challenges in being able to successfully implement such a design.

But I do believe the eventual outcome will be of interest to some. I expect that some to be a small minority, possibly in the same age group as myself. I also know there are people who have said they would like this, or a "good old fashioned preamp" or a stereo power amp, and in designing the whole, it can be split into its separate functions.

Very much so... the phono stage would be a perfect input to the power amp described on the last 4 pages. If Graham were to include a "line output" as in a "tape record out" and a "tape monitor loop" with monitor switch, the headphone amp would connect to the "line output" and the power amp stage would be muted by the "tape monitor" switch. This would make the system play music through the headphones WITH or WITHOUT the speakers by switching the "tape monitor" switch.

Terrific. Thanks for the explanation. 

Lorlin rotary switches are made in Sussex, England. They are very high quality at a very good price.

Graham has stated elsewhere that he prefers to source the parts he uses from UK manufacturers.

Oops, I used an Elma for my 'pot in a box' preamp when I first got the Proprius amps. I moved up to a Majestic and haven't looked back. Until now as this thread has me curious about 70's amp tech. It is great to revisit the circuits of my youth with someone explaining them properly, many thanks Graham.

The MFB adventure started early so I've been reading the RH544 service manuals. I will be checking the bias and MFB gain soon, last mention in here but a separate topic if anyone is interested? I must say they sounded pretty good after years in storage which shows the charm of well designed transistor amps but will get proper attention soon from me. Apologies Graham for side-tracking this great thread...

Continuing on the theme of input switching and input attenuation, the following circuit fulfils the requirements and adds tape loop and an aux out prior to the volume control.

I cannot comment on the type of circuit used but I thought these were great at the time.  Were they ever used?  Not really, initially on some cassette/old Reel to Reel Recordings and FM but then no and never on vinyl.

 

I suspect the sound produced by many systems pre mid '70s wasn't very well balanced and tone controls were used to either work around the system inadequacies (radiograms etc).  In other cases maybe user preference (how many times have you been in to Halfords or a hire car stereo with the bass and treble both fully boosted?).

 

Winding the clock on 40+ years and it would be reasonable to expect that proper matching of components should mean that the sound isn't excessively bright, dull, bass heavy or light.  Yes some recordings may well have a leaning towards being bloated, thin, bright, dull etc but my take is this is part of the listening experience and if I had my Luxman plugged in I wouldn't be dashing to adjust the tone controls between recordings.

 

I am sure others will have differing views but would be interesting to know how many users have tone controls and do/don't use them?

 

 

I have a similar experience with my Marantz PM64 MkII. Nevertheless, quite a good sounding amplifier and still going strong in a secondary system.

It now needs to go into its "test bed" along with the power supply which is next on the construction agenda.

(spotted one error so far... should say VCC not VDD... )

Not having the ground shorting switches which used to reduce crosstalk from adjacent inputs available to us today we need a different answer.

The line stage preamp circuit previously shown...

...can be redrawn with R11 on the input side of the switch contacts. In other words, each input has a series R11 between its input potential divider and the switch, as can be seen below (R11 to R16)...

The input to T1 base is actually a "virtual earth" which means it is low impedance to the selected input (or as low impedance as it can get using a simple transistor stage - op-amps being much better in this respect).

All deselected inputs are high impedance due to R11 to R16 per input and so any crosstalk between a deselected input and the selected input should be severely attenuated by the low impedance virtual earth junction.

C1 serves to block DC.

So here we start with the test-bed stage and the test-bed case isn't far from the expected size of the finished article.

The next thing we need before the power amplifier can be tested is the power supply, and we have everything but the board to connect the reservoir capacitors up to, so that has to be artworked and made next.

Not only do the power amplifiers need the power supply reservoir capacitors mounting on a board, the preamp section will also need somewhere for its regulated supply to live.

The two 1N4002 diodes protect the BD139 from reversed polarity during power down/switch off, and the 100mA fuse protects against an accidental dead-short on the regulator's output.

And the reason for the zener chain? A single higher voltage zener would dissipate more power and generate more heat (and therefore produce more noise as well). I also have hundreds of 8V2's in stock gathering dust! 

If so will you have twin mains feeds to avoid one interfering with the other?

Richard,

In the example shown, the pre amp power supply feed is from the power amp power supply.

Ian

That would be the case if it was split into preamp and power amp separates but in an integrated you try to cover all bases the best you can.

Take for example full power into 4 Ohms both channels driven continuously. This results in 5.3V of ripple (100Hz) on the collector of the BD139. In fact the 68 volts will drop slightly too. It would have to dip by as much as 10V or a little more including the 5.3 volt ripple to defeat the preamp voltage regulator. A 2 Ohm load per channel continuously might see a little ripple getting through, but with a real world musical source you'd be into serious clipping before then, and you'd simply not drive it that hard. However, at full output continuously into 2 Ohms the fuses are likely to blow.

In normal use, even hitting maximum power on transients, and even with awkward speakers, this isn't going to happen. Even into 4 Ohm speakers with impedance dips.

Propper routing of power supply component wiring is also important, and the take off points from the reservoir capacitors circled here is the correct method...

Thanks again for the educational content. I'd never questioned the term push-pull, you are right that it must be pull-pull.

The prototype power supply is now built and being made ready for testing. Once that's successful the prototype power amplifier board can be tested, and if OK, the design can move into its next phase.

Note that the tag posts are arranged so that each ground (and supply) is hard wired to its individual star point (star earthing).

The hope is to offer a broadly similar sound with one box to what I offer in multiple boxes at present, as far as analogue audio goes.

The customer base for high fidelity equipment is shifting older in age - generally speaking the younger generations have zero interest in high resolution music presentation - and with the inevitable downsizing process most of the older generations tend to be intent on, my thinking is most will want simplification.

It makes sense therefore to try and squeeze as much of the wanted analogue functions into one enclosure.

In my opinion DACs and headphone amplifiers are best as separates because they're a poor fit with the integrated amplifier architecture.

And in saying that, a DAC with headphone amplifier, although that's nothing new, would, with the right sort of inputs and outputs, probably become an acceptable "other box".

It seems I'm an exact fit!

18 months ago I decided to ditch the longstanding two-box amplification and eschew the multiple digital inputs which in my case became redundant as I upgraded the vinyl front-end instead.

Bought a second hand British-built integrated of dual mono design, a quality product with no frills. 33 watts per channel into 8 ohms which is adequate with efficient speakers. 

Sometimes less is more.