The AM Forum

  I ran across a link written by Lynn Olsen that describes many facets of vacuum tube audio amplifiers with design input, and details of why one amp may sound different than another when both may have similar frequency response and similar THD figures.

   Many of us hams might chime in with the "audio-fools" comment, but after careful read Lynn seems to make some pretty good points with data to back them up. One example he gives is what happens to a standard triode amplifier when we remove the cathode bypass capacitor.

  He also makes the case where a solid state Crown DC-300 will look almost perfect on the scope, but will be almost un-listenable with music as the source when the amp is driving quality speakers. Those same speakers and music will sound much better with an old Dynakit ST-70 stereo tube amplifier. The article makes a good case for why this is. This is not a case of SS versus tubes, but more a case against high gain and huge amounts of global negative feedback.

This is a long read, but I think a worthwhile read.

Regards,
Jim
WD5JKO
http://www.nutshellhifi.com/library/FindingCG.html

I must admit, I didn't read the article, but your summary sounds a lot like the same things Bob Carver said back in the 80s.

Of course, he did a double blind comparison of the audio phools with 5 or 6 amplifiers.  After modifying the neg feedback and output Z, the audio phools couldn't figure out which was the t00bed amplifiers and which where the SS.

Most thought all where t00bed, when in fact, only one was.

He was poo pooed as an audio phool, as well, but who knows.

Shane

I'd think a scope offers very limited dynamic range to analyze a signal. You need to use a log scale, not linear, when looking for harmonic distortion, etc.

Also,  I'd imagine the reflected load that the amplifier presents to the speakers must also play a very big part--it provides the dynamic braking for the speaker back EMF to work against. Wouldn't removing the cathode bypass on a triode greatly affect that one parameter? Without getting into a lot of others?

Pete

I would have thought that you  needed an audio spectrum analyzer, so as to see the even and odd harmonic content of the audio output signal, particularly the higher order harmonics.

I agree with Lynn Olson's assessment. I recall reading this information in a now-defunct publication called "Sound Practices"; it was a great audio magazine that catered to the DIY/homebrew vacuum tube audio enthusiast, of which I am one.

It is the higher-order odd harmonics that sound particularly dissonant to human hearing, such as the 3rd, and to an even greater extent the 9th, the 11th harmonic, etc. An 11th harmonic of the fundamental, even when it is only .01% of the fundamental, can be very irritating to listen to and the human ear is very sensitive to this kind of distortion.  And consider too, the intermodulation products created by these higher order harmonics and you can imagine what this does for the sonic purity of an amplifier.

Unfortunately, transistors tend to generate much more of the higher order odd harmonics than vacuum tubes, and this is most probably the factor responsible for the so-called "transistor sound". Vacuum tubes, on the other hand,  when operating linearly, generate mostly even-order harmonics, and these harmonics are much more consonant to human hearing,  embellishing the sound with a "warmth" that many listeners find appealing. The even harmonics can be suppressed to a large degree with push-pull operation.

Interestingly too, excessive negative feedback, normally used to correct for the non-linearities inherent in any amplifier, actually increases the magnitude of the higher-order odd harmonics, making such circuits even more painful to listen to. Feedback really only reduces the magnitude of the lower order odd and even harmonics. Some of the better designed all-triode audio amplifiers utilize zero to only 8 dB of global negative feedback to improve the linearity of the design. Highly regarded audio amplifiers such as the older McIntosh and Dynaco tube units employed as much as 20 to 23 dB of voltage feedback.

Just my 2 cents worth. Not trying to start any kind of a raging debate on this topic, which has been going on for the last 45 years.

73,

Bruce

Oh here we go.
I agree; a lot's been hashed over.

Regarding 'hi end' speakers sounding awful on certain amps. ...-anyone remember Damping factor? DF as regards transistor vs. tube amps?  The acoustic loaded speaker debates vs. the older bass reflex and tuned port designs.
 
My dad and I (guess who did all the wood screwing) even built a labarynth, two column woofer. This was waaaay before Bose put little plastic replicas in their radios.  It was a plywood box about 7 ft. long by 2 ft. wide by 8" deep box with two intertwined sound columns placed upright in a closet behind the living room with the port into the room.

-I used to love to read of "The Audio Amateur," particularly when they had numerous construction articles.  I began to loose interest when they split into "TAA" and "Speaker Builder."  TAA used to have numerous articles about converting Dynaco SP series preamps, amps, etc.

One of the best amps I ever built was Reg Williamson's twin 40 amplifier.  Parts and boards came from Old Colony Sound Lab.  Very clean audio to once younger ears. This amp had a novel power supply using a capactance value amplifier to make a smaller electrolytic cap act as a much larger one.

Audiophools used to be respected; they built their own stuff, some so different even we could hear the difference.  ;D

Shane said:

"I must admit, I didn't read the article, but your summary sounds a lot like the same things Bob Carver said back in the 80s."

Reply by Jim, WD5JKO:

   Please link to the Bob Carver stuff if it is online. I once worked in a R&D lab and we obtained a Carver amp, circa 1980. The specs were unbelievable (too good), yet the thing was small (abt a 8" cube) and was very light. The marketing blurb also touted it as having a "digital computer" to protect the amp from a mis matched load.

  The marketing dept. trumped the engineering dept. What they did was take a 250 watts / channel amp, and remove the heatsinks, and remove the power transformer. To keep the amp from over heating they added the "digital computer" which was a thermal Klixon switch. On the bench we noticed that this thing was as safe as a 5 tube AC/DC table radio, and capable of 10 watts / channel (sine wave). Any more power than that and it would thermal off and on like a Christmas tree light.

   After this experience, anything said by Bob Carver was not listened to be me.

  The article I linked to seems to try to describe much more than tubes versus transistors. A special emphasis is made towards gain stage linearity with moderate gain, instead of high gain controlled with feedback amounts approaching 40 db. Somewhere from the mid fifties to the mid 80's we relied on way too much feedback to overcome gain stage non-linearities.

73,
Jim
WD5JKO

I don't have any of it, hence the reason I didn't link to it in the beginning.

This was text referenced from the brain, circa mid 80s :)

HOWEVER, if my memory serves me well enough (below 40 yrs old, but lotsa parties in the teens -n- twenties), it was either a Sharper Image catalog, or one of the Popular series of magazines.  My stepfather had an ongoing subscription to them, the Heathkit catalogs and a bunch of robotic stuff.

The radio I picked up from Gramps, may he be resting in peace every time I stick my hands in this stuff :)

Shane

Shane,

    Yes the 1980's are a bit of a blur to me as well!  There is some stuff about Bob Carver on Wikapedia. I am quoting one part of his amplifier here:

"A power transformer connected to an amplifier, said transformer being adapted to operate at maximum power output at a relatively high frequency, in the order of 20 KHz. There is a control circuit which produces control pulses at a frequency of 20 KHz. The voltage level at the output of the transformer is compared to the amplitude of the audio signal to be amplified to produce a control signal related to the difference between the two. This control signal acts through a modulator to pass portions of each control pulse, with the duration of the pulse portions that are passed being generally proportional to the magnitude of the control signal. These pulse portions in turn open and close a switch connected to the primary winding of the transformer to control the duration of current pulses which flow through the transformer at a 20 KHz frequency. When the power requirements of the amplifier are either higher or lower, the duration of the current pulses to the transformer are made longer or shorter, respectively, to match the power requirements of the amplifier."

  What this was in reality was a low voltage 60 hz power transformer to power the amp for 5-10 watts output. Beyond that there was some form of switching network to bypass the transformer to support the load and while doing so to lose the isolation from the power line. This was revolutionary and I wonder if the UL rules as written then would have caught this issue and deemed it a problem.

   The company I was working for was really into switching regulator design at that time. It was about 1980 when IR came out with the first of their power FET's, the IRF-100. These power FET's, and some others from Siliconix started a revolution. Today we have class D and E AM transmitters from their descendants.

The Carver approach although clever sure seemed to be a method prone to trouble and controversy. We looked elsewhere for innovative designs. The Carver amp ended up on the shelf and I'm sure eventually discarded.

Jim

I used to get Speaker Builder and The Audio Critic.

Audio Critic is now an on-line 'zine:

http://www.theaudiocritic.com/cwo/Web_Zine/

According to Peter Aczel the editor, the two critical parts today are the recording acquisition and the speakers.  The amps today should all basically be similar and transparent.

Take the time to read though his stuff, including back issues.

Tom said:
"According to Peter Aczel the editor, the two critical parts today are the recording acquisition and the speakers.  The amps today should all basically be similar and transparent."

Reply by Jim:

  I suppose if the amps are all similar in design, and vintage, and without tone controls, or equalizer, then that might be true. Otherwise changing from one amp to another can make a huge difference in the sound. Again this is not tube versus solid state. Here is one example where an amp with fabulous specs (THD and freq response) sounds awful with complex waveforms (music):
http://www.stereophile.com/solidpoweramps/1292crown/index1.html

For most of my lifetime engineers have concentrated on frequency response and THD. Those variables are only part of the picture while Intermodulation distortion (IM), Transient Intermodulation distortion (TIM), and Power supply rejection ratio (PSRR) often take a back seat to THD. Addressing these issues works against profit margins, and the bean counters come into the situation too and fire the engineers so they can start production.

The article I linked in my first post needs to be read in full. I had to read it 3 times!

73,
Jim
WD5JKO

The Carver designs worked fine. The basic idea was to float a small (10 watt or so) amp on top of a rather crude amp that used voltage switching rails. The feedback around the little amp was supposed to clean up the nasty rail hops... it works in as much as you can make a very lightweight amp that puts out a ton of apparent power, but really doesn't sound all that good. Lot's of PAs used various Carver incarnations. The original "cube" did work, but had some problems, I don't recall exactly what - but the basic idea continued into other models.

As far as the harmonics of an amp, anyone who doubts all this should search online for Dr. Earl Geddes, and the "Geddes Metric". He is a highly qualified Phd, who is the antithesis of an "audiophool". Read what he has done.

My opinion of Peter Aczel is that he is the equivalent of "Consumer Reports"...
While he is correct that the ultimate determining factors (other than one's ears) is the original recording and the speakers, he is dead wrong that all amps are essentially transparent and the same. See what Earl Geddes says, if you doubt.

There is a neat site somewhere online where a fellow did some nice tests comparing a NAD to a SE Triode amp. Interesting bit that one. The SE Triode actually had LESS distortion than the lower spec'd NAD as the power level went up. Things are not all that they appear to be on the surface.

I suspect that Lynn Olsen and most others are using Audio Precision gear and/or high performance PC soundcard based FFT analysis, not a simple scope. Some folks may be using HP or similar spectrum analyzers also...

I don't know of any respectable high-end amp mfr today that is NOT concerned with TIM, THD, IM and PSRR, and more... fwiw.

If you want to know what people are thinking about in terms of solid state amp design, I strongly urge you to follow almost any thread in http://diyaudio.com in the Solid State section to see just how much minutia is being considered!! (Lynn is a frequent contributor in the Tube section there, btw.)

                     _-_-bear

Jim,

We've had threads going on receiver rf dynamic range measurements and found that the best today is really no better than the best receivers of 50 years ago.

I guess what you are saying is that today's amplifiers are not any better than 50 years ago also.  Is this right?

Bear,

What's wrong with Consumer Reports type of reporting?  What's wrong with identifying a $100 tuner that is as good as a $1000 tuner, a $200 amplifier that is as good as a $3000 amplifier?

Can this be separated out - does a tube amp sound better than a solid-state amp really, or is it just that one grew up listening to the tube amps?

Jim,

When an amplifier output stage has no feedback there will be great interaction between the output stage and the speakers.  That means that a given speaker cabinet/system will have a different frequency response and sound with a different feedback-less amplifier. I don't consider this a good situation at all.

The speaker's distortion is so much greater than the amplfier's.  When you believe you hear IMD products (which is most probably the difference in the "good" and "bad" sounds) this is most likely from the speakers.

Bear, you agree that the critical points in the audio system are the recording session and the speakers.  Even with the most mediocre listening system you can  point out a poor recording session.  We can't do anything about the recording.  Speakers is what the listener need so concentrate on.  This is what Peter is saying.  Yet people are obsessed with the amps!  Go figure.

P.S.  I can't hear the 18th harmonic of 1 kHz, and neither can anyone else.  But all of us can hear the 2nd and 3rd harmonics of 1 kHz very well.

Tom said:
Now there's an interesting concept!

If the snake oil salemen selling the so called esoteric $1000 amps are exposed, they won't make money. That's what is wrong with it.

Until solid system engineering principles are applied to a sound system, all this talk about amps is missing the forest for the trees.

Considering growing up with tube amps..

One area where tube amps were considered superior is in the overdriven state. ...(which a lot of youngsters did ... I know I did.) The harmonics are usually more musical rather than what you get from solid state due to tube operating characteristics.

Agree with Steve that 'solid engineering principles' need to be applied, further there needs to be convergence between perceived sound testing and engineered sound testing. 

All systems should be double blind tested to a set of standards.  The standards can be 'gold' or 'silver,' just as long as they're consistant.  After all my 'gold' may be your 'lead.'  Double blind speaker tests have to be in the same acoustic environment, right down to spacing from walls, room dimensions and form factor (deviation from rhombic, approaching spherical, etc.) , acoustic absorbant material placing, faux furniture, listener postion.. all 3 dimensions for both source and listener, etc.  This is sometimes hard to do given that some testers place one set of speakers outside of another, etc.  Perhaps a revolving lazy susan of speaker sets behind an acoustically damped wall might be in order.  ;D

I listen to what sounds best to me. Usually it is a tube amp. The speakers also have alot to do with the way an amp sounds. I've heard it both ways, tube amp sounds better on these speakers, ss sounds better on those speakers. But that is subjective anyway. I don't buy my hi-fi gear in the stereo store but if I did it would no be by specs but by listening.

When I owned a repair shop in the 80's, a guy brought in a Carver cube. Upon examining it, the circuit was too unusual to fix easily without a schematic and Carver would not sell me one. I told the customer it was special circuitry and suggested he take it to the authorized repair center for a prompt repair. He moaned about what they'd charged him last time, but what could I do?

Sorry folks, listening to rock music on this equipment is gay.

Tom,

   I agree with you to a point. The audio source and speakers are important. The best amp and speakers cannot overcome a poor recording. That said, an amp can certainly "colorize" the sound in many ways. One example I already linked to was the Crown reference amplifier that the reviewer gave a thumbs down to. The amp was tops driving woffers to high SPL, but poor with complex music where it suffered from IM distortion, and TI distortion problems. That amp like many is not transparent. With IM we create distortion products right where our ears can hear them. Here is an example:


From: http://www.pmillett.com/Books/bbc_feedback.pdf

"The two terms, known as the sum and difference components,
cause the unpleasant sound from an overloaded amplifier. The process
by which they are introduced is known as intermodulation, the two input
signals acting as the carrier and modulating signal, to produce sum and
difference terms which are equivalent to upper and lower sidebands res-
pectively in amplitude modulation.
As an example, suppose two input signals are applied to an amplifier,
the frequencies being 256 cfs and 320 c/s. These frequencies are those of
middle C and E and blend harmoniously to give a chord known as a major
third. Due to non-linearity in the amplifier, the output contains components
with frequencies of 64 cfs (the difference term) and 576 c/s (the addition
term]. The difference frequency is that of C two octaves below the input
signal and harmonises with 256 c/s and 320 cfs, but the addition frequency
is that of D one octave above the input signal and produces a discord with
both 256 cfs and 320 cfs."


I have another example that is more relevant to us hams. I once built a push pull plate modulator with a pair of 8417's. I had a good 550v plate supply and a regulated 300v screen supply. I set the bias for 50ma total, and proceeded to modulate a 4D32 with it. On the scope I could modulate the rig very well, and the waveforms looked quite good with no evidence of clipping. On the air the reports were terrible saying my voice was muffled. The reports got much better when I increased the idle current to a little more than 100ma total. My conclusion was that the IM distortion of the high gain Beam Power tubes and pentodes (EL34, 6550, 8417) is very high unless you run these tubes very hard at or near the plate disippation limit.

  Negative feedback reduces THD (2nd 3rd, 5th) more than the higher harmonics, and does little to reduce IM. For low IM we need to go back to making the amp as linear as possible prior to adding the negative feedback. When we do this we don't need 20 - 40 db of NFB. Maybe we can get away with 6 - 10 db NFB instead. With proper design we can still have an acceptable source versus load impedance, and damping factor.

Another factor with audio amplifiers is that when they clip, they momentarily go open loop, and with a high degree of negative feedback, that gain reduction goes down depending on the extent of clipping. The human ear is very sensitive to this, and since a solid state amp clips more abruptly, it is much more important to keep a SS amp away from the clip point.

The speaker impedance is all over the map too. Between the cross over networks and speaker and box resonance points, the amplifier has a formidable job to do. The amp may see anything from a few ohms to 80 ohms over the audio range and some reactance too. Many amp designers ignore this and instead use a 4 or 8 ohm resistive load while tweaking away for max power at min THD. This is why amps are not transparent, and are every bit as important as the audio source and the speakers chosen.

I respectfully disagree with your point that amps don't matter.

Regards,
Jim 
WD5JKO

Now there's a statement of the obvious. Overload any amp and it will distort. This issue is not how much IM is created when overloaded but how much is created when not. More BS from the snake oil salesmen.

Steve,

What in the world are you talkin' about??

You need to read up some on the more recent thinking, engineering and testing on these things. Find and read what Dr. Earl Geddes says.

The way an amp clips or even a modulator is indeed important, but that's not the main discussion, since the main discussion revolves around unclipped amplifiers.

However, if you consider that many CDs are recorded with peaks that are 20dB up from the average level, and then you go to play back said CD on your home system at say 90dB SPL at the listening position how much headroom do you need?

Assume to simplify the matter that you manage to get 90dB SPL at 1 watt because you listen very close to ur speakers (yeah, right  ;)  ) then we need to reproduce unclipped 20dB up from there. That means we need 100 watts. But if it is a real world speaker and it takes say 4 watts to get to that level, then you need 400 watts, and if it is a typical high-end speaker with only mid 80s reference sensitivity, or you like to play at a slightly higher average level, like 93dB SPL average, you then need to double all those powers - you gotz a 200 watt amp in ur system? You gotz a 1000watt amp? Whatcha got?

I'll bet dollars to donuts that most amps in most systems are clipped far more often than people would like to believe - like all the time.

What's the spectra of distortion of ur amp, no matter what the power rating, as the power gets within 6dB of clipping, how about 3dB of clipping? Betcha that you start to see much larger differences between amps when you look at them this way, and you can hear the same differences just as well...

If one doesn't care, doesn't listen that critically, doesn't notice any differences, that's fine, but there are measureably, objectively, scientifically, and audibly present differences.

                           _-_-bear

                 

Nothing at all, IF you could do such a thing.

One can't.

If you simplify, or reduce the criteria sufficiently then it is all equal.
The analogy I've used over and over is that of a full bore Formula Car vs. your "daily driver". If you never run the Formula Car over 55mph, I suppose that they're pretty equivalent. Imho, that is what Aczel and Consumer Reports both do, in their own way.

It is very rare that a cheap and inexpensive bit of audio gear offers equal or better performance than more expensive gear. There are some exceptions but very few.

A good example might be Behringer - good stuff, very cheap, right?
But not as good as most better grade or top-of-the-line gear that does the same job.

Those who are more discerning (and those who pretend to be as well) want the best performance available, not the best bang for the buck. (although it is good to know about both)

                   _-_-bear

You can't simply draw this conclusion.

It is not true in every instance.

In some cases, a variation in frequency response (usually in the LF only) is beneficial, since it results in an increase in low bass, and an improvement in the overall Q of the bass curve...

BUT, you're thinking only of tubes - in the case of a solid state amp, you can make an output stage with extremely low output Z compared to the load, no global feedback, and you won't get this same effect. Of course you can make an output stage with even lower Z using global feedback techniques.

If you happen to biamp or triamp then using a ZFB tube amp on say the tweeter is a very different thing than trying to run a multi-way speaker from a small ZFB tube amp.

Proper application is key to any situation, and improper application usually results in undesireable results. The idea behind the typical feedback amp is to make it universally applicable. Sometimes special purpose designed gear (amplifiers in this instance) are better or best.

                   _-_-bear

I agree that speakers can and do create IMD products.

It is insufficient to say that you have IMD products without defining their spectra.
This is important!
Non-trivial point.

I had a set of tweeters here that ran from 1500Hz up past 20kHz. They were printed ribbons ("leaf ribbons") that could do 128dB SPL/1M at full power, with LESS than 0.1%THD. Consider that I typically ran them at 28dB down from there... perhaps these tweeters had distortion levels on par with many amps??

Anyhow, it turns out that it's not so much which harmonics, but what the ratio of harmonics are. It turns out it's not so much HOW MUCH (amplitude) but what the ratio of harmonics are!! You can hear some combinations of harmonics (typically 9th and down, especially 5,6 &7th) in certain ratios very easily. Whereas the same harmonics in a different ratio but much higher in amplitude are almost indetectable by the ear.

This is why absolute levels of IM or THD are very poor indicators or predictors of audibility of artifacts.

As I said go search on the "Geddes Metric" - his work follows some British research from the 70s and arrives at the same conclusions.

It explains the whole issue very neatly, imho.

                     _-_-bear

<2 cents>

That's just the sum and difference. How about the 'real' IMD

3rd order:  192 & 384 Hz

5th order:  128 & 448 Hz

7th order:   64 & 512 Hz

Unless you're running a serious class A amp these are going to be an issue.

Always been curious if anyone has run double blind tests with different levels of 3rd, 5th, 7th etc order IMD to see what level is noticeable. Bet 3rd order products at 1% (-40 dB) stand out like a sore thumb to someone with good ears...

</2 cents>

Bear I think you have settled everything.  No need for anyone to do any more evaluations.  Just go for the highest MSRP.

Anyone who runs their amp into clipping doesn't know what they are doing. No reading required to know that. Obfuscation and double talk doesn't get it done. Sorry.

Of course we're talking about unclipped amps. But the so called expert tried to bring an overloaded amp into the equation. This is typical snake oil salesmen talk. It's garbage and should be pointed out as such.

Tom & Steve,

I really don't get it.

MSRP has nothing to do with anything I said.
Sorry you don't like to see what prices are for high-end gear, I don't set the prices. We can talk prices if you like, in terms of running a business. But that has nothing to do with sound. If you want lower cost high performance gear you can build ur own.

If neither of you two gentlemen have bothered to take a few minutes and search out what Dr. Geddes' research says, then there's not much more I can say.

Most amps that are in use in the home environment run into clipping all the time.
That's a simple fact. The way an amp sound when clipped plays a role in practical home use. EVEN IF you drop the issue of clipping, you still have to consider the distortion of an amp when driven to somewhere near clipping, like -3dB of clipping. Look at any amp's distortion curve (with a few minor exceptions) and the distortion goes up by maybe two orders of magnitude when dealing with HF?

So, let's not call any of this "garbage" or "snake oil salesman" talk.

Come back with facts please, not name calling.
It's innapropriate and uncalled for.

Thank you.

All I ever wanted to know is how does Bose do it in such a Small Box...the Big Room zound... ;D

My definition - The listening system ideal audio amplifiers – a perfectly linear amplifier with zero output impedance and flat response from 20 Hz to 20 kHz.  Who wants to make an argument against this?

Any and all audio amplifier designs that that come very close to this ideal will all sound the same.  Peter Aczel is saying that “today’s amps” do.  I’m not sure if he is excluding all tube amps in ”today’s amps” or not.  And just because someone puts out an amplifier doesn’t means it is not a poor implementation and can those units can be excluded also; all amplifiers existing today cannot be put into the “today’s amps” grouping.

Any amplifier that does not sound like the rest of the pack is deficient.  It’s that simple.

Jim said “He also makes the case where a solid state Crown DC-300 will look almost perfect on the scope, but will be almost un-listenable with music as the source when the amp is driving quality speakers. Those same speakers and music will sound much better with an old Dynakit ST-70 stereo tube amplifier. The article makes a good case for why this is. This is not a case of SS versus tubes, but more a case against high gain and huge amounts of global negative feedback.” 

From your BBC reference Jim, page 35 - “To obtain low distortion and a very low effective value of output impedance, both desirable for high quality reproduction, feedback must be applied over at least two stages of amplification.”

So this is exactly a case for global negative feedback.

Jim, I found nothing in the BBC pdf in support of deriding the use of negative feedback.
- - -  -
I said “When an amplifier output stage has no feedback there will be great interaction between the output stage and the speakers.  That means that a given speaker cabinet/system will have a different frequency response and sound with a different feedback-less amplifier. I don't consider this a good situation at all.”

Bear said  “You can't simply draw this conclusion.”
Yes I can.  It is quite simple and fundamental.  Passive circuits at work here Bear.  I’ve measured it too.  With no feedback at the output to lower the output impedance to around 100 milliOhms, it is probably in the neighborhood of 4 Ohms.  The speaker at the low frequency end of the spectrum will go though a series resonance.  Only with a very low output impedance will the frequency response be guaranteed to have proper flatness and roll-off.

”It is not true in every instance.”
There is a chance that one’s hi-Z output amp and speaker happen to work together o.k  Otherwise, it is possible to select a speaker and amplifier combination that does not exhibit this.  But you have to have a number of drivers on hand and have a noise generator and acoustic spectrum analyzer or sine generator and SPL meter on hand and find a “good” combination.  How many home owners can do this?  So they probably have a screwed up bass response otherwise, if they own a hi-Z output amp.  Wouldn’t it be better to be able to put an amp and speaker together and know that they work together and produce a flat response.  This is called good system engineering.  Most people own low-Z output solid-state amps today so they don’t have this problem that the no-FB output amp owners do.

”In some cases, a variation in frequency response (usually in the LF only) is beneficial, since it results in an increase in low bass, and an improvement in the overall Q of the bass curve...”
Why people would want to deliberately color the frequency response is beyond me.  Raising the Q of bass curve is bad to me, I can’t think of any good argument for doing this, Bear.

”BUT, you're thinking only of tubes - in the case of a solid state amp, you can make an output stage with extremely low output Z compared to the load, no global feedback, and you won't get this same effect. Of course you can make an output stage with even lower Z using global feedback techniques.”
No I’m not thinking only of tubes amps.  Negative feedback lowering output impedance applies to tube and transistor amps both, can be done to both.  You can make a transformer-coupled output amp. lower impedance by reducing the secondary turns; with no feedback.  This is more in the domain of tube amps today; but you will loose a lot of output power capability though.

Bear, do you have any of the following test equipment:
a.  THD meter
b. noise generator
c. acoustic measurement meter – audio spectrum analyzer or SPL meter
d. pc-based audio analyzer software for a. and c. above.

- - - - - -
Shane said, referring to Carver “Of course, he did a double blind comparison of the audio phools with 5 or 6 amplifiers.  After modifying the neg feedback and output Z, the audio phools couldn't figure out which was the t00bed amplifiers and which where the SS.

Most thought all where t00bed, when in fact, only one was.

He was poo pooed as an audio phool, as well, but who knows.”

Right on Shane.  Since people said they liked the sound of tube amps, Bob did the study to define the difference in tube amp transfer function to make a solid state amp sound like a tube amp (marketing and sales for profit).  I think it was a big hit at the time.  Of course if the tube amp sounds different than the solid state amp, that means there is a difference. And they both can’t be the ideal linear amplifier, or they would sound the same; they both can’t be right (a common HUZ-ism here)!

- - - - -
It’s interesting to see that the audio community does not resort to the 2-tone IMD measurements when analyzing audio amplifiers (which are or should be called because of the application demand – "ultra-linear” amplifiers).  THD measurements are easier to make, and useful, but 2-tone IMD measurements are what is needed also for meaningful and critical evaluation of amplifier non-linearity.  Why this technique isn’t more common in the audio world is baffling.  Audio isn’t magic, some metrics are missing.
- - - -
Jim,

 Your first post with the study on various amplifier configurations and the harmonics is a good and meticulous study.  The researcher was looking for an explanation for system listening differences based on different amplifiers that appear the same.  Interesting to note is that the variations at the 10th and 11th harmonics, is 10 to 15 db.  With this being 10 to 11 kHz and in the region of 100 dB to 120 dB down, I am skeptical that this is the explanation for differences heard.  This study probably was done with a resistive load and recent analyzers.  I doubt that home listening systems can realize more than 80 dB dynamic range, keeping peaks below clipping.  The high order harmonic products would therefore be below the noise floor, regardless of the amplifier.  This is why I still think that listening differences are based on speakers rather than the amplifiers.  There’s plenty of possibility there for the drivers and cross-over components to generate IMD and harmonics from what is being fed to them.

Now if the researcher can re-do the tests based on IMD testing and show significant amplifier IMD performance differences that correlate with the listening system results then he is on to something about specific amplifiers being the problem.  And the problem would be demonstrated amplifier nonlinearity.
- - - - - - -
Bear, I haven't read Dr. Geddes paper yet, but I will.  I don't think you posted the link, otherwise I would have read it by now. 

My home amp in the last 5 years doesn't get driven into clipping, historical fact.  I don't listen to hard rock either.  If I drove my amp into clipping in routine use Bear I would say that I was either deaf beforehand or would be deaf by now.  Perhaps you and your friends should get hearing tests.

Tom,

   Thanks for putting all this together. I'm packing for a trip to Boston, so I'll review the details later. The first thing you say struck me as not universally true and I stopped reading there:

"My definition - The listening system ideal audio amplifiers – a perfectly linear amplifier with zero output impedance and flat response from 20 Hz to 20 kHz.  Who wants to make an argument against this?"

  In order to get 20-20K at low THD out of many "low end amps" you need lots of global NFB to cover up all the shortcuts taken. If it is SS, maybe as much as 40 db NFB. With this much NFB the bandwith (open loop) needs to be much wider to maintain stability. There are consequences of doing this that compromise other spec's not stated like IMD and TIMD. Also the damping factor has a range that it needs to stay above, but not necessarily too high either. A zero source impedance amp is not always what you want. Some speakers might perform better if Rs  is ahigher. You do want some (not a lot) interaction between the woofer /enclosure resonance and the amp. With your "perfect" amp there will not be any. The low end response will be sterile.

Look at sheet 10 downward and zero in on DF:
http://www.turneraudio.com.au/monobloc845se55.html

The amp described has a Rs of 0.5 ohms with only 8 db of Global NFB.

More later,
Jim
WD5JKO

 

Insufficient bandwidth.
You'll have problems if you don't have response above 20khz.


Peter Aczel is wrong in real practice.
Read what Dr. Earl Geddes says and get off this bandwagon.


Incorrect again.
Many amps that measure technically very well have issues that make them sound like other amps with the same issues, per Geddes' research (he merely has reduced to scientific measurement what others have long been saying). Similarly many amps that measure technically very well sound different than the first group I mentioned.

What will confound you, not having read Geddes' JAES peer reviewed papers, or his books, is that some amps that do not measure particularly well also sound the same as those in one of the two above groups of technically good measuring amps!

You're working with old information that has been rendered obsolete and incomplete. The old idea that any "good" amp that measures less than some threshold of THD (like 0.001%?) is indistinguishable from another. This old idea is false.



It is possible to get low output Z without global negative feedback.
It is also possible to get low distortion without global negative feedback.
Feedback has both benefits and deficits and its overall benefit depends on the specific implementation - it is neither good nor bad, it is a tool.


I have no idea what output stages you are familiar with.
The "average" SS amplifier that has say 2 or 4 output devices per channel will have a high output Z without feedback, agreed. But this is not the case where there are multiple output devices in parallel. Say something like 24 or 48? There are quite a few amps like this...

Otoh, the Crown MacroReference amp uses both multiple output pairs and a double loop of feedback which yields a crazy DF of something like >60,000!!

There are some speakers that happen to be "overdamped" - meaning their Qt = <0.5 for example. In this case a high DF (low output Z) amp will appear to have anemic bass, whereas an amp with a low  DF will have actually greater bass extension and perhaps even a "flatter" response.


Let's keep one thing in mind, very very very very few speakers have a "flat" response at all, and even fewer have a "flat response" in the room, and fewer still have a "flat" response in the room and off axis. So, what exactly do you mean by what you've said WRT "flat response" or "proper roll-off"? Please don't tell me that the speakers that are like this are not properly designed...


Actually today anyone who wants to have excellent test gear and owns a computer can do this and more. But there are no combinations of amps and speakers that anyone can put together and know a prori that the results are going to be just fine. Sure, amps with identical or nearly identical DF will produce a frequency response curve that is nominally identical, but so will cars with similar power to weight ratios produce similar 0-60 times. It doesn't really tell us much.

Similarly, anyone who has a ZFB power amp and doesn't know how or when to apply it, is no different than someone who buys anything else that they don't know how to apply it. Audio is no different, you get the proper item for the application. The guy with the Bose Wave Radio is on the opposite side of the audio world from a person who builds his own or buys a ZFB tube amp... should ZFB amps be illegal??

....

Quite the contray, you're making a false assumption.
The assumption is that the only way to make an accurate or even good sounding system is one way - with a high FB low output Z solid state amp. Again there are speakers that demand low output Z. Bass horns are one possible system that this works for. Another is the so-called "EBS" alignment. A fellow named Bob Cordell champions this idea. Do you recognize the name? Check his website. I'm not saying that it is good in every case. Raising the Qt of a speaker system depends entirely on what it IS to begin with!


I'm saying the opposite. You can get a low output Z without adding feedback.
I'm not talking about tube amps here either, I'm talking about SS amps.

Then there is Hawksford's feedforward correction...


Does it matter?
I don't have a particle accelerator, but I can read and know the physics?
But yes, I have measurement capability. And what I don't have I can get my hands on, need be.
What are you getting at?


Say what??
Huh??

Look up CCIR...
Report back.

How many watts is your amp?
How far from your speakers do you sit?
What is the typical listening level in SPL at ur listening position?
What are the legit specs for your speakers (sensitivity, impedance, etc.)
Let's figure out if you ever clip ur amp or not... ;D

           _-_-bear

Bear,

Are these the Dr. Geddes papers you are referring to?  See the 2 pdf's below.

I will make amplitude measurements on my listening system today if I can and report back.

I never made any limitations on bandwidth.  I stated a minimum (perhaps implied) frequency response only.

Bear said -  "You're working with old information that has been rendered obsolete and incomplete. The old idea that any "good" amp that measures less than some threshold of THD (like 0.001%?) is indistinguishable from another. This old idea is false."

I never made such a claim.  My stance is that the "low" THD numbers do not tell the whole story.  One needs to advance to IMD measurements.  The human hearing is much more sensitive to IMD than harmonic distortion and the amps produce noticeable IMD at "low" THD levels.  * But the situation is that most people's hearing is poor and all over the lot.  They can't tell the difference in the "good" amps.

* What I really mean to say here is that most people do not or cannot judge sound quality.

But you must first drive your amp into distortion, then tell everyone it sounds better than other amps.  ::)

CCIR is just one standard used to do IM tests, iirc it is 19kHz & 20 kHz tones... fyi.
No one who designs a "serious" amplifier uses mere THD anymore - that's for the marketing people for mid-fi companies, imho.

I think those are probably the Geddes papers, if they are from his site, and they include the term "Geddes Metric"... I didn't look yet.

I agree that most people can not tell the difference.
That doesn't mean that the difference isn't there, or isn't real, or isn't audible.

Thus the Bose Wave Radio is said to be all that you need... ha ha.

However, I believe that if an average person, like you or me, got to hear two amps on a system that is sufficiently clean (whatever that threshold is...) you could hear such a difference quite clearly, without ambiguity.

Personally, I think that it is cool that Geddes has brought to light what many people have been saying for a long time. It reveals something very interesting about sound and the human ear. It also shows why some tube amps can sound much better than some solid state amps of seemingly lower distortion!!

         ;D ;D

                            _-_-bear

Ummm... no.

Obviously, once driven into clipping, the distortion levels increase momentarily. This is usually more audible than whatever spectra of distortion the amp has the rest of the time. Unless the spectra of distortion happens to conform to the Geddes Metric, and is essentially inaudible the rest of the time... in which case the amp only begins to sound noticeably distressed when clipped on peaks...

Then it depends how much other masking effects you are getting from your speakers (a big source of distortions) - but again much depends on the spectra of distortion not the absolute level of distortion!, and the rest of the signal chain.

For most average folks with mid level systems, especially those that are not set up optimally (like speakers kind of randomly placed for example) these things don't make a whole lot of difference.

It's once you begin on the path to optimization and getting better or higher performance gear that you suddenly are confronted by the stark realization that things are not as simple as they seem when you started.

It's akin to watching an old 1960s color tv program broadcast using 1960s technology and then moving up to (or beyond) today's HD, or maybe a high res computer monitor and suddenly you can see things that formerly were glossed over and were unimportant. You can see them and they're there and you notice them, even if you would prefer to not.

Good example, there is tape on the floor on The Price Is Right, you couldn't see it at all in the old days...  ;D

Or maybe it is the difference between SSB and AM?
Do you notice the difference in sound quality?
Sure that's a gross difference, but in sound reproduction there are differences of several plateau levels, it's not just a binary thing...

                     _-_-bear

Umm, yes. Let me take you back in the thread.



I added the bold on overloaded. You defended the above statement with great vigor. Now you say no. Which is it?

Sorry, Steve, I'm totally confused...

The books on Pete Millet's site are useful, but generally old books that have no copyright. In some cases the information there has been eclipsed.

There are various issues at play when an amp is clipped (overloaded?), especially when a NFB loop is involved. I don't know what ur pointing at. Clipped amps produce distortion. Unclipped amps produce distortion.

The two questions are:
- how much
- what spectra
- oh and a third, is it audible?

This is for the unclipped and clipped states... is there something else I am missing about it?

                 _-_-bear

Bear,

I just looked at my stereo system which is a slighty modified Harman Kardon 630 receiver and modified Utah HS4-B 3-way bass-reflex speakers. 
As I recall, the amps are rated at 30 Watts per channel.
I sit 13 feet from the speakers.

The audio meter’s SPL function is set to:
SPL Weighting = FLAT
SPL Response = FAST

The amp, no load puts out 60 Volt peak-to-peak.

I ran the receiver in mono, listening to a local FM station, only one speaker channel on.  This eliminates any question of channel imbalance.
Adjust for 85 – 90 dB SPL reading.  I wouldn't care to listen any louder than this.  I believe you mentioned the same SPL.
I saw maximum of 10 Volts peak-to-peak.
This is 16 dB headroom.  I assume running stereo for the same SPL would produce a little more headroom.

Next, I ran the receiver from the CD player.
Adjust for 85 – 90 dB SPL reading (perhaps a couple dB louder than FM actually was).
I saw a maximum of 20 Volts peak-to-peak.
This is 10 dB headroom. 

- - - - - -

There's always been tape on the studio floors.  It was visible.  It may be more visible today due to the solid state technology.

Looking at the Audio Engineering Society website (aes.org) a bit I spotted these 2 items:

AES Journal Forum

1. A New Method for Measuring Distortion Using a Multitone Stimulus and Noncoherence

Read comments here:
http://www.aes.org/journal/online/comment/?ID=14381


2. A paper that must be purchased but just the description is interesting.  Another listening study by Geddes, this one 3 years more recent than the Geddes Metric paper.

AES E-Library:
Audibility of Linear Distortion with Variations in Sound Pressure Level and Group Delay by Geddes, Earl R.; Lee, Lidia W.

Recent psychoacoustic studies of nonlinear distortion have yielded some new insights into what audible problems in loudspeaker might be related to. This paper will show the results of recent subjective tests which extend the work of various previous works to show that sound level significantly affects the perception of linear distortion in audio systems. This means that the hearing system itself is nonlinear and what has been thought of as being nonlinear distortion in the audio system may actually be a nonlinear perception directly in the receiver itself.
Paper Number:   6888    AES Convention:   121 (October 2006)
Authors:   Geddes, Earl R.; Lee, Lidia W.
Affiliations:   Eastern Michigan University; GedLee LLC    (See document for exact affiliation information.)

AES Store: Purchase Full Paper
$5.00 AES Member      $20.00 Non-Member

The "receiver" being the human listener.

A paper on perceived distortion from Geddes' website by Keith Howard below:

20 volts peak to peak? take 1/2 and then take .707 of that, square it and divide by eight. That gives us 6 watts into 8 ohms.  Take the 60v p-p take half and then .707, square and divide by 8 and you get 56 watts. So, must be a 60 watt per channel amp.

Yeah you have an apparent ~10db (not 16dB) of headroom, assuming there are no peaks over the average level, which is a poor assumption using a CD that is not compressed up the yinyang, which apparently is not the sort of music ur listening to?

You don't want the flat setting on the SPL meter, you want the one that rolls off the lows... but we can still work with these readings for now... also it is best to use the slow setting to average the level, we want the average level...

You might want to use pseudo random noise as the level setting source, FM interstation is almost right, but it suffers (as you probably know) from the de-emphasis boost built into the receiver... I think you can probably generate a .wav file using shareware that will do that trick, if you wanted to. Heck, if ur not in a hurry I can email you a wav file or mp3 that will work...

You can see the problem though, if you pushed that CD playback to just 3 dB more (two just noticeable increases) you'd have dropped the headroom down to only 6dB.
That's a guarantee that peaks will be into the amps clipping...

Those old Utahs are likely up in the high end of the 90-100dB sensitivity range, above 95 almost certainly, maybe as high as 100dB/1w/1m...btw.

Which, interestingly enough makes them a candidate for tube amplification - which if it a modern tube amp, using modern power supplies and modern resistors and caps might be very very surprising to you! I suggest that you try that, not some old tube amp, some time?

The tape on the floors is more visible due to higher resolution, regardless of the technology that produces higher resolution. It was always there, but it was not very noticeable at all - that is the point of my analogy.

                 _-_-bear

Thanks for the link(s) Tom!

I recall reading the last link when it came out in Multimedia Mfr mag - an Ed Dell publication...

I hope that what I've been saying, and really what Geddes and Shorter before him are/were makes sense now?

                   _-_-bear

Hi All,

   I am up near Boston on business. Boy the WX sure stinks!

It seems that the division among us posting on this thread has narrowed a bit. I started this, and to be honest, I have learned quite a bit considering all the various point of views. Understanding good sound is like trying to understand what motivates my XYL. Science doesn't always get you there. ;-)

 One point worth making is that IM can be high WITHOUT OVERLOAD. All we need is a non linear transfer function on a gain stage. This is kind of represented with my old 8417 modulator when I had the idle current too low. I had plenty of power, and it looked OK on the scope. The audio reports were terrible and it sounded like a carbon Mic that needed to be whacked. After raising the idle current it sounded pretty decent. This was NOT OVERLOAD.

   On a slightly different twist, some years ago Joe Curico published a paper, "The Audibility of Power supplies". see link:


http://www.curcioaudio.com/Audibility_of_PwrSupply.htm

 
    I read this a few times over several years and said to myself, yea yea yea, BS, snake oil, since his business is selling kits.
Well I've had a Sherwood S8000 tube amp / FM tuner for 30 years (30w / channel amp). About 5 years ago it showed serious fatigue as some tubes were aging, caps were drying out, and some of the wiring had insulation falling off. The output tubes are 7868's (Novar version of the Octal 7591), and are the same Sylvania's since 1981 which are going strong to this day.

   This amp has been the center of my main sound system. For several months in 2004 I took the amp out of service and did some cleaning up and updating. I used some of Joe Curcio's ideas using my own designs. I beefed up the plate voltage (450V) with 200 mfd, and electronicially regulated the 7868 screen voltage, audio driver, FM tuner, preamp filaments, and 7868 grid 1 bias settings.

   The important point here is that the basic Sherwood tube amplifier circuitry remained stock except for the ability to set the bias on each 7868 output tube. My changes concentrated on power distribution. A summary of my changes are at this link:

http://pages.prodigy.net/jcandela/Sherwood_S8000/HV_REG/S8000_Mods2.htm

   So did any of those changes make a difference? Well at conservative volume levels, I'd say no. If I crank, especially with deep bass, I can say absolutely yes. This amp is now playing HD DVD's, TV, CD's, Play station 2, XBox 360, and Wi. When the kids play games the house shakes. The other night we played "King Kong" on HD DVD, and the noises from the big Ape (mostly under 50 hz) were resonating things around the house. That Sherwood low end was never like this when it was stock and much newer.

    Back to the point. A low end amp is likely to skimp on the power supply whereas a high end amp might have better magnetics (pwr transformer) and bigger electrolytics. Also the low end amp will have less heatsinking (SS amps) and hope for providing high peak power when the average is very low. I don't think the "Snake Oil" salesmen are as prevalent as some think selling $1000 amps that are no better than $100 amps. Instead I think they have moved to selling $100 HDMI cables, and other high end cables for HDTV installations.

Good info on this thread from all,
73
Jim
WD5JKO

Jim,

No doubt power supplies or lack of regulated supplies in the amplifier is a source of many problems and complaints, transient phenomenon.  For many years when I hear of the line cords, penny in fusebox etc. improvements I said to myself well maybe there is power supply problems.

My receiver/amp here is a Harmon Kardon 630 which is 38 years old now.  Many years ago after noticing the significant hum level (saw-toothed also) at the speaker terminals, I investigated and took out the simple series pass transistor/zener diode regulator for the +22 Volt supply and replaced it with a LM723 circuit I think it was.  Now pure dc.  Also the main speaker amp supplies had bad ground loops involving the filter capacitor currents.  I put all of the raw filter stuff on isolated buss wire which came to a single point ground on the chassis.  I moved the speaker ground return points.   Afterwards the hum was in the noise level and sinusoidal.

How many high end amps use a linear regulated supply ahead of the output amps?  Does anyone at all?

Anatoly,

Did you ever build and test your high-powered source follower amp design?

Time to update ur amplification... many reasons.

But anyhow, many amps and preamps use regulation in the power supply...

There are issues with "regulation" it's not a simple cure-all. You can look at regulation as a type of amplification, in as much as it uses feedback to control the "linearity" of the output (nominally DC). What it does steady state is not the same as it does in dynamic conditions, just like the amplification stage it is feeding.

To quote Bill Witlock (Jensen Transformers, and AES Fellow):
"A variation of your method was presented in a paper by the late Deane Jensen and Gary Sokolich, "Spectral Contamination Measurement", at the AES 85th Convention, 1988 (preprint #2725). In their (and my) opinion, much of what audiophiles describe as "the veil" consists of low-level, non-harmonic distortion products. I believe a major source of these is due to widespread use of op-amps under heavy negative feedback. As frequency increases (at ultrasonic frequencies, for example), the op-amps essentially run out of open-loop gain and start approaching open-loop conditions. This allows their inherent non-linearity to intermodulate distortion products from previous stages and the result is non-harmonic distortion products. It is one very good reason for passive bandlimiting at the input of every active amplifier stage. Otherwise, the effect is cumulative - and demonstrably ugly sounding. The test Jensen and Sokolich proposed is very similar to the AP (Cabot) multi-tone test.

Bill Whitlock
President & Chief Engineer
Jensen Transformers, Inc.
(and recent AES Fellow)"

Of course that's his approach to the "cure" and CD is bandwidth limited already...

             _-_-bear

  I have another story where bad power worked against me. I had a RCA AR-88 receiver and after a lot of work I got it going. At one point I tried to beef up the audio output that I think was a single ended 6K6 good for 2-3 watts. I did the usual coupling cap change and added some NFB. The efforts were not responding well, and any attempt to reproduce the lower frequencies became futile. I was suspecting the audio output transformer, but before I got too far I hooked up my scope and snooped around. Much to my surprise I had over 100v P-P of B+ ripple to the 6K6 transformer whenever I cranked the audio while reproducing a male broadcast voice. The B+ was about 250v without audio, and it was all over with audio. The frequency was around 80 Hz, and somewhat sinusoidal. It turns out the the HV power supply used a 5Y3 and a dual pi C-L-C-L-C filter. The cap values were very small and the L values were very high. I'm pretty sure I had a filter resonance condition working against me. When I fixed the power supply (another story) the 6K6 behaved exactly how I had intended.

Don't overlook the power supply folks...

Jim
Wd5JKO 

"Time to update ur amplification... many reasons."

Update to tubes and transformers Bear?

"I believe a major source of these is due to widespread use of op-amps under heavy negative feedback. As frequency increases (at ultrasonic frequencies, for example), the op-amps essentially run out of open-loop gain and start approaching open-loop conditions. This allows their inherent non-linearity to intermodulate distortion products from previous stages and the result is non-harmonic distortion products. It is one very good reason for passive bandlimiting at the input of every active amplifier stage. Otherwise, the effect is cumulative - and demonstrably ugly sounding."

Bear, what breakpoint frequency would you choose "for passive bandlimiting" to filter out the offending audible harmonics yet allow the overall response to go to 20 kHz.?

Jim, yep, never forget that supply LC filtering are resonant L-C circuits.  O.k. with steady loads but can swing greatly with varying loads.  This is why the SSB linear amp B+ supplies are just capacitors; very stiff.

Tom,

Please don't assume?

I said a few things that seem to have gotten lost in there...

Whitlock's cure is his, not mine. He sells transformers?

There have been substantial improvements in solid state amplifier topology, with attendent improvements in steady state and transient performance since your old HK was designed and built. A newer design potentially would both measure better and sound better - potentially.

Tubes or solid state depends on things that are unclear from here, and more or less up to you. I'd bet that your old Utah speakers would sound a lot "nicer" on a modern tube amp of high quality design. They might even perform better in terms of measurement too...

The ironic thing is that ur an AMer? You run tubes?
It would be easy, no simple to build a really excellent P-P tube amp for you using modern designs and parts... I suspect you'd wonder why you ever used that HK when you were done, truly!

FYI, I'd suggest a pair of triodes run in AB2, using a DC coupled cathode driver - hey wait a sec, that sounds a lot like a modulator!!   ::)  The thing is that you don't want to push for max power at all, so you can keep the B+ modest... all you need is excellent output iron... high quality parts (resistors, caps etc...) and if you insist separate supplies for each stage (not a bad idea at all).

For a killer result consider some 845s, 211s in class A... etc... these are not your grandfather's tubes, using modern parts and power supplies!

If ur interested in modern solid state, take a look at the kit offerings from http://Borbelyaudio.com they are first rate when done in a first rate implementation. Just an example of excellent modern solid state design...

If you want to see what a very leading edge design for SS looks like, just one person's project, go here: http://www.synaesthesia.ca/PGP.html  follow it through, it is a fairly amazing design... dunno how it sounds, but the circuitry is outstanding. There are some others that are equally interesting out there...

                _-_-bear

Bear,

More on my Utah HS4-B 3-way bass-reflex cabinets with 12" woofer.  I bought them in the early 1970's.  I did a major overhaul on them in 1986 for many reasons.  The original mid-range drivers were horns.  They were replaced with Peerless K040MRF closed-back midrange cone drivers.  The spec on them is 91.5 dB SPL. 1 W., 1 M.  The loss through the cross-over is negligible, in fact a slight step-up in voltage at 1 kHz., just measured.  So I did not do my own SPL measurement at 1 W, just going with the mfr. spec.

The original “cross-over network” was a joke.  Got rid of rotting front foam cover (no kids here to poke holes in woofer).  Incorporated a new much more complex cross-over design and tweaked it.  Changed driver positions in cabinets (one cabinet became mirror image of other).   Reinforced cabinet corner seams and joints from inside with wood strips, nails and glue, and nailed in finishing nails from outside of cabinet.  This eliminated a “bark” I would get from the cabinet at loud peaks sometimes (loose sidewall).  Yes, one transient distortion source identified and fixed!

Tom,

Sounds like ur poised to make a breakthrough here!  ;D

You might want to try downloading "sound workshop" - it's a freeware FFT type program that runs on the PC... use it an ur ratshack spl meter for a microphone and you can get a really good idea of what ur freq reponse looks like. I'd measure the system at <3ft on axis to the tweeter, and on axis to each driver, then measure it up close, like 4-5" in front of the cabinet on axis to each driver to see what the individual driver responses look like... that will tell you one heck of a lot more than you probably want to know about what they are doing or can do.

You regular everyday onboard soundcard will work fine for this.

No matter what if you have any electrolytics in the xover, at least bypass them with the largest value film caps you can find, and if at all possible don't use any electrolytics at all, use all film, and better still use polypropylene only. At least use polypropylene for the tweeter.

I think that you will be surprised what that does, changing nothing else.

                       _-_-bear

Bear, there are many many much higher priority projects here. 

When I did the overhaul work I used a stand-alone 20 band/half-octave filter hand-held spectrum analyzer with pink noise generator.  Each driver was individually connected to the amplifier output and tested for frequency response.  I used 3 snapshots at on-axis and 2 off-axis readings and averaged them together for the frequency response.  This is how the original midrange horns were kicked out of the design.

The two new cross-over frequencies were then chosen to be in the middle of the drivers’ overlap regions.  The cross-overs are a mixture of film and NP electrolytics.  They stand for now.

After all the work, shortly after I moved into this house in 1990 one original 1” soft dome tweeter died.  I just stuck another different tweeter on hand in it’s place and haven’t bothered to mess with that either.

Ok, if you want to have some fun, download that Speaker Workshop program and have a go at it with almost any mic you have on hand - if the mic from that spectral analyser can be tapped, that would work. About a half hour or so dorking with the program would open up a whole lot of insight that the bar graph type display simply glossed over. The fact that the mic you pick might be non-flat won't change the value of taking a look at it. Of course the Panansonic $1 electret mic element is flat enough until out to ~15khz so as to be virtually perfect for this application (looking at the real response through the xover areas...)

But let's leave it here, since we're trying to bridge two points of reference that are very far apart, making it hard to accomplish.

I can try to use an analogy to explain, and hope that it doesn't insult you or anyone, I'm simply trying to make it clear and easy to understand. In effect we're comparing a 3x5 snapshot (of the same subject, for example, corresponding to the same recording) to a professional large format photograph blown up to poster size and printed on Cibachrome, with color controls for hue and saturation... Of course if you never ever saw the big pro print, you might not ever realize what was possible, or how much attention to detail is required to produce it!

           _-_-bear