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Today is the day I begin my tube amp adventure. I am going to pick up either a VAC 770 or CJ 12 to audition. I spoke to the dealer yesterday on the phone, and asked if tubes amps were generally slower than solid state amps, and was told that just the opposite was true. The dealer told me that with ss the resistors have to turn on and off, while the tubes stay on all the time. He said this is the reason tubes amps are faster. I am pretty clueless when it comes to the technical stuff, but isn't his explanation a little too over simplified? There have to be a lot of other factors that determine the speed of an amp, and I find it difficult to believe his statement that all tube amps are faster than ss amps. Or am I wrong?Suzy
There are several things that determine the apparent speed of a power amp, and the most common answer would be slew rate. However, this is not necessarily the case, as a raw slew rate number does not tell us how well the amp will handle the signal just below the slew rate limit, which is where most of the real world signals live.A better indication might be risetime, which is more of a small signal measure of speed, and more appropriate since much of music lives at the 1 or 2 watt level. A good indication of an effortless sounding amp is one which has a small signal risetime and power bandwidth that are essentially the same (as long as this bandwidth was at an adequately high frequency). This would mean that the HF response of the amp would not change with drive level, and that it was fully input filter limited rahter than slew rate limited. All to often I can hear the output stage of a power amp lugging down as the level goes up, even though it is capable of delivering lots of continuous current.
Another factor is how fast the amp recovers from overload. This may not seem appparent at first, but unless you have several hundred watts and/or a very high sensitivity loudspeaker system, or play at very low levels, most people are clipping their amps to play at realistic levels.
Quick analysis:
typical 88 dB sensitivity hi-fi speaker system, listened to at about 10 feet. In order to hit 100 dB, this requires approx. 120 W. Try for 110 dB, and the requirements go up to 1,200 watts. Even 103 dB will need 240 watts. You see, the power requirements escalate rather steeply. The only way a 100W amp (or less) will not be in clipping some part of the time is to play at unrealistically low playback levels.So what you might ask? Well, how long the amp takes to recover from clipping varies quite a bit from amp to amp, with SS being much worse, on average, than tubes. Typically, the more power, the longer the recovery from clipping, so the intermediate power levels tend to be worse for most people, as they are still clipping some, and doing it worse. Most tube amps clip and are back tracking the signal almost right away as they come back out of clipping. Many SS amps 'stick' to the rails, and take a long time to return to tracking the signal.
This will influence how quick the amp is perceived to be as well.
The overall balance and HF and LF roll-off points, as Trevor has pointed out, will also influence the apparent speed. One thing not mentioned very often is that many SS amps use a DC servo to control DC offset at the output. These are often designed improperly, causing a infrasonic boost. Rather than a controlled and well behaved roll-off at LF, these amps actually overshoot on LF transients, causing the sound to become somewhat dark and wooly. Amps that do not do this would tend to sound quicker in their apparent speed.
Jon Risch
Hi John,
Thanx for the explanation. Makes sense, though I'm still not sure how this translates into the "sound of speed"... I usually listen to stuff at pretty low levels so I haven't got a feel for what flat-topped sine waves sound like... guess I'll have to throw my little amp on my maggies and crank it. OBTW, I still have that little roll of speaker-coil wire you sent me last year, but I still haven't gotten around to experimenting with making ICs out of it, what with moving, laziness and all. Have you tried this stuff for IC's yet? http://www.cablewave.com/products/CablewProd/cables/foam_dielectric.htm
Looks like the small solid core stuff would make a good DIY JPS-labs alternative... seen a 75 ohm version but not sure where to get it.
In general, a typical tube power amp, is slower than a SS one. The reason for this, is the (usual) presence of an output transformer. The output transformer tends to restrict frequency response, at the low and high ends of the spectrum.Both tube amps and SS ones (assuming no interstage, or output transformer are used) may have very wide frequency response figures, indeed. IOW: tube and SS PREAMPS are essentially unrestricted in their theroretical frequency response figures. It is a trivial issue to build a SS, or tube preamp, possessing a frequency response, well past 1MHz.
What REALLY makes an amp sound 'fast', is the restricted bass performance of certain products. Many amplfifiers have been produced,over the years, which have a measureable restricted LF response. Naim, is a good example of this type of product. Such products are often characterised, by listeners, as possessing a 'fast' bass performance. Psycho acoustics, tend to make the listener feel that the amp is 'fast', even though it may have a frequency response, which is no better than another, similar amp, at the HF end.
Be VERY wary of the dealer you have spoken to.
Trevor
Allowing more bass always seems to widen the SS , give the illusion of "granduer" and as you say , reduce the "speed" of the system.
I also beleive that the way the system responds to the transient in the whole Freq spectrum is important. to deliniate transients more , one can whack up high mids and lower treble (1.5-4k range)
more importantly , I also believe that decay chatracteristics of a system are very important in respect to its appernet speed , truncating low level detial or not doing it well enhances the speed , good suystems let the note tail off more thus diluting the "pace" somewhat
I find that high end systems normally enhance the former (leading edge) and most negate it agian by allowing a much higher level of "note decay" , thus in general dont sound as fast as lesser systems , I think the latter (decay) is more important in perception of the transient than the former leading edge.
As you say , considering a transient is normally a complex full freq signal , the way a system handles the whole musical freq spectrum is important.
Generally I find speed a trade off , increasing relative pace often requires a decrease in the lower bass and the "air" and wide SS of the system.
Trevor--I'm just wondering if you are looking at the spec sheet for Naim amps or if you have actually heard, say, a 250 driving DBLs or ProAc R5s or something similar. I am confident if you had you would not say that Naim amps sound fast because they are (purposely) bandwidth limited. While they are, they do not sound that way, offering grip, control and authority to the lowest frequency limits of speakers. They also play in tune, which I have never heard a monster amp do.
Regards,
Bob
I have heard many, many Naim amps, over the years. Since 1978, in fact. I recall, quite vividly, the appeal of that brand, way back then. In fact, being a technical guy, I determined to find out exactly why Naim amps, sound the way they did/do. It is precisely a combination of their bandwidth limited design and high transient current ability, which predominantly determine the way they sound.My comments were not a criticism of Naim amps, merely a statement of fact. I have, for instance heard Naim's sound better than almost any other product, under certain situations.
Trevor
Trevor--That makes some sense, even if Julian (RIP) says it is only part of the reason. What I don't understand is why Naim amps, which are bandlimited, have better bass than so many other amplifiers. Why don't other amps do the same ? To cite one example, a Naim 250 had convincingly better bass on a pair of ProAc R2.5s than a Krell FPB300 or a Levinson 332.
Regards,
Bob
Trevor,You didn't expect such a remark on Naim without a naimiac responding, did you :-)
So here :
What you say generally is true. Exaggerated (sp ?) bass and lower mid indeed makes a "slow" impression. But, the important thing musically is not only speed but also the timing, and this was nicely described by the second reply below. It has to do with the coherency of the notes as they are played.
If you hear a drum hit, and you hear the high frquencies apart (time-wise) from the mid, then this gives a "diffuse" drum-hit and the music loses its impact, although it may have gained in details (hearing them separately), imaging etc.
And for evidence, not all "thin" sounding systems time well, listen to an Audiolab and understand what I mean.
Specifically for Naim, these guys are sometimes OTT with their concern for timing and "impact", willing to sacrifice other things. But the better your Naims, the less you sacrifice.
Ahh, and I can always blame the setup :-)
He probably meant that transistors turn on and off rather than resistors (if you think of electricity flowing through a wire like water or gas through a hose, then you can think of resistors as being like those little hose clamps they use in chemistry labs to control the flow of a gas or liquid... most are fixed but some, like the volume control, are variable... rough analogy). Technically wrong anyway since transistors can switch orders of magnitude faster than tubes, and in class A ss amps, the transistors are on all the time like a tube anyway, and in class AB amps they don't switch on and off either... transistor is already on to the linear part of it's conduction curve before it drives the speaker (there are pulse amplifiers and integrating amplifiers that do switch but not typically used for audio). Don't know what is meant by amp speed in an audio sense... perhaps tube amps having more even harmonics while SS amps tend to also have stronger 3rd, 5th and other odd-order harmonics. Or maybe an amp not acting like a variable delay line that does not distort the sound envelope (high frequency components of a violin note reaching the output at the same time as the low frequency components instead of one part being speeded up or delayed and altering the envelope of a note, but I'd call that smear not speed). Someone want to expain just what "speed" means in an audio sense... is it just slew rate or something else?
. . . your question reminds me of an often quoted remark by Justice Stewart about the definition of obscenity in the 1964 case, Jacobellis v. Ohio, which has woven its way into contemporary vernacular:"I shall not today attempt further to define the kinds of material I understand to be embraced within that shorthand description; and perhaps I could never succeed in intelligibly doing so. But I know it when I see it, and the motion picture involved in this case is not that."
Perhaps the perception of the speed of an amplifier, in contrast to the electrical performance on small signals, is one of those things that you know when you see [hear] it. p
Pam wrote:
"Technically wrong anyway since transistors can switch orders of magnitude faster than tubes..."**This is a common misconception. An AVERAGE tube used in a typical tube amplifier, is at least as fast, as an AVERAGE transistor used in a typical SS amplifier. Don't forget, that before transistors, we were using tubes to opeate with FM radios, TV sets, etc (ca. 100MHz). Radio amateurs were (and still are) using tubes to operate at the GHz region. It is not the active devices, which limit the upper frequency limit of amplifiers, it is the coupling devices and circuit topology. In fact, it was not until the mid 1970's that high power transistors became available, which would allow SS amps to be built with a decent HF response, at low distortion.
What is different, is the COST of building amplifiers to deliver such performance.
As for your comments re. speed, please see my comments, elsewhere.
Trevor
Hi Trevor,
Um, well your GHz region maser, klyslotron and magnetron tubes aren't exactly operated the same way as a triode or pentode, and yes, I know tubes can get up there (got an old LCZ bridge that uses tubes and measures up to 170MHz... was tinkering with radio before audio), but the major frequency limiting parameters are geometry (stray capacitance and inductance acting to limit f.... one good reason for chip caps and resistors) and physical size (takes time for electrons to mozey from the emitter to the plate, determined by the gap and voltage differential)... I can find little IC amps that operate to 10GHz for a few bucks, but would be real hard pressed to find a tube amp that would operate that high. Kind of a moot point in the audio world when both tubes and x-sistors can provide useful amplification in the MHz region, but my reply was meant in a practical sense for commonly available parts. Tubes still rule for high voltage high current switching... EE&G makes a tritium doped tube that operates as a triggered spark gap and was used in A-bomb triggers, but it is equivalent to an SCR not a transistor. In any case, I do agree with you in the sense that what the dealer told Suzy was wrong by virtue of being totally irrelevant.
The ability of an amplifier to quickly respond to an oncoming rapid transient waveform (percussion, including piano, for example) is indicated by rapid rise time, high slew rate, and/or wide bandwidth. In general it is easier to optimize solid state gear for high slew rate than it is tube gear because of the bandwidth limitations of the output transformer.This is where the output transformerless tube amp shines. Examples are the amazing little Berning, the Transcendent, Atma-Sphere amps, Fouriers, and probably one or two others I've overlooked. I am familiar with Atma-Sphere amps, and they have an ease about them that may well arise from their extremely high slew rate (600 volts per microsecond). Atma-Spheres are among the fastest amps ever made and, to the best of my knowledge, are both the fastest tube amps and the fastest amps that employ no negative feedback.
Back in the early 80's high slew rate was all the rage, but then along came "perfect sound forever", and suddenly everything above about 10 kHz was hash anyway. Suddenly slew rate made no sonic difference, except to the vinylist. Now with SACD and upsampling DAC's available and becoming more affordable, slew rate is going to matter again.
Duke LeJeune
AudioKinesis
Duke wrote:"Back in the early 80's high slew rate was all the rage, but then along came "perfect sound forever", and suddenly everything above about 10 kHz was hash anyway. Suddenly slew rate made no sonic difference, except to the vinylist. Now with SACD and upsampling DAC's available and becoming more affordable, slew rate is going to matter again."
Uh, why?
At 48kHz (the high frequency limit of 96kHz sampling), if you were to swing it +/- 60 volts (or the equivalent of about 225 watts RMS into an 8 ohm load) its maximum rate of change would be a little over 12 volts per microsecond. Of course this scenario would never exist in any realworld home audio system. Under realworld conditions, anything anywhere near 48kHz would be so low in level I doubt you'd have anything with a rate of change greater than a volt per microsecond at typical power levels. So most any amplifier with a slew rate of a few volts per microsecond wouldn't be in any danger of slew rate limiting even with the increased bandwidth of the new formats.
se
just picked up copy of the march 2000 hi fi world, page 55 by noel
keywood has some spectral analysis of cd and dvd-a.
the john basile quartet on dvd-a at 24/96 (no other info on which
segment, length, etc) shows peak at -5db(300hz?) , 20khz at -40db and
-46db at 30khz. there is stuff at -70db at 40khz. do not know the
(octave,third, sixth?) of the analyser but this shows there is
energy up into the supersonic. might be also why there are tweeters
made for the japanese market up to 120khz.leelock
leelock wrote:"just picked up copy of the march 2000 hi fi world, page 55 by noel
keywood has some spectral analysis of cd and dvd-a.the john basile quartet on dvd-a at 24/96 (no other info on which
segment, length, etc) shows peak at -5db(300hz?) , 20khz at -40db and
-46db at 30khz. there is stuff at -70db at 40khz. do not know the
(octave,third, sixth?) of the analyser but this shows there is
energy up into the supersonic. might be also why there are tweeters
made for the japanese market up to 120khz."Ok. Let's take 40kHz and assume a full-scale swing and set full scale at +/- 2 volts peak. That's a rate of change of 0.5V/us. Run that into an amplifier with a gain of 20 (26dB), and we get 10V/us. Add John's recommended 5x guardband, and we get 50V/us. Piece of cake. :)
se
just saw in another mag that the chief engineer of tannoy says that
cymbals go up to 100khz. when we get the 192khz stuff in place does
that mean the 50 v/us goes away and we trade up?i seem to remember some old articles that says marantz japan has
tried sampling rates up to 500khz and in each case sound quality
improvement was audible. is this where we are headed after the 192khz?of course by that time i will need hearing aids to get it--hmmmmm-- do
they make some lightweight single ended tubed triodes with bone
conduction------leelock
The microphones will have to improve if we want to get 100K response, except
for special, very noisy, instrumentation mikes. The standard is about 40KHz for the best practical mikes today, and has been for the last 30 years.
it is true the popular and mainly colored(musical?) mics in wide use
do not go up in response as the capsule needs to be small and the
electronics very quiet(expensive?). there is also no choice in polar
patterns--only omni for the really high end specs.however earthworks(founded by ex-dbx guys) makes instrument and
music mics with outstanding characteristics. their instrument mic
is omni pattern and get out to 55khz at -3db and with self noise of
26dba with max spl of 150db. the low end is something like 5hz at
-3db.i guess this is a chicken and egg thing where there was no demand for
the response possible with the likes of the earthworks (i think the
small diameter b&k mics also went up to very high khz) since the
domestic playback systems could not take it. but just like the super
tweeters now coming out in japan with 100khz response i think the
industry will respond with 50khz and higher mics once sacd and dvd-a
starts hitting the market in volume. imagine a digital mic with smarts
to average the noise out and microfabrication in silicon....
leelock
Thanks for the input. I have not followed this manufacturer, but I have helped design mikes over the decades and know the tradeoffs. This is the problem: It requires a small area diaphragm to have a high resonant frequency and to have a flat response at high frequencies. The electronics has to handle VERY high Z and still be very quiet. This is a difficult tradeoff, added to the fact that small diaphragms have low output as well. B&K made very small instrumentation mikes, but they were very noisy and could only be used in very high ambient environments or measurement purposes.
Cymbals may, indeed, have harmonics out to 100kHz. That is pretty much meaningless, though, since 16/44 digital systems prevent any signal past 22kHz.Trevor
Trevor Wilson wrote:"Cymbals may, indeed, have harmonics out to 100kHz. That is pretty much meaningless, though, since 16/44 digital systems prevent any signal past 22kHz."
Yes. But what got this sub-thread started was commentary on the new digital audio formats and their extended bandwidths as it relates to slew rate.
se
CD's, I don't listen to no stinking CD's! ;-)
.
Your statement makes sense, but once upon a time so did the assertion that a 44.1 kHz sampling rate would accurately replicate signals up to 22 kHz. Why does a higher sampling rate sound better to me, when I can't even hear 22 kHz? Okay maybe this is apples and oranges, but it seems to me sampling rate is in some ways analogous to slew rate.Now that I have an upsampling DAC I hear a greater difference between amps, specifically on piano and cymbals. Over the past two weeks I have experimented with six different amps - solid state high power, solid state low power Class A, single-ended 845 triode, and three OTL tube amps. I hear a greater nuance and sweetness in both the crash and shimmer of cymbals, as well as the impact (yes, impact) of piano, with the high slew rate OTL gear. Is this cause and effect? I don't know, but for now I believe so.
Slew rate matters because of transient modulation distortion (TID).You want your amp to be relatively immune to the effects of RF noise.
RF noise of sufficient level will cause amplification stages to
hit a "brick wall" of sorts if you do not have sufficient slew-rate
(and/or power supply rejection), causing audible distortion.A lot of high quality amplifiers have relatively large bandwidths,
over 200 kHz. This is a way to ensure that you still have a fair amount
of useful gain at 20 kHz. This gain, if taken linearly, will decrease
distortion, phase, and output impedance (improve damping factor).But a high bandwidth also means higher frequency noise. To properly
support this "noise bandwidth" without TID you necessarily need a high
slew-rate.Matt
Steve, I have to correct a wrong assumption on your part as to what is the optimum slew rate for an amplifier. It is important that the slew rate limit is NOT approached, as this represents VERY HIGH distortion. Also, only small amounts of hi frequency need be present to create a VERY HIGH RATE-OF-CHANGE of the audio signal. Think about a 100HZ sq wave with a 1us rise time. The info at 100KHZ will only be 1/1000 the amplitude at 100HZ, but if it is not there, then the risetime will be compromised. This is an extreme case, but it shows that sine waves can be a poor judge of slew rate potential. We have found for audio that a minimum slew rate of .5V/us for every volt of output either + or -, is necessary. This gives 50V/us for a 100W amp. Most serious designs, even Parasound, has more than twice this number. This info can be shown in Eero Leinonen, Matti Otala, and my paper: 'A Method for Measuring Transient Intermodulation Distortion' in the AES Journal in Apr.1977.
john curl wrote:"Steve, I have to correct a wrong assumption on your part as to what is the optimum slew rate for an amplifier. It is important that the slew rate limit is NOT approached, as this represents VERY HIGH distortion."
Howdy, John.
I wasn't attempting to ascribe any optimum slew rates for amplifiers. Just giving it a sense of scale. And a signal with a rate of change less than the slew rate of the amplifier will not slew limit the amplifier or else the slew rate of the amplifier has obviously been mis-specified.
"Also, only small amounts of hi frequency need be present to create a VERY HIGH RATE-OF-CHANGE of the audio signal. Think about a 100HZ sq wave with a 1us rise time. The info at 100KHZ will only be 1/1000 the amplitude at 100HZ, but if it is not there, then the risetime will be compromised."
Well technically if you have a square wave with a fundamental frequency of 100Hz, the magnitude at 100kHz should be zero since that's an even multiple of the fundamental. But I see what you're trying to say. So let's take a closer look.
Let's say you've got a 1 volt square wave with a fundamental of 100Hz and you're carrying that out to the 999th harmonic. The rate of change of a 99.9kHz sine wave (and therefore the maximum rate of change of a 100Hz square wave carried out to the 999th harmonic) is 0.6276902 V/us. If you remove that harmonic (leaving you with the 997th), the rate of change would be 0.6264336 V/us. A difference of 0.0012566 V/us or 0.002% by my reckoning. Not very significant I don't think.
If you were trying to make some other point, please clarify.
"This is an extreme case, but it shows that sine waves can be a poor judge of slew rate potential."
Depends how you use the sine wave. The rate of change of a square wave can be no greater than that of a sine wave equal to the highest frequency component of that square wave.
"We have found for audio that a minimum slew rate of .5V/us for every volt of output either + or -, is necessary. This gives 50V/us for a 100W amp."
That last part doesn't seem to add up. An amplifier delivering 100W RMS into an 8 ohm load will be swinging +/- 40 volts, no? And if you're recommending 0.5V/us for every volt of output, that'd come to 20V/us instead of 50.
se
Steve,I have to agree with John on this one, I followed the whole TIM thing from the beginning, and read the papers and literature. Some tried to dismiss it as 'good design practice' and unnecessary to be specifically addressed, others tried to play the slew rate numbers game and trivialize what was needed.
The slew rate, is by definition, the point at which the amp is literally pushing the signal out as fast at it will go. The feedback loop has been broken, the amp is racing to catch-up. The actual input signal has dissappeared, and all that is being output is an error signal of the amp racing to catch up. This is indeed 100% distortion and then some. Many amps that were specified for the best numbers use a huge amount of overdrive to maximize the slew rate, pushing the amp very hard into non-linear operation. Others may remove a front end filter or some of the protection circuitry to allow the amp to even reach the specified slew rate on the test bench. For most, to assure some semblance of linear operation (not necessarily low distortion), the slew rate of the signal must be kept to a fraction of the amps rated slew rate.
CD is not the only source, neither is 48 kHz DAT tapes. As a long time owner of a MC cartridge front-end TT system, and quite a few direct-to-disk or half speed vinyl recordings, I am regularly faced with significant information out past 20 kHz.
Jon Risch
John Risch wrote:"I have to agree with John on this one, I followed the whole TIM thing from the beginning, and read the papers and literature. Some tried to dismiss it as 'good design practice' and unnecessary to be specifically addressed, others tried to play the slew rate numbers game and trivialize what was needed."
I wasn't attempting to dismiss or trivialize anything. I was simply trying to give some sense of scale to the issue and express my opinion that I don't believe the new digital audio formats necessarily represent a call to arms with regard to slew rate (and it was the new digital audio formats that was the context of the message I originally responded to).
se
Of course, you are correct in this, Steve. It doesn't change my calculations, however. Most amps, today, are fast enough. This was not true perhaps 20 years ago. Both Crown and Marantz then made large amps with slew rates of 5V/us. This could be a problem, but today amps are generally more than 50V/us. It has been my experience that 10-20V/us is a bit too low for best performance, but it is mostly because of other factors related to slew rate, not the slew rate itself that is responsible.
Small point of correction/clarification, here, John. My old Marantz Model 500 (ca. 1974) has a slew rate of 11V/uS. It was a very impressive sounding beast, in it's time (when it didn't blow up). I must rebuild it sometime.....Trevor
Did you measure it, or did you read the spec sheet?
Both. A really long time ago. I acquired the amp and used in my system, whilst I was the Australian service manager, for Marantz. I vividly recall the sonic improvement, over the Japanese product, I had been using, up until that time. On the bench, both amps measured almost identically. The sound was something else. It was the amp, which started me on my journey. It was the amp which taught me to view spec sheets with some degree of scepticism. It was the amp, which showed that all amps do not sound the same.Trevor
I still maintain that the specified slew rate of the Marantz 500 is 5V/us. We used the Marantz 500 in the '70's for a low frequency amp. I can't find a direct record of that spec. and will consider any proof that it is greater, but I stand on my memory. For the record, the slew rate of the Phase Linear 700 is 11V/us.
I've just checked my service manual, for the numbers. It states:"Slewing rate: Faster than 11 Volts per microsecond."
The owner's handbook says the same thing. Yeah, before you ask, I've retained the original packing box, too. Whasat? Did someone mention 'Anal Retentive'?
How long did your 500 last? As service manager, I repaired every one of the things that made it to Australia (only 3 did), at least 5 times, before I got a hold of some decent Japanese output devices, to replace the horrible old Motorola things. Legend has it, that Marantz built 300 Model 500 amps. From initial design, to the end of the warranty, the whole episode was a US$3 million loss, for the company. Legend has it that Jim Bongiorno had something to do with the design (I'm not suggesting that the two facts are related, BTW). Marantz replaced the 500, with the 510/M. Possibly the foulest sounding amps, ever released by ANY hi fi company. It was reliable, though.
Trevor
... when he identified AA as the best audio forum on the Web.John and Trevor,
I once watched a frined pop the electrolitics in his 500, not once, but twice! Both times he was trying to test at full power on a McAdams tester, but the caps blew like firecrackers before clipping set in. The second time it happened, he was trying to prove to everyone that the first time was a fluke. Both times he was showing off the new gear to a crowd, and both times Marantz repaired the amp. After the second incident, the amp was returned with a note requesting that he not do any more testing on the McAdams. I guess those oil-cooled resistors just presented too complex a load for the poor thing. It sure was an embarassment to Dennis.
I actually sold one of the things to a guy once, but only at his insistance. Not that it sounded bad, but after the confetti came out of two of the things I found it hard to recommend them.
Interesting conversation guys.
Thanks,
Charles
The McAdam, I used, had air cooled, aluminium-clad Dale, 250 Watt resistors, mounted on a large heatsink. I melted the solder, on the cables, connected to the resistors, under full power tests, with my Marantz 500, once. The 500's most serious problem, was the fragility of the output devices. There are two electros, used on the driver supply, which were prone to failure. I wouldn't like to see the main electros fail, though. Those puppies were BIG. The mess would be terrible.Trevor
Steve, we have been at this amp business for a long time. We've also done the research and published it decades ago. We have heard it all before. Now, the .5V/us per volt "rule of thumb" is related to the volts peak to peak of the audio signal, since a square wave can start at the - and end at the + extreme of the amp. I won't quibble over 40 vs 50 V/us. Most of us do over 100V/us anyway because we can do it easily today with modern transistors/FET's and it's probably better to be on the safe side. Slew rate limiting is essentially 100% distortion of part of the audio signal. To keep this distortion below 1% or so on this rapidly changing part of the audio signal, it is important that we not approach clipping the input or second stage of the amp. It is interesting that low slew rate amps usually have very nonlinear input stages, because emitter degeneration is typically omitted, and this makes the situation worse. This is why we suggest at least a guardband of approximately 5 times, in order to not generate significant TIM below the slew rate limit. We have found transients that approach .1V/us pp and this would lead to .5V/us perV pp for an amplifier. I have measured it myself with a fast storage scope and mistracking MC cartridges after RIAA EQ is applied. I hope this clarifies the situation.
john curl wrote:"Steve, we have been at this amp business for a long time. We've also done the research and published it decades ago. We have heard it all before."
I'm sure you have.
"Now, the .5V/us per volt "rule of thumb" is related to the volts peak to peak of the audio signal, since a square wave can start at the - and end at the + extreme of the amp. I won't quibble over 40 vs 50 V/us. Most of us do over 100V/us anyway because we can do it easily today with modern transistors/FET's and it's probably better to be on the safe side."
I was responding to what you said in your initial message, that there should be 0.5V/us for every volt of output "either + or -." Apparently that's not what you meant to say, but that's what you did say and so I took you at your literal word (it's been perhaps better than a dozen years since I read your's and Otala's papers so none of this is still on the tip of my brain). + OR - would indicate peak voltage, as opposed to peak-to-peak which would be + AND -. And for 40 volts peak, that would have equated to a slew rate of 20V/us instead of the 50V/us you specified and that's why I said it didn't appear to add up.
se
Steve, typing on the fly makes for bad grammar, or at least confusing statements at times. I agree that my statement was ambiguous, after reading it later.
Actually ss usually sound faster, at least subjectively. However tube amps usually has greater and more fluid/continuous speed gradation from slow to fast and THAT is what counts. :-)
I assumed the speed of an amp is determined by what the signal sees from input to output. Similiar to Aerodynamics it has to pass through many different paths and materials. The more pure the path the faster the speed of flow. As SS amps have tons of circuts on pcb boards and tube amps, with their simplicity (point to point wiring and so on), provides a quicker route out for the signal.Computer chips are also tested in this manner and if you put the point far enough there are dramatic speed differences in the same chip.
. . . this makes nice speculation, but it really has no bearing on audio circuits. Digital circuits depend upon delay, but audio circuits are not clocked. It really doesnt matter how long it takes a signal to propogate in an amplifier, as long as both channels are essentially the same. The speed of an amplifier has nothing to do with such propagation times.See a true experts post below. John Curl has more experience in audio design, and more respect than just about anyone alive.
Suzy, this explanation is really dumb! There are significant differences between tubes and ss, but the fact that tubes are somewhat more class A than solid state, does not change the 'speed' of the amp. Technically, ss is usually faster, because it can have a higher slew rate than tubes and much higher peak current output as well. Tubes are usually 'smoother' sounding because the harmonic distortion is primarily low order. This is not always so with ss. Many tube designs slowly roll off the high frequencies, well before solid state amps would begin to roll off. This is normal, because tubes have less global negative feedback than ss, and feedback tends to extend bandwith. I would be very wary of this salesperson, in future.
After several hours of listening to the VAC 770 last night and this morning, I would have to say this dealer's theory about tubes being faster than SS is definitely wrong! I will write more later, but my first impression of the VAC amplifier is that it is probably not my cup of tea.Suzy
Stick with tubes Suzy! They can sound very, very, good. I have spent a lifetime trying to beat tubes at their game. I have only partially succeeded in getting there. I still have some modified Dyna MK-4's with good tubes that I listen to occasionally, to remember what I am competing against.
Hi Suzy and John,I only read a few scattered posts here-and-there in the Asylum, but you two are going to change that if you're not careful. Suzy, I've been reading a few of your posts for a while now, and I think you're a major asset here in the Asylum. I've read every word in this particular thread, and I think its one of the best, if not the best, thread I've seen on the forum to date. A good question, well asked. This is the way the Web SHOULD work.
As for you John, your posts, the few I've read at least, are consistently excellent. I've simply got to find more time for reading this stuff. I wish you'd respond to more of the theoretical posts to help sort out some of the confusion, but I think I can understand why you don't. You are wise for your years grasshopper. :-D
This is fun. Thanks guys.
You too moderators. This is excellent.
Enjoy,
Charles
I always associate the term "fast" with overall integration of frequency ranges. That is, the bass is not so boomy such that it lags behind the rest of the music, the midrange is responsive and tight enough to not lag behind the treble, etc. Note that here I'm not necessarily referring to speaker designs (such as the "time-coherent" Theil or Hales designs) but rather the responsiveness of the entire system chain. As to whether tubes are generally "faster" than SS, I couldn't tell you. I also can't really provide technical information as to what allows components to be "fast." Just my opinions!
I agree. I also cannot explain these things, but know what I hear.
I must say that I've never heard of resistors being able to turn on and off as they are passive devices. Quote: "All general statements are false."I don't now exactly what it is that determines the sonic description of "speed" of an amp but my gut feeling is wide frequency respose on both ends of the audio spectrum (10hz-40Khz or wider), high slew rate, and a low impedance power supply which is characterized by large power transformers and copious amounts of capacitance.
As for SS Vs. Tube "speed," both devices can be made to sound "slower" or "faster" than the other depending how they are implemented in the circuitry.
Tom §.
sorry couldn't resist that.
but seriously this is something i have also been wondering about in the
context of speakers, in particular woofers and subwoofer.incidentally i think your dealer meant transistors not resistors.
apart from frequency response (which is at low power) there is also
power bandwidth--which is the response at high power.low impedance helps in delivering current and coping with strange loads
but the impedance has to be uniform or very low at all frequencies used
or even beyond since the effect on phase response might be bad.there are amp makers who consider wide bandwidth into the megaHz
region very necessary for good sound such as spectral and goldmund and
both these are solid state. i am not aware of tube makers who
emphasise this although my experience in very limited--only used arc
and sound valve tubes to date. the jensen transformer guys also have
interesting paper on their web site stating the phase change over the
audio range must be less than ten degrees if i remember right and this
implies flat power response for several hundred Khz and down to few Hz.so back to the speakers---if the cones are heavy are they able to start
off at the same time as the mids and the tweeters, and is this why so
many makers use lots of little cones--apart from the fact of cone
breakup common in large cones? power/weight ratio comes into this of
course and if the panel drivers also get help from the air itself in
terms of damping the motion then so much the better.leelock
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