Hi,
Agreed. Simply put a 60dB L-Pad on the generator (51K:51R will do nicely) and adjust for 10mV output on the Amp (signal). Don't attch a load. Clip out the 1.8K resistor, however this may trigger the DC Protection, so don't use a cliplead but a large value capacitor reliably discharged.
Ciao T
Agreed. Simply put a 60dB L-Pad on the generator (51K:51R will do nicely) and adjust for 10mV output on the Amp (signal). Don't attch a load. Clip out the 1.8K resistor, however this may trigger the DC Protection, so don't use a cliplead but a large value capacitor reliably discharged.
Ciao T
Hi tvrgeek,
Your equipment is nice sounding stuff. I did warranty service on all of it. Note that part of Nakamichi's magic lay in the details of the power supply for all of their signal level stuff.
Denon actually did make some very nice units. Their CD players were actually very hard to beat. I ended up with a DCD-S10 CD player. Really nice.
The higher end with the Marantz name is another very good effort. If you are familiar with the Marantz products of the late 70's - early 80's, you would notice that there is a progression of design. The Marantz designs have actually continuously improved, skipping over that horrible Philips gap of course! The early products can sound extremely good with some effort. The basic circuits are sound.
I haven't followed McIntosh that closely. Their designs have been reliable, although I'm no fan of the auto-former output designs. The ones I worked on in the mid 90's were much better sounding than the earlier models. Yes, I had a pair of MC-40 tubed mono blocks. They always sounded "cold" no matter what I did. Didn't miss them, but the 240 and 275 sounded much better. Go figure.
Your system will certainly reveal problems with amplifiers or speakers. Once it's set up the way you like it, you will enjoy it for many years. It's all reliable stuff. The only problem with the Nak CA-5II are the tone switch contacts - big shock. It's best to clean the switch contacts with a Q-Tip and contact cleaner. That means it's coming apart. The CA-7 types used a special (discontinued) Fujitsu relay. No direct replacement. Staying with the "5" model was either good luck or management.
I do like some 1" soft dome tweeters though. Peerless had some really nice ones a while back. Cheap ones do hurt the ears (like Philips ... or worse). Metallic domes can be brutal if they aren't good ones. Even the good metallic domes can hurt when your electronics chain misbehaves.
Hi John,
-Chris
Your equipment is nice sounding stuff. I did warranty service on all of it. Note that part of Nakamichi's magic lay in the details of the power supply for all of their signal level stuff.
Denon actually did make some very nice units. Their CD players were actually very hard to beat. I ended up with a DCD-S10 CD player. Really nice.
The higher end with the Marantz name is another very good effort. If you are familiar with the Marantz products of the late 70's - early 80's, you would notice that there is a progression of design. The Marantz designs have actually continuously improved, skipping over that horrible Philips gap of course! The early products can sound extremely good with some effort. The basic circuits are sound.
I haven't followed McIntosh that closely. Their designs have been reliable, although I'm no fan of the auto-former output designs. The ones I worked on in the mid 90's were much better sounding than the earlier models. Yes, I had a pair of MC-40 tubed mono blocks. They always sounded "cold" no matter what I did. Didn't miss them, but the 240 and 275 sounded much better. Go figure.
Your system will certainly reveal problems with amplifiers or speakers. Once it's set up the way you like it, you will enjoy it for many years. It's all reliable stuff. The only problem with the Nak CA-5II are the tone switch contacts - big shock. It's best to clean the switch contacts with a Q-Tip and contact cleaner. That means it's coming apart. The CA-7 types used a special (discontinued) Fujitsu relay. No direct replacement. Staying with the "5" model was either good luck or management.
I do as well. Ferrofluid tends to make the driver non-linear. It does help dissipate heat, but that should have been dealt with by the designer of the speaker system. Most of the time I see it being used when the crossover has insufficient cutoff slope or the tweeter is crossed in too low. If the power requirements are that high, change the tweeter type or run with multiples.
I do like some 1" soft dome tweeters though. Peerless had some really nice ones a while back. Cheap ones do hurt the ears (like Philips ... or worse). Metallic domes can be brutal if they aren't good ones. Even the good metallic domes can hurt when your electronics chain misbehaves.
Hi John,
Can't argue that one bit!
-Chris
Chris DA MAN!
Completely agreed on the Marantz comment. However, I have not followed their higher tier products, but their midrange stuff.
In my view, they reached their peak with the series produced 1978-1980, of which the best known models are 1152 DC, 1180 DC, and 3250B pre and 170 DC power amp. That was the last series which was designed in the U.S.A. and manufactured in Japan.
After that series, the only effort worth mentioning is that they tried making even their small stuff with electronically regulated power supplies, but didn't get anywhere with it, and it was soon dropped. Around 1985, they arrived at a topology they still use to this very day, a good example would be their model 520.
They were also the first to invent a local icon, a man backed by their marketing department and hyped up to be the genius's genius - I refer to the colorful Mr Ken Ishiwata, whose pictures can be seen in their ads from around 1988 onwards. Personally, I remain unconvinced, to put it mildly.
The trouble with vintage gear is that most people don't realize that to get the real deal, you do need to have it fully refreshed - after 30+ years, capacitors do tend to dry up. While refreshing costs time and money, it is well worth the effort - do it, adjust it as per the service manual, and you will be rewarded by a quality of sound which will more often than not make you wonder what has the audio industry been doing for the last 30+ years. Not perfect, but way better than most of the modern products.
Thorsten, I have learnt long ago that specs are one thing and the sound is another. It may measure great but still sound poor, or it may measure just barely so-so, yet still sound wonderful. I have been "ambushed" by devices which have no right to play as well as they do, looking at their architecture and open loop performance.
My H/K 6550 is rated at 0.3% THD, full power into 8 Ohms; when measured, it actually does 0.2% under the same conditions. However, start reducing the power and the THD decreases as well. By the time you get to what we actually use in rooms for normal listening, THD is down below 0.05%.
And 0.07% THD peak is not my dictum, it's simply a proposed starting point. If you feel that is too much, no problem, you propose another value and let's get the ball rolling. I don't insist on anything in particular.
And what I do insist on, I use in my own work. My view is that we need power more as a reserve than an everyday need, at least in most European rooms, which typically have up to 30 m2, and using reasonably efficient loudspeakers, say 90+ dB/2.83V/1m. Those with less efficient speakers, like say the Dahlquists (82 dB/2.83V/1m), and/or larger rooms, will need more power for a faithful rendition of hard transients. We do however need a high load tolerance, and this costs money in output stages and power supplies, so it is often reserved for more expensive models, which are forced by their competitors to also be more powerful.
The only dictums I do have apply only to my own work; I may think so, but that's no guarantee that it is really so. And since most fathers love their babies, how well I do the job is not for me to say.
The only dictum most people involved in commercial work must adhere to is that of product economics. Ask John how he feels every time he sits down to design whatever, and the first thing he gets as a must is the price. One then has to squeeze whatever one can into that price bracket. Compromises we don't like simply have to be made.
But I'm reasonably sure you've been there, done that.
This here is different insofar that here, I can always say "damn the cost" because I'm doing it for myself.
This is a very interesting point, of commercial work vs economics. Having recently been through design and product successfully being sold on eBay, the gathering of materials to bring about the test of that design...takes extraordinary time patience and skill.
My principles are to get the best possible sound, whatever it takes. I use the best components, however I also use components in clever ways that others sometimes just bypass. For instance rather than large caps, I use transistor based cap multipliers, because they improve audio. It was impossible to do this with design detailed in Powering op amps thread as ground is not identified as a pin out with op amps ( rather ground is external ) however very good audio improvement was found with using transistors.
Cheers / Chris
Anatech,
One of the tweets I was looking at was the HDS. I expected it to be the ringer at twice the price of the others, but the SB and Vifa came out better in my book. Not to cast any doubts on the HDS, a very fine tweeter. I have a suspicion whatever my wife can hear is in fact an interaction of a missbehaving tweeter combined with something going on in the amp. Something there, or maybe something missing, don't know.
The Studio 20's I have are barely acceptable, the Vandersteins were fine, just too ugly and the Sequals too expensive and too big. Nothing else in a store ever passed her instant no, no, no right down the line. Whatever it is. is really clear. Epos through a pair of Cary's was OK too, but not on solid state. B&W, Kef, Canton, and most of what you find is stores are a no. I never got her to hear Thiel or Snell, a couple of my favorites I can't afford.
None of my own speakers has yet matched the Paradigms, but I am getting close. I think I am ready to spend some for the much better drivers as my crossovers and boxes are about ready. I am using up a pair of Seas metal domes with Zaph mids as a final test of my crossover design craft. The very narrow box was a bigger challenge than I expected for refraction issues.
One of the tweets I was looking at was the HDS. I expected it to be the ringer at twice the price of the others, but the SB and Vifa came out better in my book. Not to cast any doubts on the HDS, a very fine tweeter. I have a suspicion whatever my wife can hear is in fact an interaction of a missbehaving tweeter combined with something going on in the amp. Something there, or maybe something missing, don't know.
The Studio 20's I have are barely acceptable, the Vandersteins were fine, just too ugly and the Sequals too expensive and too big. Nothing else in a store ever passed her instant no, no, no right down the line. Whatever it is. is really clear. Epos through a pair of Cary's was OK too, but not on solid state. B&W, Kef, Canton, and most of what you find is stores are a no. I never got her to hear Thiel or Snell, a couple of my favorites I can't afford.
None of my own speakers has yet matched the Paradigms, but I am getting close. I think I am ready to spend some for the much better drivers as my crossovers and boxes are about ready. I am using up a pair of Seas metal domes with Zaph mids as a final test of my crossover design craft. The very narrow box was a bigger challenge than I expected for refraction issues.
"Thorsten, I have learnt long ago that specs are one thing and the sound is another. It may measure great but still sound poor, or it may measure just barely so-so, yet still sound wonderful. I have been "ambushed" by devices which have no right to play as well as they do, looking at their architecture and open loop performance."
Exactly. RB 951. Why? This is why I have been learning what I can about amps for the last 3000 posts.
Exactly. RB 951. Why? This is why I have been learning what I can about amps for the last 3000 posts.
Marantz 2270 in the walnut case. Yea I dreamed. I had a KLH 51 back then and , gasp, L26 JBL's. That era equipment controls had weight and feel. Big old tuner dial! You could tell a Marantz from a Yamaha from an Onlyo, Luxman, or whatever from across the room, and could identify most by the sound. My old Denon DRA-35 was a jewel of a cheap receiver only moving to my garage when I needed a much better tuner due to the noise from HDFM. (Replaced by a Kenwood tuner and Creek amp)
This Kenwood 9150 I need to fix was from that era. The phono stage is out, so is should be repairable. I wonder how far the owner will let me go with restoration. My web crawling suggests it was one of the good ones. Main caps, coupling caps. He knows how good it is.
At least the Nak CA-5 and ST-7 fit on my office shelf. I need to decide on the amp. ( My office, so no wife super hearing limitations). I have stuck with a NAD C520 CD as it fits on the shelf and is not a bad unit. Not great, but not bad. My Denon DVD 1940 plays SA and sounds decent, but the tray is so slow it ticks me off. My Rotel RCD 1070 sounds the best, but is way too big, so it is on my living room system. All of them will go in favor of external DAC's and a music server sometime.
This Kenwood 9150 I need to fix was from that era. The phono stage is out, so is should be repairable. I wonder how far the owner will let me go with restoration. My web crawling suggests it was one of the good ones. Main caps, coupling caps. He knows how good it is.
At least the Nak CA-5 and ST-7 fit on my office shelf. I need to decide on the amp. ( My office, so no wife super hearing limitations). I have stuck with a NAD C520 CD as it fits on the shelf and is not a bad unit. Not great, but not bad. My Denon DVD 1940 plays SA and sounds decent, but the tray is so slow it ticks me off. My Rotel RCD 1070 sounds the best, but is way too big, so it is on my living room system. All of them will go in favor of external DAC's and a music server sometime.
Tvr, IF you listened to me, you might be better off, and you might learn something. The Parasound power amps are designed with a FIXED closed loop gain, about 27.2 times, and the circuitry is made as LINEAR as possible (open loop) especially when operated at its ideal current on the output stage. This is between .015 and .026V across each EMITTER RESISTOR. Are you there yet? There is no other modification that is necessary for its designed in performance.
The feedback is maximized within the design's capabilities to supply open loop gain, without adding additional complication, such as extra followers or active loads on the input stage. The REASON that we use high feedback in Parasound components is because they are usually designed for home theater use, and they must meet the THX spec. which is independently verified by THX. THX means Tom Holman eXperimental, and he set the specs., not Charlie Hanson or me. To meet his spec. a fair amount of negative feedback is necessary, and that is why it is there. However, the basic design is optimized (usually) for best overall open loop linearity, the largest problem being the limitations of the heat-sink size on a lowish priced product. If you want a bigger heat-sink, pay more money, and you can have it with basically the same circuit, and get better performance.
The feedback is maximized within the design's capabilities to supply open loop gain, without adding additional complication, such as extra followers or active loads on the input stage. The REASON that we use high feedback in Parasound components is because they are usually designed for home theater use, and they must meet the THX spec. which is independently verified by THX. THX means Tom Holman eXperimental, and he set the specs., not Charlie Hanson or me. To meet his spec. a fair amount of negative feedback is necessary, and that is why it is there. However, the basic design is optimized (usually) for best overall open loop linearity, the largest problem being the limitations of the heat-sink size on a lowish priced product. If you want a bigger heat-sink, pay more money, and you can have it with basically the same circuit, and get better performance.
I agree, price should be a constraint, not a compromise. By this I mean for engineering optimisation (which is where the fun is) to take place in the design process, there needs to be a reason, a 'why' attached to that price. Only then can price enter the design process - if its a single figure then the designer will be hemmed in and not enjoy the designing as much.
Pricing is always a trade-off in practice, not a fixed sum. Margins, sales volumes, distribution channels etc. do have some flexibility.
There are some, mostly tube amps around, with switches which vary feedback, typically "O", "3 dB" and "6 dB". They do sound different at different settings, so if that works for you, go tube. I am not aware of any solid state designs with similar features.
Feedback is a VERY serious matter, I would say crucial for every amp. I for one would not mess around with variable feedback levels simply because you cannot optimize ANYTHING over a wide range - perhaps only over a small range, but that sort of defeats your idea.
Some amps do have usually two settings of bias, "low" and "high". Again, I see no particular point to this, again because you cannot optimize a fairly complex circuit for grossly different bias points.
Wayne, by definition, an optimal point is only one - theory of systems says so, there can be no two optimal points (see works by Oscar Lange et al.). I would strongly advise you read Norbert Wiener's book "The Human Use Of Human Beings". It will most clearly show the unbelievable similarity between electronics and cybernetics, as its equivalent in social sciences.
As John pointed out elsewhere, an optimal bias point is determined for each and every amp on a case to case basis. The optimal point is one which will yield a reasonable current flowing through the class AB output stage (since everything else begore it is running in pure class A) to reduce switching distortion and distortion in general, but without wasting a lot of power as in class A (around 90% of the power used is turned into heat) and without demanding very large heat sinks.
You can, usually very simply, increase the bias current of most amps (although some use fixed bias scehemes, e.g. Sony), typically without any ill effects, BUT your overall THD will actually deteriorate a little (because you have left the optimal point for that design and are in non-optimal territory) and your heat sinks will get hotter.
You can adjust this by ear only, monitoring the results with a simple multimeter. Assuming your heat sinks hold out (which is by no means guaranteed, especially in cheaper products), you will find that the available range within which you can hear the difference is fairly limited.
I tried that with my HK 680. The service manual states that it's bias should be 50 mA per output transistors (uses two pairs per channel). I cranked it up in 10 mA increments to 140 mA per trannie, at which point I could no longer hear any change in sound. Back to 130 mA, no sense surpassing it.
What I got was an audibly warmer sound, sort of more rounded off, kind of cuddly, but at the cost of some resolution. It sounded warmer, but it lost some of the finer detail. At first listening, this may seduce you to leave it there, but I guarantee that after a while, you'll be back down to say 70 mA. You can second guess the designer (in my case, Richard Miller) only so much. At least its heat sink held out well enough.
The key problem in this method is that by changing the bias, we actually change the operating mode of everyting after the bias adjust circuit, meaning the predrivers, drivers and output stage. It would have gone down much better if I could change only the output stage bias, as Otala did by adding resistors to the output stage only - that way, what preceeds is can still work at its own ideal bias point.
The concusion is clear anough - to have a high bias point, you need an amp specifically designed for it, and everything else will yield doubtful results. Second guessing the designer is not a wise thing to do.
Nice in theory.
Unfortunately, it doesn't work that way in big companies.
The problem is that there is far too much economics and showmanship in planning mass produced products. A big transformer, big capacitors and large heat sinks will make it sound better, but they are all inside the case and cannot be seen, whereas flashy LEDs are on the fascia and can be seen. Not many, if any, customers will even ask about the insides, they will only consider the outsides.
Hence, we have sheer madness out there. Audio products are judged by their sound less and less, and are judged by their looks more and more. These days, if your front plate is less than 10 mm thick, you don't sell. If your power on LED is not blue, it ain't High End, no way, hermano. If it doesn't have a remote, it won't sell. Show time.
And if it has all that, and actually plays music as well, all the better! It will actually sell.
Realizing this, many (but not all) Chinese manufacturers play this card. They have run the prices of manufacturing down into the ground and are playing the game of looks mostly - cheap manufacturing, crappy but cheap electronics inside, but hey! it's cheap and it looks expensive. And this world has become a cheapskate world, people want $1,000 value for no more than $100.
Let's face it folks, our worst audiophile nightmare is fast coming to life, and I guess that very soon, our beloved audio will become just one more VLSI chip in a tablet which does everything but wash your laundry.
Audio as we know and love it will progressively be pushed more and more out of the mainstream, and will live on only at the outskirts of the elctronics industry. It will be relegated to a handful of crazies like us only beacuse that's how we were brought up and we can't change skins now, it's too late.
Today, I can order digital amps for less money than the sheer material costs of an analog amp I want to make for myself. They don't need almost any heat sinks because they are 90+% efficient, they don't need as large transforemrs for the same reason, and in terms of economics, they wipe the floor with analog gear. The trouble is, they are still a far way off from sounding as well as competently made analog gear. Most don't care, most of them can't hear the difference anyway, but I can, so I do care.
I guess you're right, I'm just in the main fortunate enough to have worked for a company where it did work fairly well along the lines I've described. Such companies have 'competitive advantage' over the mainstream mass producers.
Yep - that's how the current market operates I agree. It is madness, so a new market opportunity presents itself - after all, 'profits are made from differential stupidity'.
Oddly, your audiophile nightmare is my audiophile dream. One (schizo) market dies as another (slightly healthier) one springs to life
My plan is the opposite - bring decent (but not religious audiophile i.e. 'good enough') sound into the mainstream, at a mainstream price where people pay attention more to the sound than they do to the means of production. Has worked in other fields, why not this one?
I'm banking on them getting there, eventually.
Hi,
THD is completely meaningless if we want to assess distortion audibility.
0.05% THD may already be way too much and objectionable, while 3% THD or even more may go unnoticed with music. For any funloving number of THD you care to choose* I can illustrate that it is the "wrong number". Because THD is an artifical measurement with no relation to distortion audibility.
* except around -120dB re 111dB SPL for all harmonics (that is < 0.0001% for any given harmonic), at all levels and frequencies
In order to assess distortion audibility for simple (HD and resultant IMD) Distortion we need to know:
1) SPL Level of the signal
2) Frequency of the signal
3) Level of harmonic and order
While Earl Geddes has provided more precise and evidence based weighting (and more stringent on higher orders than all earlier suggestions) for this type of distortion, the scheme D.E.L. Shorter of the BBC suggested in the 1950's# should suffice for a very decent quick approximation.
# D.E.L. Shorter suggested that the level of each harmonic shall be multiplied by N^2/4 where N is harmonic order. So for example H2 needs to be mulplied by a factor of 1, H3 by a factor of 2.25 and H7 levels need to be multiplied by 12.25 to get the equivalent audibility.
In other words, if we give 0.1% H2 an audibility score of 0dB (that is it is at the edge of being detectable), than 0.044% H3 and 0.008% H7 have a similar level of audibility.
D.E.L Shorters scheme does not account for SPL though
I would also submit a further thought, which is pure speculation of course...
We know that the human hearing mechanism is not a microphone, the hair-cells are not linear piezoelectric transducers and the analogue signal processing that happens in the grey matter that most humans are alleged to have between their ears (reading this board often makes one doubt the veracity of such an allegation) is not an Fast Fourier Transform of a limited windowed signal and so on etc. et al.
In his Heyser Memorial Lecture John Atkinson proposed that when listening to music (real or recorded) we effectively use an internalised model and judge and react on the basis of differences to this model.
What I would propose is that we as such do not directly react to the individual harmonics in themselves (in the way an FFT Analyser does), but rather we react to, for lack of a better term the whole transfer function, with no specific "audibility" of a harmonic as such.
As said, speculation, purely, lest someone knows research that supports my deductive/inductive leap of intuition...
I find it generally surprising how little the extensive work of a range of professional and industry bodies on many subjects closely impacting audio is among those who practice the craft.
Luckily enough we can take a a rather sturdy swordhilt to this gordion knot by looking closely at the EBU (adopted from IRT recommendations) for maximum SPL we come to 108dB continuous (or 111dB peak) if we look at THX we get 105/108dB so both sets of numbers are quite close.
Given that a stereo pair of speakers at 3m listening distance each fed with equal power gives around 3dB less SPL than the individual speaker at 1m we can easily work back to SPL requirements.
Basically we need between 111dB and 114dB peaks.
So for a 82dB/2.83V/1m Speakers we need 2.83V + 32dB = 113V peak (800W/8R).
On the other hand if we have (say) a speaker with 98dB/2.83V we only need 2.83V +16dB = 17V peak (18W/8R).
I suspect our 82dB/2.83V/1m speaker has long vaporised the voice coils should we use a sustained tone for this test.
Ciao T
THD is completely meaningless if we want to assess distortion audibility.
0.05% THD may already be way too much and objectionable, while 3% THD or even more may go unnoticed with music. For any funloving number of THD you care to choose* I can illustrate that it is the "wrong number". Because THD is an artifical measurement with no relation to distortion audibility.
* except around -120dB re 111dB SPL for all harmonics (that is < 0.0001% for any given harmonic), at all levels and frequencies
In order to assess distortion audibility for simple (HD and resultant IMD) Distortion we need to know:
1) SPL Level of the signal
2) Frequency of the signal
3) Level of harmonic and order
While Earl Geddes has provided more precise and evidence based weighting (and more stringent on higher orders than all earlier suggestions) for this type of distortion, the scheme D.E.L. Shorter of the BBC suggested in the 1950's# should suffice for a very decent quick approximation.
# D.E.L. Shorter suggested that the level of each harmonic shall be multiplied by N^2/4 where N is harmonic order. So for example H2 needs to be mulplied by a factor of 1, H3 by a factor of 2.25 and H7 levels need to be multiplied by 12.25 to get the equivalent audibility.
In other words, if we give 0.1% H2 an audibility score of 0dB (that is it is at the edge of being detectable), than 0.044% H3 and 0.008% H7 have a similar level of audibility.
D.E.L Shorters scheme does not account for SPL though
I would also submit a further thought, which is pure speculation of course...
We know that the human hearing mechanism is not a microphone, the hair-cells are not linear piezoelectric transducers and the analogue signal processing that happens in the grey matter that most humans are alleged to have between their ears (reading this board often makes one doubt the veracity of such an allegation) is not an Fast Fourier Transform of a limited windowed signal and so on etc. et al.
In his Heyser Memorial Lecture John Atkinson proposed that when listening to music (real or recorded) we effectively use an internalised model and judge and react on the basis of differences to this model.
What I would propose is that we as such do not directly react to the individual harmonics in themselves (in the way an FFT Analyser does), but rather we react to, for lack of a better term the whole transfer function, with no specific "audibility" of a harmonic as such.
As said, speculation, purely, lest someone knows research that supports my deductive/inductive leap of intuition...
I find it generally surprising how little the extensive work of a range of professional and industry bodies on many subjects closely impacting audio is among those who practice the craft.
Luckily enough we can take a a rather sturdy swordhilt to this gordion knot by looking closely at the EBU (adopted from IRT recommendations) for maximum SPL we come to 108dB continuous (or 111dB peak) if we look at THX we get 105/108dB so both sets of numbers are quite close.
Given that a stereo pair of speakers at 3m listening distance each fed with equal power gives around 3dB less SPL than the individual speaker at 1m we can easily work back to SPL requirements.
Basically we need between 111dB and 114dB peaks.
So for a 82dB/2.83V/1m Speakers we need 2.83V + 32dB = 113V peak (800W/8R).
On the other hand if we have (say) a speaker with 98dB/2.83V we only need 2.83V +16dB = 17V peak (18W/8R).
I suspect our 82dB/2.83V/1m speaker has long vaporised the voice coils should we use a sustained tone for this test.
Ciao T
Hi,
Sure, easy enough to implement.
Kind of like the "measure/listen" switch on some DAC's that selects between a filter that measures well (to illustrate "we can") and one that sounds good (to illustrate "we can but it sounds rubbish and this is what it should sound like").
Design an amplifier with decent open loop linearity (0.1% H2 dominant at rated power into 8R is non too difficult) and then add a feedback loop one way or the other with some 60dB NFB to be switched in to show "this is what 60dB of NFB and 0.0001% THD really sound like".
And adding a "optimum Class AB bias or overbiased Class AB" switch might be good. Overbias increases THD and H3 but REDUCES higher order components - so perhaps "optimum class AB bias" is much higher than THD minimum?
Ciao T
Sure, easy enough to implement.
Kind of like the "measure/listen" switch on some DAC's that selects between a filter that measures well (to illustrate "we can") and one that sounds good (to illustrate "we can but it sounds rubbish and this is what it should sound like").
Design an amplifier with decent open loop linearity (0.1% H2 dominant at rated power into 8R is non too difficult) and then add a feedback loop one way or the other with some 60dB NFB to be switched in to show "this is what 60dB of NFB and 0.0001% THD really sound like".
And adding a "optimum Class AB bias or overbiased Class AB" switch might be good. Overbias increases THD and H3 but REDUCES higher order components - so perhaps "optimum class AB bias" is much higher than THD minimum?
Ciao T
It is also my vision of what distortions are. If I would have opportunity to measure distortions in my preferred manner, I would first determine time shift between input and output signals, next I would take a fragment of "difficult" music (1 sec of very high quality recording of orchestral music), next I would compare input signal to "normalized" output signal, both precisely digitized. Normalization means both time shift and level scaling of the output signal.
Comparison will be based on definite metrics, this will be a subject for researches.
I guess you're right, I'm just in the main fortunate enough to have worked for a company where it did work fairly well along the lines I've described. Such companies have 'competitive advantage' over the mainstream mass producers.
Yep - that's how the current market operates I agree. It is madness, so a new market opportunity presents itself - after all, 'profits are made from differential stupidity'.
Oddly, your audiophile nightmare is my audiophile dream. One (schizo) market dies as another (slightly healthier) one springs to life
My plan is the opposite - bring decent (but not religious audiophile i.e. 'good enough') sound into the mainstream, at a mainstream price where people pay attention more to the sound than they do to the means of production. Has worked in other fields, why not this one?
I'm banking on them getting there, eventually.
Again, sounds nice in theory, but I very much doubt that any such practice will follow.
You and a thousand Chinese manufacturers have the same idea. Back to square one.
I am saddened to see fewer and fewer people pay attention to the actual sound. Mobile phones have introduced another aspect to us all, that of changeability. If you want to be in these days, you need to change your cell phone at least twice a year, or about as often as somebody introduces new "features", often useless and in fact bothersome, but hyped up like you wouldn't believe. We are encouraged by various deals offered us by cell phone proividers to change, change, change. A 3 month old cell phone model is worth less than 20% of its initial value, and a 6 month old phone you just can't sell for any meaningful sum.
Look at the American market - it is 100% saturated, NOBODY can even count who's in it any more, they come and go daily. I have abandoned all American audio sites except this one because they all boil down to the same main theme - I've had it all, what's left for me to buy now? They buy new because it's new and they are the first kid on the block to own it. In 3 months' time, when the neighbors have also got it, he's off to a new acquisition.
Not that elsewhere is much different, just less pronounced. Ask people what it sounds like after say 3 months of onwership, and you're likely to get an answer like - oh, I dunno, all right, I guess. After 3 months he GUESSES it sounds all right?
Fresh young blood - virtually zero. Kids don't give a flying f**k about sound quality, all they want is unlimited free quantitty. Enter the Internet, backed up with the glorious MP3 technology, which is an insult flung in our face. We screamed for 20 years that 16 bit was just not enough, they manage to give us a 24 bit format, but almost nobody goes there because tehy have to pay for it, whereas MP3 is free for all. Most of it ripped off, legally speaking stolen, but nobody seems to mind, people have come to think of it as their birthright.
So, my dear fellow - where are those customers you expect? Some will be there, of course, but so will a hell of a lot of your competitors, many of which will offer more external machining than you, at a quarter of your price.
Enjoy these days while they last, it won't be long now.
Hi,
Amplifiers with perfectly adequate distortion performance (that is not audible) AND damping factor (that is no appreciable differences in the actual damping factor at the woofer cone) can be made without global loop feedback. Local degeneration, which is generally considered negative feedback by some will be needed though.
That again is debatable. In the end it boils down to how we define "optimum" in technical terms and how we verify "optimum".
There can be only one optimal point if there is a wide agreement what "optimal" is, in respect to a given quantifiable quality.
If there is no such agreement, there will different "optimal points", especially when we need weigh up multiple and conceivably interactive qualities (as is the case for audio).
This is debatable.
What can be substantiated is the existence of a minimum THD bias point.
This Bias Points actually varies with signal levels (there is thermal memory component involved).
This bias point actually minimises THD and low order HD, but it leads to higher levels of high order distortion than could be attained using a different (higher) bias.
So the optimum differs if we want to minimise audible distortion or measured THD. Which then is the optimum?
I would argue that it is at a level of quiescent current where the given Amplifier no longer shows any appreciable reduction in higher order distortion components, EVEN IF THD IS INCREASED over the THD minimum.
D.Self would argue that it is THD minimum, though he'd probably start to waffle and wobble a little if you asked him to substantiate at what power level the THD minimum should be set and why it does not matter that this optimum point is signal dependent...
For the accountants D. Self Optimum is preferred, measured at the lowest possible power, as this minimises the need to spend money on heatsinks and power supplies...
Ciao T
Amplifiers with perfectly adequate distortion performance (that is not audible) AND damping factor (that is no appreciable differences in the actual damping factor at the woofer cone) can be made without global loop feedback. Local degeneration, which is generally considered negative feedback by some will be needed though.
That again is debatable. In the end it boils down to how we define "optimum" in technical terms and how we verify "optimum".
There can be only one optimal point if there is a wide agreement what "optimal" is, in respect to a given quantifiable quality.
If there is no such agreement, there will different "optimal points", especially when we need weigh up multiple and conceivably interactive qualities (as is the case for audio).
This is debatable.
What can be substantiated is the existence of a minimum THD bias point.
This Bias Points actually varies with signal levels (there is thermal memory component involved).
This bias point actually minimises THD and low order HD, but it leads to higher levels of high order distortion than could be attained using a different (higher) bias.
So the optimum differs if we want to minimise audible distortion or measured THD. Which then is the optimum?
I would argue that it is at a level of quiescent current where the given Amplifier no longer shows any appreciable reduction in higher order distortion components, EVEN IF THD IS INCREASED over the THD minimum.
D.Self would argue that it is THD minimum, though he'd probably start to waffle and wobble a little if you asked him to substantiate at what power level the THD minimum should be set and why it does not matter that this optimum point is signal dependent...
For the accountants D. Self Optimum is preferred, measured at the lowest possible power, as this minimises the need to spend money on heatsinks and power supplies...
Ciao T
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