What would you want to see in a book on electronics for vinyl replay? Douglas Self.

You could checkout Welcome to PyInstaller official website which works for some people so I read.

Right now Anaconda Python downloads just about every dependency you could ever want, certainly more than enough for these problems. I don't see this any more of a hassle than having the right version of Javascript enabled.
That succeeded. I wrote a tiny little Python GUI which implements a crude LCD clock:
Code:
import sys
from PyQt5 import QtCore, QtGui, QtWidgets, uic

qtCreatorFile = "MyClockGUI.ui" # Output of QtDesigner
Ui_MainWindow, QtBaseClass = uic.loadUiType(qtCreatorFile)

class MyApp(QtWidgets.QMainWindow, Ui_MainWindow):
    def __init__(self, parent=None):
        QtWidgets.QMainWindow.__init__(self, parent)
        Ui_MainWindow.__init__(self)
        self.setupUi(self)
        timer = QtCore.QTimer(self)
        timer.timeout.connect(self.showlcd)
        timer.start(500)
        self.showlcd()

    def showlcd(self):
        time = QtCore.QTime.currentTime()
        text = time.toString('hh:mm:ss')
        self.lcdNumber.display(text)

if __name__ == "__main__":
    app = QtWidgets.QApplication(sys.argv)
    window = MyApp()
    window.show()
    sys.exit(app.exec_())
Then I applied PyInstaller to it, and got a self-contained Windows .exe file that runs on other Windows computers which do not have Python installed. Yaay! However there is one little snag (see Figure 1 below): the .exe is thirty thousand times larger! 1Kb input, 29Mb output. Cowabunga. Figure 2 shows this marvel of engineering in action.
 

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My publishers inform me that the 2nd edition of Small Signal Audio Design has reached the practicable limits for the size of a book. (It is 750 pages long) Perhaps it is as well not to write a book you can't pick up with one hand.

I therefore propose to split off all the material in SSAD2 on phono amplifiers and vinyl replay generally, add a bunch of new stuff which I have thought up since, and make it into a separate book.

This is some way down the line (next year at earliest) but I would be very glad to hear opinions on what the book should contain, especially stuff not currently in SSAD2.

As a diy'er in the margins I would first ask who you are writing the book for.
If it's for seasoned professionals, which is how it looks it could turn out from the responders in this thread, the book will be laborious for those at my level to read and I'm unlikely to do more than scan the preview on Google books.

On the other hand, if the writing doesn't exclude those with less depth of experience, then I'd ask for a thorough treatment of filters. Passive first , then active.

In my view the thing that is unique to the phono stage is the filter and judging from the number of papers written by engineers explaining filter concepts or design topologies to other engineers I'd hazard a guess that they require a more thorough introduction than is presented in EE courses.

From the notes I've seen published online as well as the back and forths between EEs on the forums I believe that even to the initiated they are either easily misunderstood or that designers can make the mistake of being too unilateral in their approach and miss unintended consequences of what appears to be a simple choice.

A well written work on filters in a language that doesn't make references to unexplained advanced concepts would be a blessing to a reader like me.

Thanks
 
The good Mr. Self has allready written a book on filters:"The design of active crossovers". As the name implies the focus is on filers for crossovers, but there is also a lot of general filter "stuff". Definitely worth reading.


Take a look at Texas Instruments Analog, Embedded Processing, Semiconductor Company, Texas Instruments - TI.com and search for Filterpro, a free program for you to download and play with. It lets you design relatively complex filters without having to dig deep into the math-part.

Another must have for diy and professionals alike, is the book "Op Amp Applications Handbook" from Analog Devices. There's a large section on filters and of course a lot of audio and op-amp stuff. 800+ pages.

You could also take a look at http://sound.whsites.net/ where you can find a some projects on filters and some easy to read articles on filters and a lot of good audio related stuff.

And finally - if you are serious about filters - take a look at "Electronic Filter Design Handbook" by A. Williams and F. Taylor from McGraw-Hill.
 
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That succeeded. I wrote a tiny little Python GUI which implements a crude LCD clock:
Code:
import sys
from PyQt5 import QtCore, QtGui, QtWidgets, uic

qtCreatorFile = "MyClockGUI.ui" # Output of QtDesigner
Ui_MainWindow, QtBaseClass = uic.loadUiType(qtCreatorFile)

class MyApp(QtWidgets.QMainWindow, Ui_MainWindow):
    def __init__(self, parent=None):
        QtWidgets.QMainWindow.__init__(self, parent)
        Ui_MainWindow.__init__(self)
        self.setupUi(self)
        timer = QtCore.QTimer(self)
        timer.timeout.connect(self.showlcd)
        timer.start(500)
        self.showlcd()

    def showlcd(self):
        time = QtCore.QTime.currentTime()
        text = time.toString('hh:mm:ss')
        self.lcdNumber.display(text)

if __name__ == "__main__":
    app = QtWidgets.QApplication(sys.argv)
    window = MyApp()
    window.show()
    sys.exit(app.exec_())
Then I applied PyInstaller to it, and got a self-contained Windows .exe file that runs on other Windows computers which do not have Python installed. Yaay! However there is one little snag (see Figure 1 below): the .exe is thirty thousand times larger! 1Kb input, 29Mb output. Cowabunga. Figure 2 shows this marvel of engineering in action.

And to think my first computer had 1k of RAM and 4k EPROM and cost me ~$500 back in '78 (Hate to think of what that is in 2017 $). Oh, and I had to assemble it myself and it did not have a case.
 
The good Mr. Self . . .

Hi there LLPe,
Thanks for taking the time to write all that but I wonder if we're on the same page here.
My post above is simply a direct response to the " What would you want to see in a book on electronics for vinyl replay?" question as was your original post.
You are interested in seeing him write about noise issues and I'm interested in reading more about filters. Both subjects have been treated elsewhere - voluminously - but that doesn't mean Mr Self couldn't shed light on it in a way that gives additional understanding to those who read his work.

I've read some of Self's work and find his writing clear and easy to follow. I'd hoped he would write about passive before active because it's my experience that when the author covers the building blocks of what he is about to teach in his own words, the basics will be described in the same language that is used later on and that makes the later complexities easier to understand. When you have to look elsewhere for those explanations it is often not as easy to integrate them back into the picture the author is painting.

Anyway, when I reviewed the thread I also saw that Mr. Self has already decided that passives are a waste of time. I wonder if I could persuade Mark to write a paper . . . . . . 🙂

Thanks
 
Regardless of what this book contains I suspect we all will buy one. What would please me is the simplest possible way of doing any EQ I want be looked at ( 1/3 page ) . I think I already found this earlier in having passive 75 uS with variable gain then active 3180/318 with optional 2 uS final passive filter ( very useful with some amps ). To vary that is very easy and accurate. ( I usually make the 3180/318 gain of 16 at 1 kHz ).

When I measured a design like this many years ago I thought it rather better than suggested in text books. If the signal exits at a typical 15 mV at stage 1 ( 1 kHz ) that could be in very unusual circumstances 150 mV at 20 kHz. If we say 8 Vrms as the output max we should look at 34 dB to play with, more if stage 2 ( + 75 uS ) is a light load. This is attenuated by circa 20 dB when passed on ( at 20 kHz and greater reduced in the area suspected of being harmful ). This means from then on the design problems are like a full active design with reduced bandwidth requirements ( and maths ).

It always seemed to me that the arguements for overload take a very narrow view without any experimentation attempted. I would suggest the real requirement for overload can be as low as 6 dB. This will depend on circuit type and how the next stage likes things. I like passive-active as often the active stage can drive a volume pot into the bargain. When using a Shure M44-7 this design could work with a stage 1 gain of 1, 3 is likely to be ideal. For some MC gain of 60 ( not a problem ). As 15 mV out max is very low I use class A SE using a current source to -ve rail. However it is likely that the op amp is in PP class A if lightly loaded. My preference is possibly that the overal combination of SE stage 1 and PP class AB stage 2 is inserting a little curve distortion that balances out the 3 rd harmonic bias of a feedback design. I don't go for SE on stage 2 as to me it sounds a bit limp.

My idea of good audio is an ECL86 and Decca Deram PU on steroids ( Ongaku being a step too far ). That is 5 devices between LP and me ( set gain to suit ), two of those are transducers. I have 2 volumes of the AES on the history of the LP ( 1953-80 including local hero Percy Wilson who set 63.5 + 119 mm zero points circa 1928. Many of the pages of Vol 2 are devoted to ceramic PU's. Some were very serrious attempts at good engineering and at least as chalenging as the famed all in one RIAA calculations ( many pages shown and very complex ). I fitted a LP12 Ittok with a Sonotone 9 ceramic and found the sound to be very like MC except more open ( held with Blutak to damp and set mass ). Alas one could hear the LP being asked to work hard. For all that I never forgot how open the sound was. I did play with loading. I settled on simple loading as usually used. One can use something like RIAA if the ideal resistance loading found. I was happy with that as I had no real wish to continue. I feel a certain pride as at least when I offer an opinion I did invest time to find out. I am often wrong. That's what finding out risks. I would say 90% of what I do is a waste of time and I should have known better. 10% isn't and perhaps 2% is OMG!

AES Vol 2 Page 244 ( Hallgren )

Noise ( 2N4001 reference )

PU ( Shure typical ) into 47K 78.7 uV
Passive parts. 12.9 uV
Semiconductors 38.0 uV ( seldom bettered )
Total 88.3 uV

His figures 67.10 db weighted 2mV/1kHz.

This seems to tally with figures given by Self. I found this very useful as it allows me to say we can't better this in reality. Try measuring it, it's very close to the truth. Thus to chase a -80db figure is for fun and pride mostly.

Extra bit of data ( Real example, Shure M75 RIAA weighting and IEC suggestion )
Ideal 67.95 dB ( 69.64 IEC )
uA 741 = 54.73 dB ( 62.08 IEC )
uA 739 ( circa 5 nV/rHz ) = 65.08 dB ( 68.7 IEC )

IEC is IEC 268-1 A curve.
 
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If you are really serious about filters you need this book in your library. It is outstanding.
https://www.amazon.com/Handbook-Filter-Synthesis-Anatol-Zverev/dp/0471749427

It's title suggests what's to come after the 1960's with no real vision of what it would be ( I infer from the index ), all the more fun for that. The preview is too small to say more.

One thing that might favour passive designs is the phase shifts are exactly as given by the 3180,318,75 uS ( and I am sure the active when done with care ). The thing to understand is a typical LP is made using a test cut with a RIAA playback amplifier. AB by ear is used to check the sound is close to the cutting engineers plan for the cut. It might be out of need the cut sounds different, for example the bass. The very typical SME 12 inch arm and often Shure M97 or whatever colours the sound correction ( see Shure curves ). I have to say the sound of the cut can be very close to the master, at 78 microgrove it is almost a replica. I have pointed out this before. A CD recorded from a master I heard was like a tribute band, the CD was for sale so not to prove LP better. Before anyone runs away with thinking that's because the master was cooked up for vinyl I should point out this was just AB testing. Many have never heard the dynamic range these tapes have. Much of what we seek is the sources we don't own.

Very few cutting lathes are exactly RIAA. RIAA gets the best out of them . It is the playback that is arrived at using RIAA ( AB via RIAA ). The reason for this is the lathe is a mass of compromises and RIAA if you like sets the colour cast. 3180 uS I am told was an RCA recomendation that should work well.

I asked a German engineer why RIAA +/-0.05dB was better than +/-0.2 dB. His answer was we should at least get right what we can. I like that answer. It has no reality. What toubles me is when a newby to DIY is strongly told something is wrong when by analogy the critic is seeing infrared and ultra violet, they can't. I note a return to the 1970's when everything measured to be " good " is good. In my repair work usually things that measured well were good and also things that measured less well could be equally good. Some things measured almost badly yet sounded very good and Radford was the opposite ( HD 50 ). The Radford even looks on paper how it should be done. One day I should buy one and find out why. B&O looked awful technicaly, yet were OK. I think B&O knew what they were doing and made things as simple as possible. The designs look so unlikely to work ( crossover distortion ). The research dept of B&O was second to none. Sadly much of it never got into their products. Could be they found out it wasn't the prime factor. Sad they didn't have a make like Quad to show off their best. Everyone in the industry disliked B&O, me too. Didn't make me hate them when I heard them. Sonab should have been better, not sure it was as good.

One engineer told me the measurements of an amplifier came good quickly, the sound took time ( 18 months if I am right ). The amplifiers were usually tested mk1 to mkX and usually were almost identical. However the sound wasn't. Subtle differences like decoupling and PSU type was the key to better sound. In less critical markets where price was King the mk1 was sold. He reflected that the praise of the techno's was not translated into sales to Joe Public who only knew what they liked without any science to help them. The NAD 3020 being the absolute example.

I would imagine if Mr Self made this book a little bit more imformative with his own preferences stated he might sell 300%. I suspect 20% more pages to do so. 2 pages of the how and why of obsolete EQ's would be helpful as might using it as a microphone amp ( 1/3 page ? ). If not it is Rod Elliot alone at ESP.

http://www.audiosignal.co.uk/Resources/Why_do_equalisers_sound_different_A4.pdf
 
( 1953-80 including local hero Percy Wilson who set 63.5 + 119 mm zero points circa 1928.

Funny how many folks are baffled by high school trigonometry. Also Nige I prefer the approach of the National app note of 1977 for noise budget where the real part of the total cartrige source impedance vs. frequency is considered weighted by the RIAA and then the chosen perceptive weighting. There is no simple one rms noise level.
 
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The reason Percy won was making it easy . People were fitting pick up arms to acoustic turntables. Some may not have done any maths. His method was easy. Many years later others found more accurate ways. Problem is most do not have the magnifying devices as used in surgery to see the 0.1mm positioning it needs to question Percy's easy way. Even if you do it is not easy to allow for where the true contact area is. I have seen pages devoted to this that ignore the reality of how it might be done. This is why a linear tracking arm is so much easier. You need only a line too adjust to which can be a slot machined in metal.

When in Cuba I watched a 3 hour TV program on geometry, one didn't need Spanish as every other word was the ones we could know from geometry. The teaching method was to show many similar parts of a divided rhombus ( Euclid ). I wish I had been taught this way.

In 2011 I was given a drug for what was thought to be Parkingson's. It made it harder for me to do things I thought were easy. It was found I don't have it, although I have something that shows on scans. I don't take the drugs and returned to being able do things much as I did including servicing the car, it takes longer as the only downside. Writting is a rare thing that sometimes I can do. Once upon a time I would calculate everything in my head and usually be right. In the drugged period I couldn't. Now I can again. I don't as I found there is no need. When people look at things and say " that's ingenious, how did you do that "I just say " not really, it's the gift I was given put to use ". I never see where I am in the world as more than that.
 
Dear mr Self:

I would like to see current input phono amplifiers discussed and designed. Pros and cons.
With current input, I mean that the cartridge "sees" a dead short, and thus the cartridge supplies only a current to the phono amplifier.
It is in fact an IV converter.

Some cartridges work better than other regarding the current capability (low DCR, strong motors).

There are several commercial current input amplifiers on the market.
Erno Borbely co-designed one for Linear Audio.
 
He won't I'm sure. If I know our Mr Self he would do it if you had data you could offer. I must say it seems very reasonable it should work. It isn't exactly a short as the arm wires are about 1R5. Know what you mean. I ran a Reslo microphone into the emitter resistance of a common base amplifier without an external Re as best sound I could get. Even ran the current it needed through the ribbon( far below 0R1 ). Usually a transformer was used. The sound was very remarkable. The Reslo wasn't noted as a top performer. I suspect the usually used transformer was to blame. I could imagine this would work for a pick up.

I use the PU as the 75 uS. I then use 3180/318 uS active with 2uS passive ( no 20 Hz filter ). I have to do it by maths alone when 75 uS ( sounds correct ). If Mr Self had a calibration LP to go with his book I would like that. £20 is about what people pay for an LP's. If you think about it RIAA that included the pick up would be a first. It never will be perfect. It can be best we can do.
 
In my view the thing that is unique to the phono stage is the filter and judging from the number of papers written by engineers explaining filter concepts or design topologies to other engineers I'd hazard a guess that they require a more thorough introduction than is presented in EE courses.

Are you referring to the RIAA equalisation or to subsonic filtering?

If the latter, that chapter is now written, and I think I can say without fear of successful contradiction that it is the most thorough account of subsonic filters ever put together.
 
Funny how many folks are baffled by high school trigonometry. Also Nige I prefer the approach of the National app note of 1977 for noise budget where the real part of the total cartrige source impedance vs. frequency is considered weighted by the RIAA and then the chosen perceptive weighting. There is no simple one rms noise level.



Here is the app note 'AN104 - Noise Specs Confusing?'

http://www.ti.com/lit/an/snva515c/snva515c.pdf
 
Years ago I used LT spice to look at the noise of a simulated cartridge. I believe I modeled the cartridge as 0H5 in series with 1k. To my surprise, the terminating capacity made little difference in the noise plots.

It's the 47K||100pF||.5-.8H you can't get a resonant peak without two reactances. We are talking about the noise current contribution of a low noise bipolar op-amp and MM cart. Also there's no chance in most setups that there is not a contribution from cable capacitance. I posted a picture here of an LT1028 and a Stanton cart (physical measurement) pretty dramatically showing this.

Doug, this would make a nice addition say an Excel sheet as an appendix where the user simply enters cartridge/cable parameters and op-amp specs no SPICE needed.
 
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