That should correlate, but the model with lumped elements does not describe all the details. I have often found that the practical results are slightly better than the simulation based on lumped elements, just like simulating a transmission line with not enough cells produces pessimistic results.The bigger cores are on back order so they want come soon
But from the 3k9 source measurement one can extrapolate the freq response of my toroid with a 220r source impedance, I know the high end behavior is more complex than a low pass filter, besides capacitance it has leakage inductance etc... it has self resonanse I cant see.
In the low end the -3db is at 2Hz because inductance rises at low Hz to 16 H.
At the high end the measurement indicates -3db roll off at 120KHz
Would it be correct to calculate self resonance freq. by having total capacitance from freq plot: 5n8 + measured leakage inductance?
Valve mic preamp design incoherent rambling
- Thread starter GussyLoveridge
- Start date
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I discovered an error in my earlier measurment. The source impedance wasnt 3k9 it was really 1K3. Because the 3k9 resistor was in parrallel with 2K...
So lets set the record straight. Its less flattering but.. High freq roll off -3db at 40K. And like Jakob predicted there was no increased inductance at lower freq. Inductance is flat....
That put the Lundahls -3db at 80k under the same circumstances.
The flaws of capacitance are magnified trough the transformers impedance convertion. My toroid and lundahl starts out with pretty much the same total capacitance. Lundahl has 1:25 impedande ratio and the toroid has 1:42 that pretty much accounts for the earlier hi freq roll off...
I measured leakage inductance across the primary: 50uH with shorted (secondarys)
The 1k3 is reflected to secondary as 55K, -3db at 7Khz = 413p. Total capacitance.
With a 50 procent bigger core plus 50 procent higher primary inductance 10H. I think my next IPT should rival the THD to the LL1538.
The big challange is to get low capacitance. Interwinding C will be lower with electrostatic sheild. I dont think it is possible to progressive winding by hand, but many small sektions in 20-30 turns range (by winding the first 10 turns side by side and the rest on top of each other) will lower C between turns and evenly distribute winding through out the core... The last thing is by lowering turns ratio to 1:5 hi freq response will almost double...
The goal is to rival the LL1538 so to even the plane field I will go for a 1:5 toroidal design next... and for a DIY design maybe that is the sane end point...
So lets set the record straight. Its less flattering but.. High freq roll off -3db at 40K. And like Jakob predicted there was no increased inductance at lower freq. Inductance is flat....
That put the Lundahls -3db at 80k under the same circumstances.
The flaws of capacitance are magnified trough the transformers impedance convertion. My toroid and lundahl starts out with pretty much the same total capacitance. Lundahl has 1:25 impedande ratio and the toroid has 1:42 that pretty much accounts for the earlier hi freq roll off...
I measured leakage inductance across the primary: 50uH with shorted (secondarys)
The 1k3 is reflected to secondary as 55K, -3db at 7Khz = 413p. Total capacitance.
With a 50 procent bigger core plus 50 procent higher primary inductance 10H. I think my next IPT should rival the THD to the LL1538.
The big challange is to get low capacitance. Interwinding C will be lower with electrostatic sheild. I dont think it is possible to progressive winding by hand, but many small sektions in 20-30 turns range (by winding the first 10 turns side by side and the rest on top of each other) will lower C between turns and evenly distribute winding through out the core... The last thing is by lowering turns ratio to 1:5 hi freq response will almost double...
The goal is to rival the LL1538 so to even the plane field I will go for a 1:5 toroidal design next... and for a DIY design maybe that is the sane end point...
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True, but the capacitance to ground will increase. In most cases, the addition of an electrostatic shield increases the stray capacitance. And the leakage inductance will also increase. These are two factors that contribute to decrease the HF response.
So the main benefit of a electrostatic screen would be improved cmrr. Noise rejection is already good beacuse of symmetric primary. So maybe I should reject screen and focus on battle capacitance in the secondarys.... decreasing ratio and more symmetrically wound secondarys.... hmmm
Yes.
A xfmr primary is symmetric uo to a point, where the capacitances at both expremities of teh winding start to differe significantly, impairing HF CMRR.
Many input circuits, particularly German, used a trimmer capacitor to balance HF CMRR.
A particularly well balanced secondary will not result in a perfectly balanced circuit it it's connected to an unbalanced input, which is the most common case.
I wished Bill Whitlock (Mr CMRR here) would chime on it.
You sure your enemy is the distributed capacitance (winding-to-itself), not the interwinding capacitance, i.e. capacitance between pri and sec's?
Remember: distance is your friend here - have you tried physically spacing the windings a bit further by some extra layers of inter-layer-tape?
Coupling is usually excellent in toroids, you probably don't need the closeness of windings for efficiency..
/Jakob E.
Remember: distance is your friend here - have you tried physically spacing the windings a bit further by some extra layers of inter-layer-tape?
Coupling is usually excellent in toroids, you probably don't need the closeness of windings for efficiency..
/Jakob E.
DIY Arduiuno based Toroid coil winding Machine - Innovative store
In this post I'll show you how to make DIY Arduino based toroid coil winding machine at home, this machine have userinterface like LCD & rotary knob.
electricdiylab.com
He makes a nice job of it with a simple self contained controler and bread board approach .
Attachments
yes, but having the raw-wire spool inside the ring makes it usable only for relatively large core diameters
Hi friends -
I'm going to be making a whole pile of notes here, kind of mostly for me to talk things out...out loud.
It's kind of long winded so I get it if you'd rather move on. Again. If you have some insight you'd like to share, I would seriously appreciate it. I'm in no rush - I figure I might live another 40 years or so I have time.
Designing valve based microphone amps
I am interested in designing some simple microphone amplifiers using tubes. I have a fairly decent knowledge of guitar and bass amplifier circuits and have been regularly studying, repairing and modifying guitar amps for about 8 years or so. I’m no genius but I can more or less understand how all the components work and generally how stages interact on a very basic level. I feel confident that I could copy a design and layout and wire someone else's design now, but I'm just looking to learn, progress and have fun.
I have been studying designing tube based guitar and bass amps by Merlin Blencowe for quite some time and am fairly familiar with his writings. But between that and Small Signal Audio Design by Douglas Self, I’m at a bit of a loss for where to turn. I have studied NYDave’s one bottle and Mila and checked out some of the Gyraf stuff including the dual micamp as well as looked at some classic circuits including the Pultec MB-1 and have been following some of the other threads here as well like Professor Dan’s. Of course I’m pulling knowledge from forum members PRR, abbey road d enfer, ruff records and Winston OBoogie who tend to dig in on the valve design stuff. Among others of course.
So some of the things I don’t understand (I have zero formal training) I think are pretty basic but I’m not entirely sure of even how to ask them.
For example, I more or less understand that we want reasonably high input impedances and also wish to have output impedances as low as we can really. At least I think those things to be true. There are somethings about creating oscillations that give some real limitations and the fact that we can’t have infinite input impedance and zero output impedance. I don’t think so anyway. I don’t really understand what kind of signal we are going to see at the output of the microphone and how much signal we should be trying to aim for at the output? Are we just looking for +4dBu (3.472 Vpp)? There is only some much head room in whatever is following the preamp be it an external EQ, compressor, desk, tape machine, converters, etc. so we don't need to amplify more than that but we do need enough gain for low output mics and quiet signals. Would all of these devices look for the same (nominal?) level at their inputs? Would most of them have approximately the same input impedance? I am using fairly high end gear at my studio limited now perhaps by my 003 interface which will likely be replaced in the coming 2 years (Probably long before I finish this project)
So I guess starting from the beginning, we are going to have an XLR input, positive voltage on pin 2 with respect to pin 3. That’s a good standard. I think I can figure out getting phantom power there, there are enough reasonable examples of how to do that, I can also figure out getting a pad there, no problem. I could easily figure out how to get a polarity switch here too, but I’m not sure I would do that here or at the output? I’m not sure why you would choose one over the other…? Anyway most of this stuff is covered by the Jensen papers. No big deal. So what’s next - we need to get to the first gain stage.
Here’s where I think I’m really running off the rails.
Okay so let’s say I want to have a transformer input…if I look at one version of NYDave’s One Bottle there is a 150:15K which I think would step up my voltage by a factor of 100 (Is that right? That seems like a massive step up? - Surely I'm missing something there, should I be taking the square root of this?) and reduce the current from by the same factor? On another schematic of that circuit I see a different transformer spec’d which is 200:50K upping my voltage by a factor of 250 (again - this seems like I'm wrong - same as above?) and reducing my current by the same amount? The Gyraf Gyratec IX shows an input transformer with a 1:6.45 ratio which only would step up the voltage by a factor of 6.45 right? These are good right? (Probably, I suspect that these things are well designed) The microphone output would be relatively small and the reduction of current for gain in voltage is good. I want to send a fluctuating voltage signal to the grid. But why such a difference between the Gyratec and the one bottles? Because the one bottle wants to have a similar amount of gain as the Gyratec but with half the gain stages, so we get a boost up front by the transformer? It also gives us a little bonus with the signal to noise ration as well?
ZAJ: Tehát az egyik oka annak, hogy bemeneti transzformátort adjunk hozzá, az az, hogy magának a csőnek lesz némi zaja – azt a zajt, amelyet nem tudunk megváltoztatni, szerintem nem. Növelhetjük a jel feszültségét, de a mikrofon zaját is - de mivel mindkettőt ugyanannyival növeljük, magának a mikrofonnak az S/N aránya ugyanaz marad, és a a csőből származó zaj (ha hozzáadjuk) ennek a zajnak kisebb százalékát teszi ki? Jól értem?
Hozok ide néhány számot az űrből… viseljetek el.
Transzformátor nélküli:
1 mV jel a mikrofonból
1 µV zaj a mikrofonból
5 µV zaj a csőből
Tehát ha a cső erősítése 100. 100 mV jel és 105 µV zaj lenne a kimeneten?
Tehát a csőzaj a teljes zaj 4,67%-a.
transzformátorral (1:6,45)
1 mV jel a mikrofonból transzformátoron keresztül: 6,45 mV
1 µV zaj a mikrofonból a transzformátoron keresztül: 6,45 µV
5 µV zaj a csőből
Tehát ismét 100-as erősítéssel 645 mV jelet és 650 µV zajt kapunk (igen - tudom, hogy ezek a számok durva - de ez elméletben legalább helyes?)
És itt a csőzaj 0,7%?
Még mindig nem értem, hogyan néz ki a mikrofonból érkező jel. Meg tudom nézni egy mikrofon érzékenységét, például az SM7B-emet, amely egy kissé népszerű mikrofonnak tűnik, viszonylag olcsó és némileg alacsony teljesítményű, szigorúan a többi mikrofonommal való összehasonlítás alapján, de ez nem igazán jelent semmit. én még. Nem igazán értem, hogyan kell olvasni vagy értelmezni. Tehát - 1 kHz-en, szakadt áramköri feszültségnél (bármit is jelentsen ez) azt látom, hogy -59 dBV/Pa (1,12 mV) - 1 Pascal = 94 db SPL.
Tehát azt hiszem, ha a mikrofonkapszula 1 pascal nyomást lát (1 kHz-es frekvencián), akkor a mikrofon 1,12 mV-ot vagy -59 dBv-t ad ki? Keverem az RMS-t és a PP-t?
A mikrofon olvasmánya azt írja, hogy a normál beszédhez három hüvelykre a rácstól +60 dB erősítésű mikrofon előerősítőre van szükség, és sok modern mikrofonerősítő csak 40-50 dB erősítést biztosít.
Látod, nagyjából értem, hogyan működik az áramkör DC funkciója. Megértem az előfeszítések típusait, hogyan kell kiolvasni a DC terhelési vonalakat, többé-kevésbé hogyan kell specifikálni az egyes szakaszokhoz tartozó alkatrészeket, sávszélesség szempontjából/nem teljesen (lemezellenállás, katódellenállás/bypass sapka, szakaszközi csatoló kondenzátorok, rácsszivárgás , grid stopper) Kezdetlegesen ismerem a negatív visszacsatolást és annak működését többnyire, de úgy érzem, hiányzik néhány kulcsfontosságú tudás, ami a dolgok kis jeloldalát illeti. Hogyan néznek ki ezek a jelek – hová akarjuk vinni őket? Mennyire kell felerősítenünk ezeket a jeleket. Úgy érzem, hogy D. Self tíz parancsolatának egyike az, hogy ne erősítsd, majd gyengítsd, de lehet, hogy egy másik is az, hogy ne gyengíts, majd erősíts… Szeretném, ha képes lennék megtervezni, hogy megfeleljen ezeknek a kritériumoknak. Tudom, hogy félvezetőkről beszél, de feltételezem, hogy valószínűleg ez a legjobb gyakorlat mindkettőnél? Végső soron minél kevesebb csillapítást csinálunk, annál kevesebb ellenállás van sorba kapcsolva a jellel, és annál kevesebb a Johnson-zaj? Jól értem? Kell egyáltalán aggódnom ezek miatt a zajproblémák miatt, amikor csövekkel dolgozom, vagy maga a csőzaj elnyomja-e a zaj más formái miatti aggodalmakat? Nincs szükségem tiszta, zajtalan előerősítőre. Van egy szép GML 8403-am és néhány másik szép mikrofonerősítőm ehhez.
Nem akarok új terméket gyártani, hogy piacra vigyem. Csak szórakozni szeretnék, és elkészíteni néhány eszközt az otthoni stúdiómban. Ennek a közösségnek és néhány más segítőkész embernek köszönhetően úgy érzem, hogy egyre közelebb kerülök, de nagyon nehezen értem meg, mi az, amit nem tudok.
Hamarosan teszek itt egy újabb bejegyzést néhány konkrétabb tervvel, amin körbejártam.
If you allow me, I would like to give you some advice. I am a retired electrical engineer, my favorite is audio technology. I was a rock band sound engineer, I serviced the sound technology of Hungarian theaters for years, and I also have my own developments. First of all the microphones output: 200 Ohm balanced, and some millivolts to 100 millivolts. The tube noise is very different! For example, it is already difficult to get the original AC701 tube for the Neumann KM-56 microphones. As an alternative, I found a Russian 6S6B-V military tube, with much less noise and distortion, and "eternal life" (According to the catalog, it can withstand 100 G acceleration! It is also excellent for the UM-57 microphone! For condenser microphones, check and replace the 150 - 500 MeaOhm resistors! Almost all of them are defective, noisy! I made the required value myself from 0508 size 47MOhm resistors with a miniature PCB vacuum epoxy insulation. The noise is much lower! The Japanese company KOA produces such smd resistor with type designation HV732ATTD476J. There are 5000 pieces on a reel. Check I use an HP 3456A 6.5-digit desktop multimeter. It is also important to check the leakage resistance of the capsule and capacitors, it is also a strong source of noise. I can check it with a RADIOMETER IM-6 Megohmmeter up to 1000 TeraOhm, the test voltage can be set from 1 to 999 Volts! Yes I also have a lot of experience, for example the Philips PCF201 tube is also an excellent microphone preamplifier, although it has a Dekal (10 pin) socket.
Hello, I also live in Budapest and I am looking for someone with solid knowledge about microphones and servicing them.
How could I get in contact with you?
You should send him a PM (Personaml Message).
Click on his banner and then on "Start conversation".
Thank you, but I cannot do that. I guess as I am new member this options is not available yet.
metalockpick
Well-known member
".. an issue with more droop at the high end with 200 ohms sources on a 1:10 vs a 1:7 or 1:8 once a noble zobel is used".
You figured out that the voltage gain ratio is not the same as voltage gain. Winding techniques, core type and makeup, insulation type, wire type all affect how the transformer will perform in reality. However, (all else equal) all transformers have a degree of loss (expressed as a percentage, though never mentioned in product marketing claims.). The further a transformer gets from 1:1, the greater the loss. Gain ratios 1:2 = +6dB. 1:4 = +12 dB, 1:7 = +16.9dB, 1:8 = 18dB (ideal gain, not including losses at the frequencies and load on the transformer, which are much greater). The losses can be low IF the transformer isn't loaded hard. Do that and your losses increase drastically. Transformers are typically frequency rated when loaded @ 1/10 of the rated impedance. That's why you need headroom in excess of what you think you want, because when you have to use a stepdown transformer to enable you to drive (Pick up to three) Inductive/resistive/capacitive loads, at least you can.
Transformers can also do magical things for the sound in real world situations. And they often sound very different when used in the same circuit with the same test material. There are trade offs.
Input Transformers can be noisy when designed to be many times a high source impedance. Keeping your source impedance down (all else equal) will reduce noise.
You figured out that the voltage gain ratio is not the same as voltage gain. Winding techniques, core type and makeup, insulation type, wire type all affect how the transformer will perform in reality. However, (all else equal) all transformers have a degree of loss (expressed as a percentage, though never mentioned in product marketing claims.). The further a transformer gets from 1:1, the greater the loss. Gain ratios 1:2 = +6dB. 1:4 = +12 dB, 1:7 = +16.9dB, 1:8 = 18dB (ideal gain, not including losses at the frequencies and load on the transformer, which are much greater). The losses can be low IF the transformer isn't loaded hard. Do that and your losses increase drastically. Transformers are typically frequency rated when loaded @ 1/10 of the rated impedance. That's why you need headroom in excess of what you think you want, because when you have to use a stepdown transformer to enable you to drive (Pick up to three) Inductive/resistive/capacitive loads, at least you can.
Transformers can also do magical things for the sound in real world situations. And they often sound very different when used in the same circuit with the same test material. There are trade offs.
Input Transformers can be noisy when designed to be many times a high source impedance. Keeping your source impedance down (all else equal) will reduce noise.
