Hi, I'm waiting for some coponents wich are missing but as soon I receive it I'll give my impression about this baby :)
I've built 6 jsr 03 at differents voltages and I was impressed by the noise floor so I hope this headphones amp will be on the same quality :-).
I have a complete symmetrical system so perhaps will I do a second one to use them symmetricaly ...
As I can't find the good EI transfo here in belgium, I've choose a torroid one wich'll be separeted by an aluminium case.
(http://www.membres.lycos.fr:80/galinou37/casque_amp.jpg)
Thanks Peranders for your kindness and your answers to all my mail :wink:
Looks very good so far.
Not so good ... one condensator was touching a litle small pad ... and tow pad of a zener weren't soldered ... :-D
I can't give my impression about it because I hear a very small sound behind my preamp ... farnell have sent me 1uF condensators instead of 100 n but when I put wima 100 nF instead of those 1uF the result is the same ... I'm gonna try to find if I've made a mistake whit the coponents placement ... I've had several doubts about it in the amp part of the pcb ...
The DC servo should have 100 nF to create the right speed but nothing stops you have a slower servo. The input cap could be 1 uF the type should be good. What kind of type did you get?
For the moment I've put wima mkp 100 nF just to test. it's the same thing on the two channels a very very little sound ... regulator part is okay, the problem must be in amp part ... I think about a placement mistake ? I'm verifying it because I was having a doubt for two coponents if I remember well ...
If the supply voltage is OK, you should check the DC levels and idle currents. Don't for get to have a input coupling cap and also trim the bias current at the input.
I've tested everything ... :? :-D
I have 13.88 on each amp entry so I think it's ok for the regul.
A 560R resistor was dead on the left channel ... but it's not the problem .... When I test the input bias I see no current at all in this circuit ... ?
R3 is extremely important. This resistor sets the bias for the whole amp. Assume this: Vbe = 0.65 V, Hfe = >10000
All transistors should have 0.6-0.7 V in Vbe, if not something is wrong.
I've tested all the vbe this afternoon and all the resistor, all the voltages points ... I'm gonna do a simulation to help ...thanks for helping ... the problem is on each channel so I don't thing about aparticular prob except a wrong coponent ( farnell has made two mistake in this command ) or a wrong coponent placement ( I've tried to see on your pictures but it's hard ... :-D ).
I've made a simulation and I find the same values ... Tomorrow I'll look with a cd player at the input and a oscilloscope at the output ... very curious ...
With the oscilloscope I've seen immediatly the problem ... the 220pF input condensater for the rf filter was down and all the signal was going to the ground ... :-)that's for the left channel, I'm gonna searche for the right channel but ... ouah ... the sound on my left ear is really good ...
When you are 100% finished you'll notice that this headphone amp is not mainstream.
My english is not perfect at all, I dont understant mainstream.
The right channel has the same problem ... the input rf condensater was out too ... :? I've heard this baby with my AKG k-701 and a the output of my professionnal sound card on the computer ( I've not finished yet the cd pro2 cd player with pcm1794 DAC ).
I don't know how to describe what I hear as I'm a new audiophile but as a professionnal musician working in studio I can say I'm near of what I'm used to ear in studio on the recorder ... it's the best sound I've ever eared in a headphone :-)
Thanks for this baby ... very cool :) I solder since 20 years but I've killed one resistor, two condensator and one transistor ... :D what a shame
I can give my impression of this baby on my french forum now ... :)
Mainstream = average, not particulary cool but not "uncool" either.
I'm rather as you even in my native language. I don't have the nice words for decribing small things in sound impressions. I'll guess we'll agree, the QRV08 is a masterpiece of mine. This is my "objective" opinion and I'm glad you'll think the same. A very important thing is the power supply which certainly not is mainstream, a quadruple 40 MHz ultra low noise super regulator.
What I'm looking for is the neutral sound and this headphones amp is really neutral ... without being chirurgical ... It has its place behind my cd pro2 ...
Indeed it doesn't need a preamp ...
I have to find another problem ...T26 and T26 have 3 V as VBE on the right channel ... I was suspecting it because of little disturb on this channel ...
Check once more the voltages across T26 and also R18. Are you really sure it is a BC860?
Yes, I've verified here's on a schematic ...
it's a 860 and the other a 850
(http://www.membres.lycos.fr/galinou37/Sans-titre-3.jpg)
Do you have adjusted the DC-servo and also the input bias? How much output offset do you have? The whole amp should be fairly symmetrical in voltages. The transistors aren't active if Vbe is less than 500 mV and not more than 1 volt or so.
to adjust the input bias, I have to solder the jumpers on the back side of the pcb ? adjust the DC servo ? how ?
I think I have a prob with a condensator because testing the input bias, I've seen 3.5 V going in 2 secondes to 1 V ( violtage shutting down in my tester ) and I every time i Test it the voltage is 3.5 V
On the other channel the imput bias is 0.7 mV
The voltage across R1 should be 0 mV. This can achieved by adjusting the P1 with either positive or negative current. If you'll adjust this the DC-servo won't have to be working overtime.
THe voltage across the left channel is 0.0004 V but on the right channel it's 0.170 V. When I turn the pot, no change at all ( I've not soldered jumpers behind ).
It's ok now ... two transistors were responsable .... I'll study tomrrow why acting the input bias potentiometers don't change the bias ... for the mioment I listen to music :-D
I've listen a lot of music ... as if I was discovering them ... :-)
The out of the super reg was on the left channel 13.9001 V and on 13.9005 V so it's ok but the out on the right channel was 13.856 V and 13.934 V so I've put a 100 k resistor in // with R55 and now it's 13.938 V and 13.934 V so it's ok too.
Trimming the supply voltage is very unnecessary in a technical point of view. 13.90, 13.93 and 13.85 is extremely good for an untrimmed power supply! The main thing is that the voltage is stable. I think you should concetrate your efforts on adjusting the input bias current and check all working points, verify the symmetry in the design.
I can't adjust the right input bias for the right channel ( I can for for the left channel) .... when I turn the pot, it's immediatly compensed by the opa134 ... I must have a prob with a condensator voltage I see in the right input ( begining from 3.5 V and shuting down in my tester slowly ) ... but for the moment I can't find it ...
You should measure over R1 = you inject current and nothing goes through R1, only AC signal.
You should also check that the opamp not is saturated. The output of the opamp (pin 6) should be max +- 10 volts.
I can easily adjust the input bias testing after C1 and it's ok but testing before C1 ( on the wago for exemple ) I can turn the pot, it doesn't change anything, if I turn quickly, I can see on the oscillo that it grows but quickly go back tothe first value as if the opa was having this on reference and was adjusting before C1. On the left channel I don't have this problem, turning the pot adjust before and after C1 DC bias
Do you really measure over R1?
over R1 it's ok ( it's after C1 )
Do you really have 560 ohms at R2?
yes, I've checked and tested every coponent ... very strange ... But the sound is very cool, the sinewave of the generator is ok at the output without apparent distortion, no pic, no DC offset just input bias on the right channel can't be adjust before C1 for exemple 600 mV behind C1 ( or over R1 or between r10/R11) and before C1 it's always 2.4 mV wich is a good value but not in the good way ... very curious I don't see why for the moment
Have connected the pot both positive and negative voltage? How about the voltage at "wiper" at P1?
I have connected the two pot at the negative voltage ( in fact the left channel don't need it at all so I'll desolder it ) the pot is well doing its job it brings -749 mV ( just after the pot ) to have a 2 mV input bias after C1 ( not easy to adjust with only one rotation, 25 rotation trimmer are more easy :) )
right opa output is -1.2 V
I'm not following you here. You said 600 mV and now it's only 2 mV? I think less than 5-10 mV is perfectly OK as voltage across R1. The rest takes the DC servos care of.
You should be able to see a shift in sign in voltage across R1 when you adjust the pot.
the 600 mV was an exemple it can be 1 V, -1V ... befor C1 it's always 2mV
You mean it's hard to adjust more accurate?
No, It exemples, I've just adjusted it :
0.2 mV after C1 so only - 0.30V at the opa output and always 2 mV before C1 on the wago. the pot bring - 686 mV
It looks good. Nearly 0 mV at the input (= you inject the needed input bias current) and the DC servo is working properly. How much output offset did you get?
The output offset with this adjuste is 8.4 mV.
when I turn the pot the output offset is changing so no prob with it.
with 1.6 V AC signal at input : before C1 2.4 mV DC after C1: 0.2 mV DC out offset : 8.4 mV
without AC signal at input : 3.5 V DC shuting down to 250 mV 0.2 8.4 mV
all those value are +- 5mV moving continuously
Do you use OPA134? You should have a bit lower offset, max 2 mV
when the input bias stay at 0.2 mV the output offset is 1 mA, 8.4 is the max output I've seen. I've find and repared 5 problems in my work on this pcb but this problem is very strange ... If there was a parasitic capacitance on the input like it seems when I test without input signal, turning the pot would change the input bias ... I don't understand at all :? :wink:
Thanks for your help ... sorry to bother you ... I have several opa134 in my stock now but I don't think it comes from it ? it have a link with it because of the static input bias before C1 ( I see quick pick if I turn quickly the pot but it shuts downd very quickly to 2 mV input bias)
If T11 and T12 had exactly the current gain the total bas current would be zero but in real life they differ. This different results in a current which either goes in or out and this current is the one you add or substract with the pot.
If I understand well after T11/T12, the signel goes through T13/T14 then R10/r11 and finish in T19/T20. the T25/26 and T27/28 is the buffer.
I'm not completly familiar with opamp ( thanks for your jung pdf files I'm reading )could you explain me a little bit how it works in this schematic ?
looking a little bit on the schematic helped me to understand about how the opa work ( I think ) it mesures the dc offset between the 2 resistances and compensate / the 0V reference. thesignal is court circuited by the condensator so just the DC is adjust.
If I'm right I can now study my problem knowing all I need ... Thanks for your patience ... :)
I've just listened the QRV-08 with the cd pro2 I've built ... very plaisant, very neutral, I hear for exemple for the first time a real bass sound (as I'm used to ear it in studio)... Thanks for this baby.
I'll put a 2 x 10k log at the input because behind the cd pro2, the sound is high ...
PS : I've not found the parasitic capacitance I have but it doesn't affect the sound so I prefer to finish the pcm1794 DAC ...
Glad to hear it. Someone else listened to the amp and thought the bass was too thin! The thing was that he used my headphones and not his. My HD545 is not very "disco", very neutral and yes not accentuated bass!
I listen with a akg 701.
No bass excess ( I like neutral as i'm a musician ) I hate bass boost :). when I talk about bass it's about the bass instrument and its caharcteristics.( 100hz to 200 hz ) with fingers picking ( 1000/3000 hz) and its caharcteristics like ( fret noises, slides, chord vibration, etc ... ). The voice is very natural too ... I've just listened some of my favorites songs ...
Finally I built QRV-08! But when I checked idle current of output stage I've measured only 3 mV across R20. It gives I=3/2.35=1.28 mA.
I'd check all resistor values by DMM, all is OK.
MicroCap 9 sim also shows around 4 mA of idle current.
Please help me to find a problem.
What is your supply voltage?
Voltages across and values:
R16
R17
R18
R19
R3 = 22k?
Voltage across?
Quote from: peranders on November 02, 2009, 03:49:50 PM
What is your supply voltage?
Voltages across and values:
R16
R17
R18
R19
R3 = 22k?
Voltage across?
Vsup=+13.92 V & -13.9 V
R16=220R V=168 mV
R17=220R V=168 mV
R18=10R V=7.4 mV
R19=10R V= 7.5 mV
R3=22k (measured 19.6k) V=24 V
This seems not to be unnormal but could you also measure the Vbe of the transistors involved?
All seems like OK - all transiskor have 0.61..0.63 Vbe. Only T24 Vbe=0.6 V.
My BJT's is in SOT-323 case from NXP, but I think this is not a root cause.
Those currents you have right now will still get you incredible low distortion but if you want to increase the idle current a bit, lower the value of R16, R17.
Quote from: peranders on November 08, 2009, 11:10:06 AM
Those currents you have right now will still get you incredible low distortion but if you want to increase the idle current a bit, lower the value of R16, R17.
This amp will work on 25 ohms headphones, so I want to increase idle current to have class A operation as much as possible.
I've tried R16(17)=R18(19)=100R, but it gives around 60 mA of idle. This current overheets output BJT's.
You can't think like that I'm afraid. There is no linear relationship with this resistor value and the current in the output transistors. Try to use half of the used value first. (Solder an another resistor on top of the original one.
Hi, i'm about half way through building the first of 2 QRV08, I am following the BOM (as close as possible) for the most part.
a few differences, please let me know if this is OK
I could not get the right size PPS caps from rifa from mouser and although I have some on the way, I decided to build it up with EVOX/RIFA PEN (is this OK?),
I am building 2 of these to use as an output stage of a balanced dac , serving as a pre for my active monitors, is it as simple as feeding +/-L to one and +/-R to the other?, or is it better to feed L/R + to one and L/R- on the other?
I would also like to use LM329 for the reference as its good and I have some on hand. what changes need to be made to the reg gain to accommodate this? I have some hole mounted 431 as well, but would prefer 329.,
I am using some nichicon 4700u for the main reservoirs, should I change the value of the multi-fuses? or is inrush not enough to worry about?
regarding the servo opamps, do the chips used have an effect on the sound signature of the amp? I was thinking I might try OPA827, but I do have the 134 on hand, I just feel if we are talking SQ, I find them a bit boring personally, if they are not influencing the sound quality so much and do the job as best you know how, I will simply stick to them
can the input film cap be omitted if DC offset is at a known low level, fed from a dac inside the same enclosure? and of course CMMR
the instructions say the trim pot is optional, I have installed it, but do I really need it if the servo is doing its job properly?
thanks it for now, sorry for all the Q's, its coming together quite nicely and it really does lok cool too, especially since I have a couple of brands of resistors, so its multicoloured ;)
thanks for your time
Cap: it's not critical as long as it fits proberly.
Balanced mode: Use one pcb to each channel.
Reference: I'll recommend a SMD part but you can of course use a holemounted part but it's a positive thing to have the parts close to each each other.
Fuse: Will probably work alright with the original values. These fuses are rather slow.
Servo: Yes, they are a part of the audio chain for frequencies up to 1 kHz but the part of the audio signal is very low. It's onlý below 10 Hz the have a major impact and at those frequencies even the the old 741 is good.
They should be decent at least. OPA134 is much more than you'll need so you won't gain much to have a really good opamp unless you already have them and don't have to pay for them.
Input cap: You should not exclude unless you have a low impedance connection to your signal source. The trimpot is a must if you have a high impedance DC-path (which you will have if you'll use an input cap) in order to inject base current let the DC-servo work less.
Quote from: peranders on April 09, 2010, 03:58:39 PM
Cap: it's not critical as long as it fits proberly.
thanks for your reply, yep, the ones I bought JUST fit, I only have about 0.5mm of pad on either side, but enough with a bit of persuasion, I just soldered one pad first and fired it up while I slid the cap over the pad, then just flowed solder onto the tip on the other side until surface tension gave way and flowed onto the cap and pad, they are the 7.3 size
QuoteBalanced mode: Use one pcb to each channel.
OK, that makes it easy
QuoteReference: I'll recommend a SMD part but you can of course use a holemounted part but it's a positive thing to have the parts close to each each other.
more important than being a higher quality reference? not sure if 329 is available in SMD, I will check; otherwise I will grab some other SMD ref (suitable). If I do end up using 329 though, can you please advise of the appropriate adjustmnt to the gain of the reg? or just let me know the position and formulae for setting it, that would be fine too
QuoteFuse: Will probably work alright with the original values. These fuses are rather slow.
ah OK, I have other fuses in this build anyway, there is a wanky silver fuse on the inlet :D
QuoteServo: Yes, they are a part of the audio chain for frequencies up to 1 kHz but the part of the audio signal is very low. It's onlý below 10 Hz the have a major impact and at those frequencies even the the old 741 is good.
right, so there IS an incremental improvement, I have the OPA827 here, so will use them; they are great opamps and not being put to any use at the moment, so probably should
QuoteThey should be decent at least. OPA134 is much more than you'll need so you won't gain much to have a really good opamp unless you already have them and don't have to pay for them.
see above
QuoteInput cap: You should not exclude unless you have a low impedance connection to your signal source. The trimpot is a must if you have a high impedance DC-path (which you will have if you'll use an input cap) in order to inject base current let the DC-servo work less.
OK, well that does sound like I could do without both then, the output impedance of the DAC is around 195R per pole and the connection will consist of about 5cm of 16AWG (equivalent) silver foil; this is appropriate yes?
i'm getting quite excited now actually, the DAC its for is coming together well, the hexateq case is gorgeous!, my custom titanium volume knobs and selectors have arrived today and look great (made to my CAD file design) and i'm keen to hear this amp in dual mono balanced mode. I may end up using my own potted transformers and your RFB03 and bypass the onboard rectifier, but for starters will build up as 'standard' there is no pot to account for with the input impedance which is cool and thinking I will wire up the gain jumpers offboard with a switch to choose for headphone use (gain of 2) and unity gain for preamp duties. will make sure to take some pics for the forum, as although I have seen a couple of people mention building 2 up and using them balanced, nobody seems to have gone through with it. fair bit of work to do yet. OH and the LEDs have to be 25mcd yes? or oter alterations need doing?
with a low impedance load of >30R (well actually more like 15R since i'm running balanced) how can I tweak the amp so it never falls out of class A into AB?
any way I cn ft the LM129 on this board? its only 2 pin and I gather is the smd LM329, or can you suggest another upgraded reference perhaps? i'm getting the tweakers itch and i've been quite well behaved on this build, only upgraded the reservoirs, servo opamp and gain setting R's
oh yeah and built a whole second amp to run dual mono ;)
Class A:
I'll leave that up to you. I don't support it. You can calculate how high volume the class A current corresponds to. I think the sound level is way over 130-140 dB in you ears!
The reference:
LM329 doesn't come in SMD at all. You have good references from Analog Devices. You can look there.
I must also point out that the design is a 20 MHz design with a 40 MHz power supply. It can be hard to tweak and I don't support it. I support my tested design. I'll hope you'll understand that.
yes, but its -possible in other designs to have the amplifier operating in class A even when its not at full volume and the write-up says the amp is already class A, but (I understand that this is not always the case depending on the load, and also you forget, this amp will ALWAYS be at full volume, i'm using digital attenuation and there is no pot to speak of. and this is also why low low noise is very important to me thus the thought of finding a lower noise reference, because full volume into 30R headphones could produce noise, I guess maybe i'll cross that bridge when /if I come to it
I was only looking at better references because you mention in your other material (and this is confirmed by my and many others) that LM329 is a much higher quality reference and not a small amount better, but then you say SMD is better. I will probably end up using TL431 anyway, but you stil havent responded as to whether the fact something is SMD makes it THAT much better that it all of a sudden is better than a considerably superior device. if the answer is yes, I will probably use 431, if not I see no reason not to use 329 and keep the leads as short as possible.
I know the design is hard to tweak (and thats a good thing), its an incredibly developed design based on some really incredible lineage, which is why I chose it. changing the reference is not really much of a tweak, but if you dont want to make a recommendation then OK.
the only other changes I was looking to make (aside from dual mono) is omitting the input cap if possible, because no cap is better than a good cap, I have installed the trimpots. and maybe further down the track using RFB03 to feed the DC instead of the onboard rectifiers but I wouldnt expect you to support that one, other than perhaps the build of the RFB03 itself if I had any queries
Class A:
This has nothing to do with volume. It's about the load and how heavy it is. If the class A current is 10 mA, this will mean that the amp is starting to work i class B for output currents (peak values) over 10 mA.
Low noise:
This has nothing to do with the volume. It's determined by the input transistors and the source impedance. A high ohmish pot at - 6dB attenuation will create extra noise. The amp itself has a bit lower PSSR than a normal opamp so it's more important with a good power supply. Noise can seek it's way into the amp but the used reference will have a very low impact on this because of the lowpass filter after the reference. Using a better reference than the LM431 is only required iof you want to feel better. Since you not are going to measure anything many of the parameters of the reference are not important. A plain zener will do but it's nice to use a precision part but not necessary.
http://www.analog.com/en/references/products/index.html
Full volume into you 30 Ohms earphones will probably blow them.....
Sure you could use RFB03 to feed the board but don't forget that short wires and compactness is a virtue when you are dealing with wideband designs. Your long wires may pick up noise instead.
I will not have the volume turned all the way up, it will be attenuated at the digital level in the DAC registers, but the amp does not have any attenuator at all as is the case pretty much any time you use a digital attenuator, so the amp is effectively at full volume, so it is as important for the amp to be low noise as it is the source, as the source effectively takes the place of the pot, so yeah I see what you mean as the situation is the same, but the issue is just moved back a place. lets just hope nobody ever unplugs the amp while i'm listening, the resulting pop would be pretty loud!, lucky I dont have to worry about the connector shorting when its plugged in or removed because its not a phono, all the same I will never be doing it with them in my ears.
oh and it actually will be being measured, this is being used as part of the development phase for a dac, testing different output stages. but that is unimportant to me
I wont deny that it is in part to make myself feel better, as with a number of things I and I dont doubt, you do at times (though i'm thinking not that much after this convo), but its OK, if you dont know of a better part then I will search one out myself after the initial build, I will just go ahead and use the hole mounted 431 or 329 now and see, then when I get the itch i'll change the ref to something else (I have to order SMD ref). I dont believe there isnt a better part as 431 is really a pretty plain vanilla part, maybe perfect for the job, maybe already over what the job needs, but for some reason you avoid answering a pretty straight forward question
is it more important to use an SMD part because its SMD than using a higher quality hole mounted part?
why does being hole mounted make a difference if using a better part does not?
lets put aside for a moment your belief that it wont make much/any difference to what I actually hear and give me a theoretical answer...please!! I have asked 3 times now.
I have hole mounted 329 and 431 right here, right now and it is the only thing stopping this amp from being finished, all the small signal T's, r's and C's are there, the larger transistors are there, the rectifiers are in place, the opamps are in place (except for 2 of the 8 AD825), all that is left of the small parts is the references and then I can go ahead and mount the rest of the large parts, once the right transformers arrive i'll be finished, the only reason I was asking about a better SMD part, is because if i'm going to make a new order for no oter reason but to get a reference and a few opamps, it better be a good part.
whether it makes much difference at this point means very little to me, feeling I have done the best I can do to get the best result in this build is very important to me, as is evidenced with the budget and the fact I have soldered probably 500-600 joints doing most of the work for just the headamp section of this dac, with at least the same again for the dac and reg section.
whether you like it or not PA, you are in a market which, funnily enough, has quite a few audiophiles and even though i'm not totally there, I do have some tendencies that way, so if you sell these extreme audio products of yours to extreme audio technology enthusiasts, you will just have to live with that, I have seen you make comments which make it sound like audiophiles are silly, I dont know if this is intentional or not.....
the amp is always wide open, no pot = no attenuation just the fact i'm using such sensitive headphones is enough to want it to be as low noise as is possible, not only are they low impedance, they are probably one of the most revealing and detailed/fast headphones on the planet. the common mode will take care of most of it too though I guess.
QuoteSure you could use RFB03 to feed the board but don't forget that short wires and compactness is a virtue when you are dealing with wideband designs. Your long wires may pick up noise instead.
cool, yeah agreed, the idea is to make my own PCB to integrate the power lines and sensing for the regs, with a large ground plane for the rest and ideally the amp would connect by pin headers or at worst very short wires.
will let you know how it goes, should flick the switch early next week on the amps
Hi there, after many hundreds of solder joints I have both PCBs stuffed, but before I power one of them up to test, I would like to know if I should be expecting the LEDs to light up or they are oriented so they dont? I have not seen any pictures in here that have them glowing at all, so I wonder what to expect??
thanks
The LED's will glow quite much. Before you solder, make sure where the cathode is. Use a DVM in diode test mode. You will see some light from the LED.
lucky you reminded me to test myself rather than relying on the datasheet. the datasheet reverses the orientation for the diagram for polarity. there are 2 diagrams next to it that are up one way and the 3rd one that shows polarity is flipped upside-down. so desoldered them tested and yes they were wrong (lucky I asked, I have had LEDs with wrong polarity do very nasty things before) so installed the correct way, flipped the switch and 4 pretty red LEDs lit up ;), then a few seconds later the left one went out :(. reflowed all the joints in the left regulator switched it on again and its firing on all cylinders!!!
I have not connected any headphones yet, I want to check the other build first, most likely i flipped the LEDs on that too. then i'll go through the test procedure; expect to hear from me in the next couple of days.
anything specific I must keep in mind when adjusting the pot for the servo? unfortunately I couldnt get any of the model with the pot on the top, I could only get side mounted, so it will be a bit awkward to adjust it
8-)
and another, how do I place the pic directly in the post on this forum??
BTW these pics were taken before I remembered to short R/E56,57 and remove the compensation caps
Uploaded pictures are presented as you'll see but if you want in the test you must host pictures yourself.
hi,
I don't find the set of 10,2 mA in the last stage output as mentionned, in simulation I find 1,8 mA so I've mesured the r20/r22 potential difference and it give 1,8 mA too
Is there something I don't understand ?
That is correct and also according to my measurements. I will change my text.
thanks for your answer as I'm a newbie I study this very good headphone amp and in wich class it works that's why I've found this.