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Messages - Kaz Kylheku

#31
The base of Q3 depends on the output voltage.

It is biased with a voltage divider that originates in the output, and is grounded through the speaker. The resistors are 470K/33K, so the output has to be at around 10V for Q3 to get 0.7V.

There is a feedback mechanism at play in that if, say, Q3 is cut off because the output voltage has dipped to low, then no currrent flows across R11 and R12, and so the whole VBE multiplier is raised toward the positive power rail. In so doing, it turns on Q5 and cuts off Q6.  But if Q5 turns on and Q6 cuts off, the output voltage has swung to the positive power rail also, and so that must turn on Q3.

For Q3 to be turned on at 0.6-0.7 VBE, the output has to be at around 10V, since the voltage divider if 470k:33k.

Perhaps there are episodes of C10 getting charged up, and then forcing the output voltage down so that Q3 is cut off when it shouldn't be.

Bang it up into a schematic and simulate ...
#32
Quote from: belleraphon88 on July 05, 2013, 05:09:04 AM
So here is what i did.
Did the tests with the send into my pc speakers, used 3 different tubes. It didnt change anything.
Next i took the tube out and did the same test, still sounded the same.
So cant be the power amp then or tube

Yes; the tube stage is after the preamp and effect send, so it makes no difference.

This is lucky for you because you don't have to mess around in the power amp where you have higher voltages (especially given the tube).

The thing to do now is to make yourself some kind of adapter so that you can use those same PC speakers as an audio probe. A simple setup would be to have a cable which has an alligator clip for ground, and a multimeter probe for signal.  Clip the ground somewhere to a ground point in the amp, and then use the probe to pick up the audio from various sections of the preamp based on following the schematic.  You can find out the first stage where it is happening. It would really help you if you reproduce the issue with some signal source other than your guitar, like using your PC's audio line out to play some continuous tones into the amp's input.

You said earlier that the issue occurs whether or not you use the gain channel; that you can hear it on a distorted tone that is produced by a pedal going into the clean channel. Assuming this is really the case, the problem is really narrowed down.  There is very little circuitry between the input and the effect send that is common to both channels. What is common to both channels is the IC1A op-amp input stage, and and the channel switching and muting FETs Q4, Q5 and Q6. (Well, Q4 and Q5 are specific to their respective channels, but the circuits could interact in some way since channel switching is supposed to be mutually exclusive).  I can't find the logic that drives the channel switching in that schematic, by the way.

If the voltage at the base of Q6 oscillates for some reason, then you will get volume fluctuations or periodic cutouts.  It's not obvious why that would be the case, since the Q6 gate is driven by a voltage divider that is straight off the power supply. Still, if you had an oscilloscope you could easily put a probe on the bases of these transistors to see what the voltage is doing.  If there is such a thing happening, it is probably a little too fast to register on a multimeter, but a scope trace will react instantly.

There is the possibility that it's an issue with the preamp power supply: that basically the entire preamp cuts out due to a periodic voltage brown-out. Your tech would almost certainly have picked up on such a glaring thing!
#33
So, it goes away after a tube swap and then comes back? That doesn't mean that it's the newness of the tube that fixes it, but simply that the tube socket was exercised.  Try confirming your theory by pulling the tube and popping it back in a few times. (Power off, of course; I don't mean live).

Maybe the tube socket needs cleaning. (I use a toothpick with a very small amount of toothpaste to polish inside tube sockets; then another toothpick and lots of 99% pure isopropyl alcohol to clean up after the toothpaste.)

Maybe the tube socket or something else nearby has a cracked solder joint.  Inspect all solder joints and repair if necessary.

Clean all jacks and potentiometers.  Pots can make weird volume changes.

Check all hookup wires, like to the speakers.

You can test whether this problem is related to vibration by taking the speaker out of the amplifier (while keeping it connected) so that you can play test notes without rattling the enclosure.

You said:

Quote
It does have a effects loop. I dont have a external cab to test it with, is there any way i can test it.

You really have to do this test! Plug the effect send into a pair of computer speakers. Get an adapter plug for this: quarter inch plug, with 1/8" socket.

Knowing whether the problem is before the power amp or not is an important piece of information: divide (the signal chain) and conquer.
#34
Unfortunately, the PDF doesn't give the schematics for the entire amp, just back end sections.

It sounds like the signal is experiencing short volume dips or cutouts, which repeat at a low frequency rate of several times per second.

This suggests itself to be a motorboating oscillation in some circuit that affects the bias of a stage.

There is no outright reason to suspect that it is in the power amplifier. Even if it happens in every channel (which I suspect is not the case from a comment given), there are other common circuits that the signal goes through besides the power amp.

Does the amp have a pre-amp out jack (or effects loop send) and can you hear the artifact in that output? If so, that tends to eliminate the power amp.

Does the artifact occur only in one channel? That tends to blame that channel.

I don't think you're hitting the tube hard just because you turn up the gain on the gain channel a little bit. What can hit the tubes hard is volume. The tube is just the differential summing stage for the amplifier's feedback, generating the error signal that is amplified.   FET TR7 provides the voltage gain after that and drives the output stage.  The gain of the amplifier is fixed, so the tube stage has to have reasonable clean headroom to pass through signal when the amplifier is played at the maximum volume.  I.e. it is overall volume that will drive the tube, not gain on the gain channel.





#35
I have entered the RA-100 schematic into LTSpice and am getting meaningful, "life-like" simulations out of this thing.

I'm lacking models for the four Sanken transistors used in the Sziklai pair driver stages; for now I simply substituted 2N2904's and 2N2906's.  xP

I had to substitute a Zener diode for that thermal transistor whose base is left floating. That just does not simulate. Thermal shutdown can be simulated by varying the voltage on that manually. If you replace the Zener by either a short or open, the circuit will mute the input stage, so it needs an in-between voltage to work properly.

I also had to tweak a resistor value in the VBE multiplier to get a reasonable quiescent current, due to the substituted output devices.

Well, the thing is amplifying.  The JFET mute on power up reproduces in simulation (I shrunk a resistor for a shorter time constant than in real life)  The foldback current limiting behavior also reproduces, complete with clip LED being activated.

This is not just idle curiosity; armed with this, I'm going to design a better current feedback circuit that doesn't cause a DC offset like the one I'm using now.
#36
Thanks for posting the schematic, Roly.

This was not obtainable on the net just half a year ago!

By the way, my earlier comment that I was surprised by the quiescent current was the result of a brain fart. About 5.5 mV across each 0.22 ohm resistor was in fact exactly what I had set it to in both channels. It's my preferred setting. So no mysterious drift had taken place.

Now I still have no explanation for the bad done which prompted all this. I believe it is a problem in my ears/brain. Sometimes I have episodes during which certain sounds bother me. Not only my guitar tone, but tones heard on records that are otherwise fine.  What was smooth and violin-like becomes irritating. It is a major frustration.
#37
I've been driven crazy with bad done over the past week or so.  There has been a bout of wet weather so I blamed the humidity. I ran hot air against the speakers in the 4x12 cab, and put a bunch of silica gel dessicant inside the thing.  I cleaned jacks. I cleaned tube sockets in the ADA MP-1. I repositioned the cab in the room. But  I did the obvious last: try switching to the other channel on the power amp. Aha!

What is the perceived problem? Something harsh and objectionable in the tone in distorted programs, and a stiff lack of touch sensitivity on cleans, which are much more fluid and silky through the other channel.

This morning I cracked open the power amp, and checked the quiescent current in that channel based on the voltage on one of the 0R22 emitter resistors (with the amp completely unloaded: no speaker or resistor). It was around 27 mA. What? I had set it to 5 mA only weeks ago. How can it drift so far, and in such a short time? This is a sign of some trouble. I also noticed some quite marked instability. On power up, the voltage was 6.1 mV on the resistor, soon dropping to around 5.5 mV in less than a minute.  Change with temperature is expected, but I don't remember it jumping by such a percentage in a short time.

I haven't had time to do any more investigating than that, so far.

Maybe a driver transistor is going south?

There is no glaring problem with the amp, like pronounced distortion. It reproduces sound and plays loud.

I'm not going to have time in the next few days, but I will run some signal through it into a load resistor and scope it.

It's nice to have a second channel for backup and comparison.
#38
There is a compression technique for lower bitrate AAC whereby the high end is basically stripped out, and all that is retained about it is the shape of its frequency profile.  (That and some info about transients and whatnot.)

When the audio is decoded, the lower order harmonics are somehow extrapolated into the high end, which is shaped by the frequency profile. Voila, the high end is back, faked out!

Turns out we don't hear the high end in detail; but we just need it there so that the lower end has definition and "air".

I'm wondering whether this might not actually end up sounding better than the original high end, since it could conceivably have the side effect of cutting away some objectionable harshness, and replacing it with a smooth, calculated high end. If so, it could be a useful post-distortion filter for a guitar signal chain, to smoothen distortion.

It would be after the EQ, so that the high end would follow the contour.

Hmm ...
#39
Tubes and Hybrids / Re: Low Voltage Tube Preamp
June 24, 2013, 09:07:22 PM
Quote from: jaylow on June 03, 2013, 03:56:39 PM
I know that the 12U7 is the preamp section. I'm just wondering if it ends before or after R7 and R9.

R7 and R9 are part of the input stage of the power amp tube. The reason is that they bias the grid.  The signal is superimposed onto the bias via the C2 capacitor.

The division between the stages runs down the middle of the capacitor, so to speak. If you split the amplifier into separate devices, then C2 ends up being cloned: the preamp will end with an output coupler, and the power amp will have an input coupler.

Without the output coupler the preamp will produce a signal with a big DC offset in it.   

Without the input coupler, the power amp's grid bias will be disturbed by DC offsets in the input.

If things are split into separate devices, you may also want some DC return shunt resistors to ground, outside of the capacitors on both sides (relative to looking into the circuit).-

If a device with a coupling capacitor is connected to another device, and that other device produces a bias current, and there is no return path for the bias current, then that current goes into the capacitor. When as capacitor charges, the current slows down and stops. When you block the other device's bias current, it will probably stop working. This could happen in either direction.  If you know that the other device doesn't have a bias current, you can omit the DC return resistor.

You can even omit a coupling cap from one device if you're sure that the other device will always provide one.

Definitely, R4 is part of the preamp. It's a necessary V2 anode load resistor which develops the output voltage as the current across it fluctuates.

I hope I haven't over-complicated things and that you now know better how to find the division between stages.

#40
Although circuits from that website may be rumored to be unreliable, this particular one is closely based on what is in the datasheet for that TDA chip.

Basically the only contribution of this circuit is the use of the switching jacks.

They also changed the input resistor from 5K (datasheet) to 22K.  This is a feeble attempt to provide a higher input impedance needed for guitar. Unfortunately, it is nowhere high enough; you need a minimum of 500K.

The raw input impedance of the chip is specified 100K, so it's best to try to preserve most of that. You need to provide a DC return resistor for any bias current flowing out of the input; perhaps a 500K resistor to ground for R1 could do the job. It's worth experimenting with, anyway.

Forget the verboard layout; make your own and carefully ensure that it's correct.

There is some Windows cad program for doing layouts on veroboard. It's called VeeCad.  I've never used it; but if strip boards are your circuit building too of choice, it behooves you to check this program out!

"Like a PCB layout program, VeeCAD takes a schematic generated netlist and assists you to build a layout - on strip (Veroboard), matrix, donut, tripad, Atarado Smart, and other prototyping boards. "

Sounds good.

#41
I have an FX-500 that I bought new in 1989! 24 years ago, man.  8|

I never took much of an interest in the internals.

A few days ago, I ripped out the six RC4558's that make up most of its analog audio path, and put in NE5532's.

Quite an improvement; it doesn't sound so cheap and boxy any more. I mean there are limitations on the digital side, but at least now they are laid bare.

The reverb is smoother, less irritating and less grainy sounding. Now when I heap on lots of reverb (externally, via analog mixing), it does not seem to occlude the dry signal as much as it did before. 

Yet the basic character of the unit is retained; it still sounds like itself, just better.  Clearly, its "personality" is from the digital side; and the analog side is just another op-amp selection cluster-funk.

It's a nice board to work with. The component side pads are solder-masked and so the solder did not capillarize through to the other side to make a double joint, as can happen with plated through-holes.  The chips gave way quite easily after just pumping solder away; they just needed a little twist.  No component side solder manipulation (pumping, wicking) was needed at all.  The holes came out mostly clean. A .028" drill bit went through most of them easily. And so all six were done in under an hour.

Now it's obvious why the headphone output was better at driving a power amp (less tone suck): it's an NJM4556 which can drive 150 ohm loads. I kept that one in there.

I didn't consider if the power supply has the margin for the bigger current draw times six op-amps. I didn't check the temperature of the 7809 regulator, or take any measurements, like ripple on the input side, taking instead the empirical route of: close it and stick it back into the rack and if it works it works.  xP
#42
Quote from: cablebling on May 19, 2013, 12:46:33 PM
All working fine now - the problem was the Latch IC, U8   :)

That will mean something once you post the schems. ;)
#43
Amplifier Discussion / Re: Debugging a new amp!
June 08, 2013, 01:25:23 AM
I recommend looking at the schematics for a vintage Roland Cube 60.  It has JFET circuits like this in the input stages, including a similar tone stack.

Speaking of which, notice how it uses a 1M pot after the tone stack which doesn't load it down as much.

And check out the proper references of the JFET gates to a voltage. For instance near Q1, there is a voltage divider R2/R3 which cuts the 38V to about 3.5V. This is conveyed to the JFET gate through R4.  This 3.5V voltage is written down next to the gate.

You need to similarly a voltage at each gate, and then superimpose your signal via AC coupling onto that voltage. The JFET is not properly biased if you leave the gate floating (every path out of it leads to a capacitor).
#44
First of all, about wattage. If you look at the ambient-temp versus dissipation graph, it gives 850 as the maximum without a heatsink, which derates above 25 degrees at 6.75 mW per degree celsius. If you want anything more than 850 mW out of these chips, you must stick a heatsink on them.

The thing is also quite sensitive for the amount of headroom. An input well below 100 mV drives to saturation with THD above 10% (grossly audible). See the Po, THD - Vin graph on  P. 6. Basically, you can easily clip this thing out by directly plugging a guitar pickup into it, and one that isn't even hot. Something like a DiMarzio PAF will put out 300 mV, supposedly.

It's not clear to what extent if at all this lack of headroom will improve with a heat sink.

If all you want is to amplify the output of a preamp into a small wattage, you basically do not need any voltage gain beyond what the preamp is capable of. You just need an output stage that can drive the speaker (e.g. class B push-pull emitter follower, with a buffer in front of it so the preamp doesn't have to deal with the low-ish impedance of that stage).

About the noise that you get when you connect the circuits together, from the description it sounds as if it might be motorboating: a positive feedback loop through the shared power supply. Motorboating is a common plague of single-supply circuits with multiple AC-coupled stages. The circuits have poor power supply rejection, and so parasitic feedback loops form through the power supply connections. These AC couplings introduce phase shift at their cutoff frequencies, and so you can end up with a low frequency phase shift oscillator.

I see the that the Fetzer is just a JFET based common-source stage. But it is an inverting stage. So if there is feedback into its input from the output currents of the amp, it will be inverted, and so 180 degrees of phase shift is enough, which could be achieved by just two capacitive couplings (90 degrees each).

Can you post the complete circuit diagram: Fetzer plus your AN7511 amp?
#45
Hi MickMad,

Have a look at the JFET input stage of the ADA MP-1. It's a neat circuit which places an N-channel JFET with a PNP transistor in a kind of hybrid complementary pair, which is then used as a source follower (unity gain: it's not a gain stage!)  The JFET provides the high input impedance, and the PNP delivers the current, allowing for a fuller voltage swing. The circuit runs on dual 15/15 rails.

I'm mentioning it because of the LTSpice appearance in this thread, and that circuit happens to be something I've been simulating in LTSpice recently. (The transistor part values are not the real ones.)

And, hey, use good parts in this USB interface! You will never beat the M-Audio or Behringer and whatever *s!!t* if you do what they do: use *s!!t* parts. 

I found JRC4558's in the preamp section when I opened a Tascam US-122L audio interface. Ugh!

Also read this document: http://www.rane.com/note151.html    And google "pin 1 problem".