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Messages - Kaz Kylheku

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THEN (lately) I started missing that kind of hump that comes with palm muting and stuff like that, so I first started messing with pedals (I've built a lot of them in years) and eqs but, I found that adding let's say, a peak around 100/150hz (where speakers usually have their inpedance peak) sounds close BUT then everything gets muddy and lifeless..especially when I hit my deepest distortion, it gets confused...not that kind of "breathe" that you can have with tube amps..

I get that lifeless sound with poor "hump" when I turn down the current feedback on my amp, and the "breathe" when I crank up the current feedback. It's a huge difference.

The Newcomer's Forum / Re: How to determine what's what on a pot?
« on: September 27, 2013, 09:25:51 PM »
Question: How do I determine what wire goes where, desoldering from a Side by Side by Side pot onto a pot as seen above, where the tabs are arranged on the bottom like that?

So you have a pot with three terminals, and you don't know which two are for the main resistive element (and in what orientation) and which are for the wiper?

Assuming the pot is functioning, what you can do is:
  • turn the pot all the way left or right and measure all three possible resistances
  • one of the resistances will be nearly zero, and that means one of the two lugs that have close to zero ohms between them  (let's call these two T1 and T2) is the wiper, and the other one is of course the terminal which contacts the wiper when the pot is turned to that position
  • now turn the pot completely to the opposite side and check the resistance from T1 to the remaining terminal T3, and from T2 to T3. If T1 is now zero to T3, then T1 is the wiper. Otherwise T2 must be zero to T3 and it is the wiper.

Then, for future reference, take a fine felt tip pen and write these letters somewhere on the pot: L, W, R: L close to the terminal that shorts to the wiper when the pot is all the way left, W close to the wiper terminal, and R close to the terminal that shorts to the wiper when the pot is all the way right.

Don't overlook the possibility that what is loading the regulator down could be ... the regulator itself, due to being toast, with an internal short.

another update: I have re-soldered all of the joints on the bottom of the PCB, no difference what-so-ever. No better, no worse :(

The hum may still be there, but the component that previously made a noise when tapped with a chopstick doesn't do that any more, right? If so, that is progress. And if it weren't for the hum, you might not have been alerted to this problem.

The Newcomer's Forum / Re: Increase low end by modifying the circuit
« on: September 09, 2013, 01:29:42 AM »
Now, a slightly different topic.

Part of the reason you might not be "feeling" enough "boom" from the amplifier is that it's a voltage amp.

You can add some current feedback to its feedback circuit: that is to say, make a fraction of the feedback voltage be proportional to the current that is flowing through the speaker.

Adding a modicum of current feedback will raise the output impedance of the amplifier and allow the speaker to resonate more at its resonant frequency. As well, current feedback will create a natural "mid scooped" EQ curve, using nothing but the speaker's own impedance characteristics. A lot of solid state amplifiers in guitar amp history have used current feedback: Roland JC-120's prior to 1982, Peavey Bandits from the 1980's, Marshall Valvestates, Crates like the Flexwave, ...

What current feedback does is ultimately up to the speaker. If your speaker likes to resonate at 80 Hz, you will get a thump there, et cetera. Exactly how it sounds cannot be easily described in words. It's like an EQ change, but not exactly. The tone becomes more "vibey" and "three dimensional" with a nice "chime" on top. Perhaps a custom circuit could be built which does it without current feedback, but it would have to be quite resonant compared to a run of the mill tone control, and it would have to be adjusted differently for each loudspeaker (or else it would be "fake": for instance it would try to put a 70 Hz peak on a speaker that wants it at 85 Hz.)

When adding current feedback, you have to be concerned about not upsetting the DC offset. You see those R14 and R16 resistors in the schematic? They are identical for a reason: similar bias currents flow from the - and + inputs of the amplifier, and generate similar offset voltages across these resistors. The idea is that these offset voltages at the inputs cancel each other out, which helps minimize the DC offset (along with that C16 cap which further minimizes the DC offset by reducing the gain to unity at DC, so the DC offset at the output is basically a copy of the offset at the inputs, not magnified by any gain).

Current feedback involves using a small-valued resistor (like 0.5 ohms) in series with the speaker for sensing the current and mixing that into the feedback signal. But this creates a low impedance which will upset the balance of those two resistors, and could cause a significant DC offset. For that reason, current feedback is usually AC-coupled through a capacitor. You see that time and time again in guitar amp schematics.

(If you can address the DC offset issue in the amp with, say, a DC servo, then you can DC-couple this current feedback signal, eliminating the blocking capacitor.)

The Newcomer's Forum / Re: Increase low end by modifying the circuit
« on: September 09, 2013, 01:04:41 AM »
In addition to C1, another cap to look at is C16.

(Of course, at this point I should tip my hat to the other comments: any check you write with the amp has to be cashed by the speaker!)

This capacitor is used to set up the feedback loop around the power amp such that the amp has no gain at DC.  This is done so that the amplifier has no DC offset (places no DC voltage across the speaker), but it rolls off some low frequency response.

It is possible to null the DC with other techniques and eliminate the capacitors.

In my power amp, I have successfully added a bias current cancellation circuit. It simply generates a similar, but opposite bias current to what the diff amp transistors put out (and with similar temperature dependent behavior). This is not really practical to do for an op-amp, because you cannot match the device which is inside it. (Well, not strictly true! You could get another TDA amplifier chip and power it, then mirror the current ...).  Anyway, in that amp, I shorted out the equivalent of your amp's C16, and nothing bad happened: the DC offset stayed about where it was! Except, the amp suddenly had gain all the way down to DC.

A more popular way to null DC is to build a "servo": a circuit which monitors the bias point and using feedback, it applies a corrective signal at the amp's input to null the offset. A DC servo uses capacitance (it's usually based around an integrator op-amp circuit). However, because it's an active device with gain, with the capacitor in the feedback position, it can effectively simulate having a huge capacitor.  The servo can have a time constant of seconds if you like. It would not be practical to replace C16 with a capacitors large enough to have a time constant in that range. With a servo, the amp doesn't have gain down to DC, but almost.

Amplifier Discussion / Re: Preamp suggestions needed!
« on: August 30, 2013, 05:31:04 PM »

I have just got around to modeling this in LTSpice and I can't see any obvious reason why it wouldn't work with just about any commonly available JFET.  It's fairly conventional but uses a couple of current sources, Q2 and Q4, to obtain Maximum Available Gain (MAG) from Q1 and Q3.

I was looking at this yesterday and also came to the conclusion that it is not cascoded stages; but a simple case of active loading for more gain.

The circuit description claims that it is cascoding, but in cascoding, the "upper" half of the stage is a common base/gate/grid circuit, and so there is a load resistor above where the output is taken, not like here, where it is between the transistors.

Maybe an interesting way to modify this circuit might be to actually convert one or both of the stages to an actual cascoding topology.

Could you post the LTSpice files somewhere?

Amplifier Discussion / Re: Marshall 5210 gain squeal question...
« on: August 27, 2013, 06:52:40 PM »
Everything points at the squeal being controlled by the source impedance at the input. When you turn down the guitar's volume, there is less impedance. Neck pickups typically have less impedance (e.g. 8K versus 16K, when we're talking typical matched-pair humbuckers, wired in series). Output of effect pedal: low-ish impedance, though sometimes they have some passive components in the output so that it is not zero.  Output of active pickups should be close zero impedance since it's just the output of an op-amp.

Bottom line, reducing the impedance at the input kills the squeal. This strongly suggests that the input stage is part of a feedback loop, and the conditions for oscillations are broken when the input is grounded, or put closely enough to ground so that the gain around the feedback loop drops.

You know, is it possible that someone in the past tried to service the amp and replaced the input jack with the wrong one?

Amplifier Discussion / Re: Marshall 5210 gain squeal question...
« on: August 23, 2013, 05:21:05 PM »
Another Marshall trick was the input jack mute.  Look at the input jack, on the sleeve contact there is a cutout.  the factory drawing shows it going to the FX send jack.  When the jack is empty, presumably it grounds off the FX send.  The flip side of that coin is that it is a long trace from the output of the preamp right back to the input, and can act like an antenna.   I have stabilized such amps before by cutting that trace over at the FX send end.

That feature of the schematic is the most striking to me. I do not understand the jack hookup that is going on there, but if I squint my eyes, it looks like a positive feedback loop waiting to happen.

Feedback over several op-amp stages can easily create a situation whereby enough phase shift accumulates to create a phase shift oscillator.   And oscillators are, of course, sensitive to gain: they need just the right amount of it. (Or in practical terms, enough of it: the surplus gets clipped.)

I second Enzo's idea of disconnecting this bullshit hookup from the jack.

If it doesn't solve the problem, you can reconnect it (if you're so inclined), so it is worth a shot.

Amplifier Discussion / Re: G-K RB400 On w/no sound
« on: August 16, 2013, 09:42:33 AM »
I got some syringes from a drug store and now I'm injecting oil into pots. Very good results.

I measured a slightly lower wiper resistance in one test unit. It went from 1.5 ohms to 1.0.

The one in my guitar is absolutely silent at high gain, and feels almost strangely smooth, like you're turning a hot glass rod stuck into paraffin wax or something.

Little by little I will work my way through all the pots, including the 31 sliders on the graphic.

Amplifier Discussion / Re: Serious Blocking Distortion
« on: August 15, 2013, 04:24:33 PM »
Haha Roly :P

Ok i've done some reading and am i right in saying the Phase Shift is basically a change in the audio signal? Like a delay perhaps? And that its for mixing two signals together? Which for a guitar would be the original note that was picked and then the harmonics that come after?

Every sinusoidal has phase. Frequency and amplitude are not enough to completely specify a sine wave. You have to know the phase. At a given point in time, a sine wave could start on a crest, or in a trough, or anywhere in between.

A phase shift is not exactly a time delay. Whereas, of course, a time delay will shift the phase, circuit components like amplifiers and filters do not introduce delays (at least not delays that matter in audio): the signals move at close to the speed of light through the circuit.   An amplifier's or filter's phase shift is a frequency-dependent response.  It is not caused by a time delay, but by a response lag (or its opposite, "eagerness", or lead).

For instance, a low-pass RC filter has lag.  Lag does not mean that there is a time delay. When we send, say, a step signal into the circuit's input, the output begins to react instantly, but it does not change instantly: it moves slowly, because the capacitor has to charge through the resistor to achieve the new voltage. It cannot follow the step.  This lag is why the circuit is a low-pass filter; higher frequencies shake back and forth more quickly, and so the lag suppresses them more than it suppresses slow oscillations.

If we look at your graph, we see that you have a positive phase shift over most of the range, which goes hand in hand with the high-pass-filtering that is going on.  In a high pass filter, the phase shift goes the other way: it leads the original signal. This is because (in the case of a simple RC high pass filter) the output is proportional to the current passing through the capacitor. Suppose we send a step signal into a such an RC filter. The coupling capacitor begins to charge instantly, and so there is an immediate, fast voltage spike on the shunt resistor. The spike then dies down as the current diminishes and ceases. The eagerness of the high pass filter to respond causes it to produce a leading phase shift over sinusoidal signals.

Phase shifts are important in audio, and at the same time they are unimportant.  If you have a stereo (or surround) system, and the signal is inverted in one of the channels, this is bad; it means that every frequency is shifted by 180 degrees. When you sit in the ideal listening position, you will get strange sounding cancelations and comb filtering.   Phase shifts affect bass.  A 90 degree phase shift in a 30 Hz signal represents a time shift. It can throw off  a kick drum in a perceptible way.

Phase shifts are not audible in the higher frequencies (except allegedly in some contrived listening tests with specially crafted material).

All filters have some kind of phase response. For example, the narrow filters in a 31 band graphic equalizer seriously affect the phase coherence of material. Yet, audio engineers and musicians use them anyway.

Phase is the subject of a lot of audiophoolery. Some claim they can hear several degrees of phase shift in a 10,000 Hz signal, and the like, which is nonsense. Snake oil devices exist which supposedly reconstruct the phase coherence of a signal.

Phase is crucially important in the stability of amplifiers which use negative feedback.  All amplifiers have a limited frequency response, and due to multiple parasitic filter poles in their stages, at some frequency they accumulate a 180 degree phase shift. Negative feedback is also 180 degrees, and so this phase shift turns negative feedback into positive feedback.  If the amplifier has a gain of 1 (or, practically speaking, at least one) at this frequency, then it will oscillate.

Phase can tell you things about the response of your circuit. For instance if we look at your graph, the amplitude response looks fairly flat toward the right. Only if we look a little closer do we see that it actually is curved: it reaches a maximum and starts to fall. So there is a frequency response peak hiding there. Where is that peak? The phase response provides a clue: right around the frequency response peak, the phase response drops down to zero degrees, and crosses into the negative (lead turns to lag). That's about the place where high pass switches to low pass.  It is clearer in the phase shift plot; the phase shift clarifies and confirms something to us.

Amplifier Discussion / Re: convert Line 6 guitar amp to battery powered
« on: August 14, 2013, 07:02:34 PM »
Landlord says you have to use phones? Really, so nobody can watch TV in your building or listen to music? Or play video games? Or have conversation with friends? Or vacuum the place?

Why don't you play at a level that is below 70 dB, and quieter than that after hours?

Converting to batteries to play outside is not the answer!  Where do you live, and is the weather/climate always good?

What if you get inspired in the middle of a hurricane, or snow blizzard, or heat wave?

Here is an idea: tilt or raise your cabinet so that it's pointed at your head. Don't sit or stand next to the amp so that it's blowing sound past your ankles. You can knock numerous decibels off if you do this, and enjoy a raw, close-miced-like tone at the same time with less room reflection.

Make sure you have plenty of furniture and coverings in the room for sound absorption.

Watch for too much bass in your tone. Those palm-muted, low-end "brootalz" will get through walls.

By the way, the wattage of this amp means nothing, unless you're cranking it. A 5W amp can get you in trouble in a rental building, depending on how you use it.  A low wattage amp is not the answer: the answer is, turn the damn thing down and respect quiet hours. You only need a different amp if the 150W unit sounds like crap at low volume. (Which, in practical terms, would mean it's a tube amp, so that point is moot. :lmao:)

I took out the main drive power transistors and need to know does the dot on the board denote base or emitter

Use the datasheet, Luke!

How did they design the board? They had a part in mind, and made a footprint by following its datasheet.

If the board designers had any brains, the dot would go to whatever is "pin 1" in the part's datasheet. But it's best not to rely on such markings.

Is this what you have?

Pin 1 is base; Pin 2 emitter; metal case is collector.

Amplifier Discussion / Re: Serious Blocking Distortion
« on: August 14, 2013, 04:58:54 PM »
On LTSpice when I run a simulation using the AC Analysis, I get these two lines. The solid line I believe shows how the volume increases with the frequency, however I cannot work out what the dotted line is?

Note how the vertical units on the far right of the LTSpice plot are degrees.

The dotted line gives the phase shift.   Frequency response is a complex number: it is two dimensional. Not only does it have amplitude, but it has phase.

Signal processing devices can change the amplitude in a frequency dependent way, but also shift the phase in a frequency dependent way, and the two are inter-related.

So frequency response has two dimensions, and this is why the output of the Fourier transform is in the two-dimensional doman of the complex numbers (even though the input is a real-valued time-domain signal).

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