Quote from: el nino
how long two 9v  battery will last for it?

There are quite a few variables with battery life, but the two main things are how hard you push it for how long, and the overall efficiency of your rig - the main factor being the speakers.  The more efficient your speakers are in converting electrical power into sound the less power you need for a given effect/situation, and the longer your batteries will last.  generally speaking smaller and car speakers are not noted for their efficiency (dB/watt).  As a guide, 80 to 110dB per watt is the range pathetic to brilliant.

I saw a buskers' rig the other day that might give you an idea or two.  It consisted of a small hand truck (aka "L" or "P" trolly) with a small sealed car battery at the bottom, a homebrew speaker box with two 10-inch or 12-inch speakers (at right-angles for dispersion) in the middle, a car radio as the power amp, all topped by a small mixer, with a looper/sampler, and CD player for backing track.  I suspect it also had a built-in battery charger.

As you can see, a lot of what passes for "design" is not much more than sticking established modules together, a bit like Lego bricks.

The core of the FET+BJT gain cell I sketched out has a very high input impedance (meaning it won't load just about any source in front of it) a very low output impedance (meaning it can drive just about anything following it), a gain of about 26dB, and a pass characteristic of a FET, meaning it's a bit triode-like.

About all you have to then take care of is making sure you have DC blocking between stages where required, then it's just the old matters of decoupling the supplies, star grounding, and layout to avoid crosstalk and instability.

You may have to move your "gain" and/or "master" volume controls forward or backward a stage to get the desired operation, say depending on the tonestack you finally settle on and its insertion loss (Baxandall low, Fender &c high).

Where it will get interesting is around the clipping stage of the dirt channel - you may spend more time tweeking this to get the desired result than you spend on the whole of the rest of the preamp.

Keep us posted on how you get on, and what you find as you go.  :tu:

I think the impedances are a non-issue here; the headphone output is likely to be 220 ohms or less (perhaps very much less if it has a buffer and isn't just tapped off the main output) and the input of the Peavey is likely to be no less than a few tens of k ohms and perhaps as high as a meg.

The output level from the headphone socket should be several hundred millivolts, perhaps more, and should be several times the sensitivity of the "deaf" Peavey input; so it has got to be something silly like a mono plug in a stereo socket, or something has died somewhere.

But having done this sort of thing more times that I can poke a stick at, the little amp should drive the Peavey nuts.  I was expecting, if anything, a problem from having too much drive.

{The characteristics of an un-buffered guitar pickup is actually quite interesting;
http://www.ozvalveamps.org/pickups.htm
}

Yeah, well I've seen the Kalamazoo link before.  The main problem with picking up a signal from the speaker circuit, however you do it, is that the speaker also acts as a pretty effective microphone and I gave up using this method when doing band recordings.

A better scratch method is to pick up the guitar signal from the other input socket (which often already has suitable resistive isolation built in).  A better line output is to pick up the signal across the Master volume control.

{In the Kalamazoo article he remarks that connecting directly across a speaker is potentially harmful to the sending amp, but the real danger is actually to the receiving gear by presenting tens of volts to an input designed for perhaps one volt.  He's also concerned about impedance matching which is unimportant in this application - line inputs typically have an input impedance of 47k while line outs are only a few ohms. This is about voltage transfer, not power transfer.}

The Music Electronics Forum advice look pretty much in line with the sort of thing I do.  Generally the speaker acting as a microphone is less of a problem with high power amps.  This is because you are aiming for about 1 volt (at full output) across 600-odd ohms to the desk, and if the amp has an output of 30 or 40 volts at full output then you will be using a resistive attenuator (or step-down isolation transformer, or combination of both) with a ratio of 30 or 40 to 1, and any "microphone" pickup will be reduced by the same ratio.  With a small low powered amp you will use a much lower ratio and background pickup may become a problem.


I should clarify; when I said where the specific sound is coming from "doesn't matter" I was being pragmatic - it's coming out the headphone socket, so that's okay.  It would become a major concern where it was coming from if it was in the speaker but not in the headphone socket (which was a possible outcome).

Unless you happen to use cans a lot, why would you expect a mono amp to have a stereo headphone socket?  As it happens, as a soundie I do use cans a fair bit and have several pairs kicking around, so I'm well aware that they are normally stereo.  A mono plug in a stereo headphone socket would certainly explain a most unexpected result, but simply not fully inserting a mono plug will allow its tip to touch the socket ring connection and things should get loud (and prove that is what the trouble was).

If you look at the right side of his circuit you will see that each of the three fans has a BD140 in series with it and that the current drawn by the LM3914 is from their base circuit, so there is considerable current gain available.

Assuming each fan required 120mA for full speed and the BD140's have an Hfe of 100, then the total base current for full fan speed will be;

3 * 120/100 or 3.6mA,

three segments worth with 2* 5k = 10k between Vref and ground.

The current scaling for the LED drivers is determined by the overall value of the two pots, not their setting.  Their settings determine the bottom and top of the LED range, that is the temperature where fan current starts to flow, and the temperature where it reaches its maximum value.

Iref = Vref/Rref = 1.25/10k = 0.125mA
0.125mA x10 internal scaling factor = 1.25mA/segment

I have a video here by JCM of his adaption in operation but it's 6 megs and too big to attach.

Firstly, in this case the impedance of the source and driven circuits is unimportant as long as the source impedance is much lower than the driven circuit.  In this case the headphone output should have a source impedance very much lower than the input impedance of the following amplifier, and thus be able to provide more than enough signal voltage to drive it, so your result of insufficient signal is most unexpected.

The voltage at the headphone output should be at last equal to the output of a guitar, and more likely 500mV to a volt, several times a guitar output, which should be enough to drive the Peavey crazy, even on its insensitive input.

Did you try it on the more sensitive input?

It doesn't really matter where the dirty tone is coming from as long as it is coming out of the headphone socket.  The problem now is to understand why the headphone socket is so poor at driving the Peavey input.

Does the lead you are using to plug into the headphone socket have a mono plug (no ring)?  If so it is possible the plug is shorting the signal via the ring contact in the socket.  If you pull the plug out about 1cm the tip of the plug should make contact with the ring contact in the socket and you may get a very strong signal.

You should try to measure the AC output voltage from the headphone socket when the amp is driven hard, and you should get at least a fraction of a volt and perhaps as much as a couple of volts.

Option 1; stick with the op-amp, but take "-ve" to ground, "Ground" (op-amp non-inverting input) to a half voltage divider, and +ve to the supply.

Option 2; use the same basic FET+BJT gain cell instead of the op-amp.

Option 1 will give better results because of the much higher available gain.

Here you go @abe.

This stage from the input to the "gain" control, follow up with 3-band Baxandall as shown here;

http://www.ssguitar.com/index.php?topic=2635.0

Except I'd put a blocking cap on the input and output, and bias the op-amp to half the supply so you can use a single supply rail, a TL071 or TL081 would do.

Then I'd follow the Baxandall with another stage like the first, except I'd omit R1 and R2, and the second U1 would become the "Master".

I'd make the "Drive" channel identical but I'd add two clipper diodes in parallel from the wiper of the "Gain" pot to ground, and if possible I'd use two different diodes, or one in one direction and two in series in the other direction. A red and a green LED (or any two LED's with different turn-on voltages) would be worth trying.

In each stage the source resistor R5 might need some tweeking if the idle voltage across R7 is too far away from half the supply.

Quote from: JC Maillet

yeah, I used 5k instead of 500 because I didn't need 12mA driving my
LED's // that's way too much IMO I was aiming for 1.2mA if I recall ...

plus I'm making my output mirror scalable, so I can overcome the 1/10 factor loss and then go beyond if need be ...
my fans were rated for 120mA full power but I got there on less if I recall

~jc

You could be right.  That's JC Maillet's adaption which I haven't built, but since the current drawn by these pots from the ref output sets the output current, and JC had some LED's lighting up, 5k does look too large.  I've written to ask him.

Just a quicky 'coz I'm away from home at the moment.

Applying Ohms Law to your 0.5 ohm resistor readings;

I = E / R, = 0.015/0.5, = 0.03 amp, or 30mA, which is a very respectable idle current for this output stage and tends to make my guess about crossover distortion due to insufficient bias look wrong.  This should be about right.

Your observation that "ghosting" is less with the reverb spingline disconnected suggest that at least some of this effect is due to vibrations from the speaker being directly picked up by the reverb springline, which I'm sure you have discovered is mechanically sensitive.

If that is the/a source then you need to improve the physical isolation of the springline by resilient mounting and by padded enclosure - there really is no other answer since this is much the same as the feedback you get by bringing a mike too close to a PA speaker.

The reason the amp continues to work with the reverb disconnected is because the reverb is in what we call a "side chain".  The amp "dry" or un-effected signal goes straight through, but some of this signal is sampled off and fed to whatever effects ("Fx") are fitted, or via external Fx loop connectors.  The "wet" or effected signal is then brought back and mixed with the dry signal down the signal path a bit, giving a mixture of dry and wet which goes on to the main amp and speaker.

Just an observation.

These days the Baxandall feedback tone control is almost universal in audio reproduction amps, ghetto blasters and so on, but it is not one that is generally favored by guitarists in music creation amps, I suspect because its operation is radically different from most other tone controls, particularly the ones favored by guitarists.

Most guitarists seem to favor a bit of "mid-band scoop" (which sounds like what you are seeking) and this matches up well to the majority of controls that have a "hinge" around some mid band frequency (typically 1kHz for reproduction amps and 440Hz for creation amps such as guitar amps).

These generally operate by introducing an adjustable slope, rising or falling, from the fixed hinge, with the slope set by the control.

The Baxandall doesn't operate like that at all.  It introduces a fixed slope which "slides in" from high and low frequencies, the control setting how far the fixed slope is between the frequency extremes and the mid range, effectively an adjustable hinge.  To get mid band scoop with a 2-band Baxandall generally requires very large values of boost or cut at the frequency extremes, and this is not what guitarists want.

Although it doesn't include Baxandall, if you haven't come across Duncan's Tone Stack Calculator I suggest that you download it and have a play.

Introducing a mid control element may get you close to what you are looking for, but you need to be careful to select a mid frequency of operation where you want your scoop to be centered, that is closer to 440Hz than 1Khz where most designs will be set.

It is not a good idea to operate any cab with one speaker inoperative; never mind the impedance, it behaves as if it has a dirty great hole in it where the un-driven speaker is, which can be most unkind to the remaining speakers.

I've used PVA woodworking glue to repair torn cones, and to "splint" gaps with toilet paper or paper towel.  The aim is to try and restore the cone as much as possible to original condition, and not to add too much mass in one place.

I've never tried pulping the paper first but I'll keep it in mind the next time I have a damage speaker cone to repair, thanks JM.  :tu:

This? (attached)

http://en.audiofanzine.com/guitar-amplifier/avenger/

Whatever you read, you listen to music with your ears, and if this little amp is giving you a sound the bigger one isn't, well that is the truth.

1. Yes, you should be able to connect the headphone output of the Avenger to the Low Gain input on the Peavey without causing damage to either amp.  The first time you will need to inch the controls up on each amp until you find a comfortable level.

2. Generally speaking a solid state amp in overload produces a wide range of harmonics and it is the speaker and its enclosure which act as a filter for this more general excitation; so often with a small amp such as this much of what you are hearing it the speaker and enclosure.  This means that you may not get what you expect out of the headphone socket.  The only way to be sure is to try it.

If you get what you want, fine, but if not you may consider using a mike in front of the Avenger's speaker and using the Peavey to amplify this, although you may need to be careful to avoid audio feedback.

Not a question - not an answer  ; yes both amps can be fitted with sockets that allow preamp out and main amp in.  This would normally be done with a couple of switch sockets inserted somewhere around the final volume control/main amp input, thus creating an effective Fx loop, pre out/main in.

HTH

My step-father got to a point where he decided it was better value for money to buy concert tickets than Hi-Fi gear.  As a kid I was allowed to have any LP I wanted - as long as it was classical.  So in my youth I was playing in rock bands and also a subscriber to Melbourne Symphony Orchestra and Australian Opera seasons, and catching chamber recitals of Bach, Vivaldi, &c.  Never been mad on modern composers, or classical dance as an art form, but a good symphony orchestra in a good venue (and Melbourne is lucky to have both) is really an experience.

I've also done a lot of live recording, but never with the kind of equipment I'd like to have, but I've found that if you are prepared to put in the effort to squeeze the best out of what you've got you can still make some pleasing recordings.

I have a group of favorite recordings that I listen to just for pleasure, and they are almost all live recordings of performances in front of an audience, and even though there are the "ice grinder" moments there is also an energy that you simply don't get with studio recordings; there is no chance of a second take here, get it right first time or die live on stage in front of thousands.  That risk seems to lift performers and performances.

So yeah, when I'm talking fundamental bass I'm thinking of a bunch of orchestral basses, or the biggest Diapason on a big pipe organ - that's the sort of chest-vibrating bottom end that a bass guitarists should have at their disposal.

{You're kidding, right?}


Signal flow goes;

INPUT -----> OUTPUT

By convention forward signal flow is from left to right.

In the "elecfree 100W" (cough cough) amp above the signal flow goes - Input, C1, Q1, Q5; it then splits, the +ve half wave going via driver Q9 and output transistor Q11 and R17 where it recombines with the -ve half of the waveform which has gone ia driver Q8 and output transistor Q10 and R18.  Then the recombined signal goes via L1 to the speaker.  See attach; forward signal flow in red, negative feedback in blue.

Q4 provides the bias offset between Q9 and Q8 to reduce crossover distortion.

Q6 and Q7 are overcurrent protection transistors (which you can ignore for the moment), and Q3 is a load for Q5 which you can also ignore.


One thing you can do is to fire up your amp, and after it has been running for about ten minutes with the volume and gain at zero, measure the voltage across each of the 0.5r resistors.  It will be quite small, in fact it's possible it won't be anything much at all.  What we are hoping for is about 25mV (0.025 on the 2VDC range) across each.  I suspect it will be lower.