[Research]

 

This page is really just for my benefit, so I can remember what I learned and where I learned it from. If you do find yourself on this page Please follow the links to the original site and material. These are just snippets used as reminders for me.

 

 

 

 

Stuff worth knowing [][][]

  • The Cap and resistor in the gain loop (to Ground) in a (non inverting) op amp - from a low pass filter
  • Adding 2 clipping diodes in series raises (about doubles) the clipping threshold (will have to be louder before clipping )
  • Low Pass - (Resistor / cap to ground) Passes Lows and Attenuates highs
  • High pass - (Cap / Resistor to ground) Passe Highs and Attenuates Lows
  1. Jfets - Clip softer than BJT's but have less gain


If you don't have the right values

Resistors in series = Both resistors added together

Capacitors in parallel = approx the values added together

 

 

Caps block certain frequencies - Caps in a high/low pass reduce certain frequencies in volume (3db per Octave) not block completely ( fact check )

 

 

 

Big muff pi - Tone control

SWT - Control

Adams Amp Page - http://amps.zugster.net/articles/tone-stacks

 

 

LBB-1 Driving Jfet

......the next logical step is using the bipolar booster to drive the jfet circuit. The bipolar can produce plenty of gain that will overdrive the jfet and produce smooth distortion with musically pleasing harmonics... a 3 volt screamer!


The two stages generate a mild overdrive with rounded edges as the signal clips. The low gain and lack of bypass capacitor on the jfet contribute to the production of tube-like harmonics. Its simplicity is part of its attraction.

 

 

 

Reverse Polarity Protection 

http://www.singlecoil.com/docs/diode.pdf

To protect your stompboxes or active guitar wirings from damages caused

by wrong polarity, you only need a simple diode, eg. 1N4148. Insert this
diode in the way shown on the drawing

 

 

 

 

 

 File:Vactrol AGC (Yushin).PNG

 http://en.wikipedia.org/wiki/Automatic_gain_control

Could this be the basis for a sustainer circuit ?? 

 

 

 

 

 

 

http://www.singlecoil.com/docs/mini_booster.pdf

 

It has slightly less gain but a great sound. You can 
use the following trannies, sorted by gain and starting with the lowest 
gain: 2N5457 / J201 / MPF102 / BF245 
 

 

 

 

 

 http://www.youtube.com/watch?v=isn-JwEu_-8&list=WL94D81FE35412423F&shuffle=36147

 Just playing with a bridged T-Network wich is used in the TR808 for generating percussive sounds. I found the circuit on the pages of Eric Archer:

 

bridged T-network

 http://ericarcher.net/devices/cowbell/

 

 808 SVC MAN FIG 11-12

 

 

 

[Article] How Vacuum tubes work  

http://www.vacuumtubes.net/How_Vacuum_Tubes_Work.htm 

 

 

 



  

 

http://www.diystompboxes.com/smfforum/index.php?topic=50095.0;prev_next=next

some questions about building a cabinet simulato 

 

Op-Amps as amplifiers and Tone Controls

 

 http://www.st-andrews.ac.uk/~www_pa/Scots_Guide/audio/part8/Page2.html

 Fig2.gif - 27Kb

 Hence the non-inverting arrangement has a very high input resistance.

 Fig3.gif - 18Kb

 Consider first the effect of the upper pot (the  one). At high frequencies the pair of 47 nF capacitors act as a short circuit and clamp the three wires of this 100k pot together. Hence adjusting the 100k pot does not alter the high frequency behaviour of the circuit. However at lower frequencies the impedance of the capacitors rises and the pot has some effect. Hence the 100k pot acts as a Bass Control and allows us to boost or cut the relative gain for low frequency signals. Now consider the lower pot (the  one). Here the effect of the associated capacitors is reversed. The 10 nF capacitors mean that the arm of the circuit which contains the 47k pot essentially loses contact with the input and output at low frequencies. Hence the 47k pot has not effect upon low frequency signals, but it does upon high frequencies. It therefore acts as a Treble Control and can be used to boost or cut the relative gain at high frequencies.

 

 

 

 

 

Building JFET Preamplifiers for musical instrument use. by pyrohaz

http://www.instructables.com/id/Building-JFET-Preamplifiers-for-musical-instrument/

 

Unfortunately, there is not a very wide choice from JFET's as not very many companies actually produce these. The general ones for sale (after a quick eBay search!) are: MPF102, J310, 2N3819, 2N5484 and 2N5457 (my choice of JFET). I will show the design step by step on how to design a JFET amplifier using any existing N channel JFET through looking at the data sheet of the product.

 

The parameters we need to design ourselves a JFET amplifier are:

• Vcc (Positive supply voltage)
• Minimum Rds On of the JFET (resistance of the JFET when fully on biased)
• Ids (Current flowing through the JFET from drain to source)
• Cut off frequency of the JFET preamplifier
• Vgs Cut off voltage

From these parameters, we can calculate the values for Cin (Input capacitor), Rg (Gate resistor), Rs (Source resistor), Rd (Drain resistor) and Cout (Output capacitor).

 

When designing the circuit, you must ensure you don't exceed the voltage and current rating of the JFET. With the JFET I am using (2N5457), the maximum VDS (maximum voltage across the drain and source) is 25v, the Vgs (off) is -2.5v and the maximum Ids (maximum current across the drain and source) is 3mA. On the data sheet, the Ids is called Idss. All that Idss means is the current flowing through the drain of the JFET with the gate grounded. This value should not be exceeded and as a general rule, you should not exceed about 60% of the absolute maximum rating, this will prevent any broken devices and failures.

 

 

Click Here - How to calculate resistor Values from Datasheet info 

 

 

 output filter.png

 

we can now calculate the cut off frequencies of the input and output filters. A capacitor with a resistor in this configuration produces a high pass filter. This  can be calculated to work with another equation.

The equation for an RC filter (Resistor - capacitor filter) is:

          1
---------------- = Frequency cut off at -3dB
2 x Pi x R x C

 

 

When designing this product for instrument use, we will want to make sure that the -3dB cut off point is BELOW the lowest frequency produced on the instrument to allow the signal to pass through cleanly. On a standard tuned guitar, the lowest frequency produced is the open E string at 82.4Hz.

 

We also want to ensure that the impedance of the circuit is high, to not load preceding effects or the guitar pickup. A good value of high impedance will be between 100k and 1Meg.  If a higher value is used, the circuit will be noisier due to thermal noise of the resistor being amplified

 

So in our current situation, R = 100k and F = 82.4Hz

               1
------------------------- = 0.0000000191F, 0.0191uF or 19.1nF
2 x Pi x 100k x 82.4

If we choose a value of 22nF, we will get a cut off frequency of 72.3Hz, this is below our lowest produced note by a standard tuned guitar. If you want the amplifier to have a full frequency range, you can make this capacitor a large value such as 470nF. This will produce a cut off frequency of 3.39Hz! This is sub sonic and below the human hearing!

 

Always make sure that Rl is larger than Rd. A standard value for the output capacitor is 1uF. This gives a cut off frequency of 15.9Hz. Low enough for nearly every instrument!

As you can also see, my bias point lies just below the -0.4v point (between -0.4v and -0.6v) it is about -0.44v. All that this means is that if the gate is 0.44v LOWER than the source, it will follow this curve. Since we are pulling the gate to ground, we can achieve the same thing by putting the source to +0.44v. Therefore the gate is -0.44v relative to the source.
We can now calculate Rs:

 

 

 Now, the only problem with this method is if you look at the curves of the JFET, once you get past the linear region and into the saturation region, not much happens in terms of transconductance, you increase Vds and the Id hardly increases. This has a profound effect on the output waveform by squashing one end of the waveform. This can be very musical on guitar and adds a fair amount of second harmonic content
(something lots of guitarists crave!). This can be changed by moving the bias point much more towards or away form the linear region, all of which can be done by replacing the source resistor for a potentiometer or variable resistor.
Once again, since these aren't standard value resistors, the values:
Rd = 10k Ohm
Rs = 560 Ohm
Could be used.

 

 JFET schematic.png

 

 

 


Capacitors: A Field Guide to Types and Habitats

 by Harry Bissell

http://www.analogrules.com/capacitors.html  

 


 

 

 

 

 

Stupidly Wonderful Tone Control 3 - AMZ  [http://www.muzique.com/lab/swtc3.htm]


A simple change to the values of the capacitors will yield a response with more high end, which is a better response for modern rock styles.

As you can see, both capacitors have been reduced in value to open up the mids of the filter more than in the original.

The 22n value may be too small, depending on the circuit in which the control is used, and the value may be increased to 33n or 47n as needed.

This tone control is best used with circuits that have a lot of high frequency content that needs to be tamed, such as a fuzz (overdrive/distortion). It can only reduce high frequency content but is calaculated to have a response roughly similar to the speakers of an amp.

Add this to a dual mini-booster, or Rat clone, and have some smooth distortion sounds! 

 

 

 

 

 

 

 

 

Blatty or Gated sound - http://www.geofex.com/fxdebug/bias_prob.htm 

NPN transistors

  • For linear amplifying, the collector must be more positive than the base; the base must be more positive than the emitter by about 0.4 to 0.7V for silicon, and 0.0 to 0.3 for germanium; the emitter should be the most negative pin. The voltage difference between the collector and emitter is the biggest voltage that the transistor can swing linearly.  If the collector is not a volt or more above the emitter, or the base is not a base emitter drop above the emitter as noted above, the transistor cannot be acting as an amplifier. This is a very good pointer to what can be wrong. See the debugging example below. If the collector is at the same voltage as the base, or even closer to the emitter than the base is, the transistor is saturated and simply can't be amplifying. Likewise, if the base-emitter is below the cutoff voltage for the transistor, no current can be flowing.
  • If no sound happens until you really hit it with a lot of signal, the signal is having to supply the bias, so look for a transistor with its base too close to its emitter - less than 0.4V or so.
  • If no sound happens until the note is very quiet, look for a transistor that's almost saturated, with its collector voltage too close to its base or emitter voltages.
  • Debugging example: A fellow had built a replica of the Fender Blender circuit. It was not working, and I asked him to list the  voltages on the transistor pins. He found:

 

 

 

 

"Secret" of stacked stages

http://www.diystompboxes.com/cnews/mods.html 

It's been said many times by R.G. Keen and others that one of the the secrets to a "good" distortion/overdrive is to:

1: Control the gain between stages

2: Control the bass of the signal

3: Control the high end harmonics generated

Look at most of the overdrives that use transistors and series gain stages....

You will usually find:

Voltage dividers to control the gain between stages as well as other tricks to control gain.

Smallish capacitors (.01uF and smaller) for coupling caps which roll off low end from the signal.

High end roll off caps across the collector/drain resistors or from signal to ground.

Of course "good" in this case is controlled, smooth distortion.

If you want fuzzy, maxxed out distortion and FUZZ, well, then slam the signal as much as you want and forget about the above :-)

 

 

 

 

Booster to distortion pedal

http://www.diystompboxes.com/cnews/mods.html 

Simply add back to back (anti-parallel) diodes from signal to ground after the output cap to turn a booster circuit into a distortion circuit.

You can put the diodes anywhere on the output as long as there's a blocking capacitor before the diodes.

You can put 2 diodes or 4 for more output. You can also use germanium diodes or silicon diodes.

 

 

 

 

 

AMP designed to be plugged into a home Stereo?

  For Headphones, Direct recording and driving full range hi-fi Speakers 

 

i.e SansAmp - 3. Can I use headphones with my SansAmp? 
 

You can, but the output was designed to be low enough so as not to overload whatever you plug it into (mixer, combo amp).  Therefore, it may not be the most satisfying level.  We suggest you plug the SansAmp into a mixer or home stereo (auxiliary input) and then you can crank the cans as loud as you like.

 

 

 

 

Review of the SansAmp - Classic

     - http://www.tech21nyc.com/reviews/images/SansAmpClassic_SOS_0912.pdf 

 

 

 

 

 

 

Articles / Links [][][]

 

 

 

 

 



 

 

 

 

 

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Taken from the Rockman website  - [Rockman Headphone amps]


If this cab sim mimicks the frequency response of a guitar speaker, it cannot duplicate its dynamic response. And that was critical!

The main difference between a real amp and any amp simulator is that that the speaker has some physical, mechanical inertia that levels the tough peaks of a guitar signal. The attack of an electric guitar is way too strong to be pleasant to hear, and that's why we need these big speakers: they kinda compress these unpleasant attacks.

Compression? Well, there is a compressor in the Rockman! And that's the real secret of Tom Scholz guitar sound, much more than EQ's, filters and saturation.

The above diagram shows how the simple compression circuit of the Rockman limits the attack of the note, and attenuates the difference between the initial peak of the note and its decay. That's what a guitar speaker does with its mechanical inertia, and that what makes the Rockman sound realistic, combined with the cab sim.

 

 

 

From GMarts-Guitar Amplifiers  http://www.gmarts.org/index.php?go=211

 

What this means is that a hi-fi speaker is designed to reproduce sound accurately (the linear region) right up to its rated power capability. After that, you will start to damage the cone and quite likely burn out the coil.

An instrument speaker on the other hand will remain linear up to near its maximum power, then gradually restrict the cone excursion to provide a relatively clean, but somewhat compressed sound reproduction. When operating a speaker near its maximum power, there are still bursts of power such as drums, picking a guitar string, striking a piano note, etc. Instrument speaker are designed to handle these short bursts. Of course if you push it too hard, you will still cause cone and coil damage.

Instrument speakers achieve this by generally having a wider but shorter voice coil winding. The cone excursion is limited smoothly at its extremes, mainly for two reasons:
 - the coil starts to leave the linear part of the magnetic field
 - the stiff suspension keeps it under control

There are other differences - here is a diagram showing frequency response.

A hi-fi speaker and cabinet is designed to cover the entire audible spectrum without any colouration (variation) at all. This is achieved often with 3 or more speakers (woofers, mid-range, tweeters, etc), with other design features such as bass ports, crossovers and internal acoustic damping. They attempt to cover the range from 20Hz to 20KHz.

Guitar speaker cabinets are rarely acoustically padded, and crossovers with high frequency horns are not popular. Even bass ports are used only by players seeking the bass-heavy chugging styles. These speakers and cabinets have a strong, but broad middle emphasis. Bass response is typically rolled off around the lowest guitar note (80Hz); middle response is strong, with some peaks and troughs in the response around 2KHz to 5KHz , then a sharp roll-off of upper response above about 5KHz. 

 

 

 

Guitar Speakers

Unlike hi-fi speakers, which are designed to keep the coil entirely within the magnetic field to maximise linearity, instrument speakers are designed to have the coil partially leave the magnetic field at the extremes of cone travel. This is partly to protect the speaker, but also produces a "soft-clipping" effect which is desirable with guitar amplifiers

 

 

http://www.gmarts.org/index.php?go=214

imited bandwidth (amplifier and speaker in combination typically cover around 80-6000 Hz)

 

 http://www.gmarts.org/index.php?go=215

 

A traditional good guitar speaker system rolls off the bass gradually below about 100Hz, and highs are rolled off abruptly in the 4KHz to 8KHz octave, with some significant peaks and troughs above about 1KHz. This would colour an emulated sound, specially one doing its own speaker emulations, which might be trying to add peaks where your own speakers have troughs! 

 

 

  • What's the simplest compressor circuit I could build ?
  • What's the simplest way to create the frequency response of s a speaker cab (or a reasonably close approximation, doesn't have to be exact, just sound good through headphones ? 

 

 

 

 

 

 

 

 The Rockman Compressor 

 http://www.rockman.fr/Reviews/GC.htm

 

 "What we have here is very close to the simple compressor of the Rockman headphone amps.

The diode and the capacitor are a feedback loop that sends a strong negative voltage when the input signal is strong, thus reducing the gain, and a low negative voltage when the audio signal is low, thus raising the gain.

That's what we wanted to build

 


 

 

 

In the Rockman Sustainor's compressor, the capacitor is discharged via a series of 3 diodes anda LED. This trick allows getting the green curve, with a fast initial release rate followed by a slower second step.

Note that this fast initial decay is also necessary to provide the long sustain that we expect from a compressor: that's another key quality of the Rockman compressor.

This series of four diodes was the answer of SR&D to the transition between a limiter and a sustainor circuit, and this solution can be found only in Rockman compressors.

 

 

 

 


 

 

Bridged Amp's [][][]


 

 

BTL Amps - Wikipedia - http://en.wikipedia.org/wiki/Bridge-tied_load

 

Power output [edit]

Because the available voltage swing across the load is doubled for the same power supply voltage, bridged output theoretically enables the design of an amplifier producing four times the power output on the same supply voltage, since.

 

 However, operating a pair of existing amplifier channels in bridge mode only doubles or triples power output to the load, since the amplifier retains its existing current rating.

 

Benefits and drawbacks [edit]

Since two amplifiers are being used in antiphase, using the same power supply, there is no need for the use of a DC blocking capacitor between the amplifier and the load. This way, it will be more linear and less expensive, use less space, and there is no power loss through the capacitor.[5]

Bridging an amplifier increases the power that can be supplied to one loudspeaker, but it does not increase the amplifier's total available power. Because a bridge amplifier operates in mono mode, a second identical amplifier is required for stereo operation. For bridged amplifiers, damping factor is cut in half, sospeaker cables should have a large diameter conductor or be kept as short as possible for minimum resistance. Because the amplifier's bridged output is floating, it should never be grounded or it may damage the amplifier.

 

 

 

Bridged and paralleled amplifiers

http://en.wikipedia.org/wiki/Bridged_and_paralleled_amplifiers 

 

 

 

Paralleled amplifier[edit]

Representative schematic of a paralleled amplifier configuration.

paralleled amplifier configuration uses multiple amplifiers in parallel, i.e., two or more amplifiers operating in-phase into a common load.

In this mode the available output CURRENT is doubled but the output voltage remains the same. The output impedance of the pair is now halved.

The image shows two identical amplifiers A1 and A2 connected in parallel configuration. This configuration is often used when a single amplifier is incapable of being operated into a low impedance load or dissipation per amplifier is to be reduced without increasing the load impedance or reducing power delivered to the load. For example, if two identical amplifiers (each rated for operation into 4 Ω) are paralleled into a 4 Ω load, each amplifier sees an equivalent of 8 Ω since the output current is now shared by both amplifiers - each amplifier supplies half the load current, and the dissipation per amplifier is halved. This configuration (ideally or theoretically) requires each amplifier to be exactly identical to the other(s), or they will appear as loads to each other. Practically, each amplifier must satisfy the following:

  • Each amplifier must have as little output DC offset as possible (ideally zero offset) at no signal, otherwise the amplifier with the higher offset will try to drive current into the one with lesser offset thereby increasing dissipation. Equal offsets are also not acceptable since this will cause unwanted current (and dissipation) in the load. These are taken care of by adding an offset nulling circuit to each amplifier.
  • The gains of the amplifiers must be as closely matched as possible so that the outputs don't try to drive each other when signal is present.

In addition, small resistors (much less than the load impedance, not shown in the schematic) are added in series with each amplifier's output to enable proper current sharing between the amplifiers. These resistances are necessary, without them the amplifiers will in practice fight each other and overheat.

Another method of parallelling amplifiers is to use current drive. With this approach the close matching and resistances are not needed.

 

 

 

 

 

 DIODES and CLIPPING

 

Here is a case in point.

The MXR Distortion+, and its various clones by DOD, Ross, etc., uses a pair of back to back Ge diodes to clip the signal it receives from the op-amp booster stage. Ge diodes will conduct in a manner so that anything in the signal greater than about 250mv will be "clipped". The clipping of signal is what we hear as distortion. Using a back to back pair means that both the negative and positive half-cycles will be clipped equally.

Two hundred and fifty millivolts isn't a very hefty signal, though, so the output of the Dist+ isn't that substantial, meaning that although you can get it to give you buzz and fuzz, you can't really overdrive your amp with it.

If you were to replace those Ge diodes with Si type diodes (1N4148, 1N914, etc.), you would hear the following differences.

1) Because Si diodes generally (more about this below) don't clip signals until about 500mv, the "ceiling" on the output of the Dist+ will be raised, and in this case, almost doubled. So, you will have more volume. I have a little Dist+ I made, with a switch to select between Ge and Si diode pairs, and the difference in output level is quite noticeable.

 

 

 

To some extent, some of the aspects of tube-type distortion can be emulated by use of asymmetrical clipping - one half cycle clips more than the other. One of the ways people often do this is to combine Ge and Si diodes in a back to back pair (each type oriented opposite to the other). Given that each diode has a different clipping threshold, one half cycle will clip more than the other.

The clipping threshold set by diodes is additive, meaning that if you were to put two Ge diodes of the same orientation in series, in parallel with two other Ge's of the opposite orientation, you would have symmetrical clipping, with Ge quality, but with a higher clipping threshold (double, or roughly 500mv). Similarly, stick a Ge and Si in series, and the threshold (for that half cycle) is now 500+250=750mv. You will sometimes see circuits that use pairs of 2 and 3 diodes in series.

This is why you will often see combinations of diodes in diode clipping circuits, and occasionally mixes (e.g., the Fulltone version of the TS-9 uses a 1N34, 1N914 and 1N4001).

 

 

 

Articles [][][]

 

 

 

 

 

 

 

 

From the Rockman Website - http://www.rockman.fr/Reviews/Rockman.htm 

 

"A few words about the Rockman Distortion. If you are receptiveto the marketing speech of these common stompboxes manufacturers, I'm sure that you think the Rockman distortion had something special.

Well, the Rockman distortion is the only classic part of the circuit!

A distortion stage is nothing: compression, that alters the dynamics, and filters, that process the harmonics balance of a sound, are the critical factor. The Rockman Distortion in itself doesn't have any real importance, just like tubes or solid-state has a very limited impact on the final sound of a guitar amp: its design is from far more important.

And here's a proof of this statement: though they sound alike, the Rockman and the X100 don't have the same distortion circuit...

The following diagram describes the clipping (saturation) stages of both the Rockman and the X100. They are clearly different, and the corresponding waveforms are also different. The Rockman uses what is called a "soft-clipping circuit" with 4 diodes, while the X100 is based on a "hard-clipping circuit" with 2 LED's".

 

 

 

 

TONE ?? PAGE

 

 

From GM arts - Guitar Amps - http://www.gmarts.org/index.php?go=212 

Tone Controls

Magnetic guitar pickups are inductive, and require compensation, although this opportunity is also used for tone enhancement, not just correction. Without compensation, they have a strong low middle emphasis and little high frequency response - overall a somewhat muddy and muffled sound.

"Hi-fi style" treble & bass controls that boost or cut bass and treble separately are occasionally used in guitar amplifiers. They are suitable for particular styles such as jazz when you want your basic tone to have a general low-middle strength. But by far, the most common tone control in guitar amplifiers is a design called a tone stack, so called because the bass, middle and treble controls are electronically connected in series, usually drawn as a vertical stack. This design has the advantage of inherent middle-cut, serving a dual purpose of general tone balance correction along with tone control.

To hear the natural sound of a pickup on an amplifier with a tone stack, set the middle set to full, and bass and treble on zero. This actually sets a flat response in the amp (see below), and I expect you will hear a muffled and muddy sound. And that's the whole point of these tone controls providing compensation for the natural sound of a pickup - the middle control simply boosts the pickup's normal middley sound. The treble and bass controls do the opposite - they boost higher and lower frequency levels, leaving a notch in-between for middle cut (see the Fender/Marshall comparison below). So with typical settings of a bit of bass, middle and treble, the overall tone equalisation complements the natural pickup sound for a balanced response of lows, mids and highs.

Fender Middle EQ

Full middle boost with no bass or treble actually gives a near-flat frequency response, allowing you to hear the natural sound of your pickups.

Here are circuit diagrams of typical Fender and Marshall tone controls. They both meet the criteria of compensating for pickups' low-middle emphasis, as well as providing a useful range of tone adjustment.

Fender Tone Circuit   Marshall Tone Circuit

The Fender and Marshall circuits are each tailored to suit their own styles, which are quite different. Although a generalisation, Fender's market and power output stage are geared towards provided clean and chunky tones at clean and early-overdrive levels. Marshall amps are best at low-middle and crunchy rock tones, played at medium to high overdrive levels.

Here are some frequency response plots showing the range and effect of Fender and Marshall tone controls. These are screen captures from the excellent Tone Stack Windows program developed by Duncan Munro. If you're interested in this sort of thing, you can download the program from Duncan's website.

 

 See Marshall vs Fender - FREQUENCY GRAPHS - here

 Bearing in mind that 6-string guitar notes don't go below 80Hz, and typical guitar speakers cut above about 5KHz, these responses are similar. Both have a middle dip that is primarily compensation for typical pickups' middle emphasis, rather than an obvious dip in middle response. The Marshall circuit has this cut about an octave higher than the Fender, leaving the low mids and bass intact for that full Marshall sound. On the other hand, Fender's tone controls allow high-mids to pass with the treble response, and add little bass boost for the sparkling and tight sounds they're famous for.

 

 

 

The Fender circuit also has the unusual side effact that if all controls are set to zero, then no sound is produced at all. The Marshall design avoids this, but the tone with all controls set to zero is not something you'd be likely to use anyway.

Here are the same charts for Marshall tone controls. As mentioned already, the main points to note are the smaller range of adjustment, the higher frequency for the middle cut control, and the higher overall signal level. The smaller adjustment range and higher level are both caused by the use of the 33K resistor in place of Fender's 100K. This also gives the tone stack a lower input impedance, requiring it to be fed from a lower output impedance (cathode follower) preamp valve stage.

 

 Valve power amplifiers often provide an additional presence control (which reduces negative feedback in the power amplifier section) to provide a small amount of boost at frequencies above the treble control.

 

 

 

 

 

 

THE TRUTH ABOUT "TRUE BYPASS " (and some other goodies).

 

Tech21 (SansAmp) - http://www.tech21nyc.com/technotes/index.html

 

There were other added benefits to using active bypass circuits. Active bypass converts the signal to low impedance making it resistant to "tone sucking" from other pedals and poor quality or extra long cables down the signal path. Designers were able to improve life expectancy over big clunky switches by using smaller, more resilient parts. Active bypass gave the option to use a fast cross-fading (about 50 ms) FET circuit, which eliminates the loud pop associated with the passive true bypass circuit. As a bonus, designers were able to easily add an LED indicator into this circuit. For passive bypass design, this addition would have required the use of a triple-pole 

 

 SansAmp Classic

The 3-position input switch gives you a choice of pre-amp styles:
LEAD
NORMAL
BASS
For Marshall®-type pre-amps with mid-range and highs emphasizedFor Mesa Boogie®-type pre-ampsFor Fender®-type pre-amps (excellent for rhythm guitar as well as bass)
The knob controls shape pre-amp contours, power amp contours, volume and final tone. With SansAmp Classic, there are countless (we know, we've tried) permutations for you to explore.
Sample style settings include: Vintage (AC/DC) and Hot-Wired (Van Halen) Marshall®; Mesa Boogie®; Hiwatt®; Fender® Lead (B.B. King), Rhythm and Bass; and Ampeg SVT®.

 

 

 

 

 

 

D.I.Y Cab Sim - Schematics


  

 

 

 

Cab Sim - [Run off groove]

(http://www.runoffgroove.com/cabsim.html)

 

From the run off groove site - "The circuit consists of two jfets in a common source configuration. Jfets were chosen because of the high input impedance and smoother breakup properties if overdriven. Following the first jfet, is two low pass RC filters. The first filter is adjustable. The 5k Treble pot set at maximum will place the corner at 10kHz. When turned to minimum, the corner will be at 2.5kHz. This will afford some adjustment of the amount of high frequencies that are allowed to pass unaltered. The second RC LPF is fixed around 15kHz to eliminate some of the harsh treble frequencies generated by fuzz boxes. These two filters working in tandem provide a fairly smooth treble rolloff that increases as you move up the frequency band. Next is the second jfet gain stage, set to recover some of the volume lost in the filtering process. On the output of the second jfet, we have a variable simple high pass filter, composed of the output cap, 33k fixed resistor, and 50k pot wired as a variable resistor to ground. With the Bass control at maximum, the low end goes down to 192Hz. When set at minimum, the corner of the filter is set at 483Hz".

NOTE: if you cannot find the specified MPF102, other FETs can be used, but the 6k8 drain resistors should be replaced with 50k trimpots. Adjust the trimpots for 4.5v at the drain of each FET. 

 

 

 

 

 

 

Simple Sim - [Joe Davidson

(http://www.diystompboxes.com/analogalchemy/sch/simplesim.html)

 

 From D.I.Y stompbox forum - "This is a relatively easy amp simulator, for recording purposes. It goes after a distortion box to smooth out the sound and provides a low-resistance output. It can be pushed into slight overdrive, like an output stage of sorts. The resulting overdrive is very smooth. The .001uF cap is largely responsible for the overall tone of the circuit. Adjust to taste. The circuit is biased so that the output emitter is near 4.5v." 

 

 

 

 

 


 

 

Condor Cab sim - [Run off Groove]

(http://www.runoffgroove.com/condor.html)

Condor

 

[From the ROG site] The circuit is composed of the following functional blocks:

 

  • JFET amplifier. Amplifies the signal to drive the filters. Also adds some overdrive at higher Gain settings, which emulates the behavior of a real speaker operating at high volume. We liked using a J201 best, but an MPF102 or 2N5457 will provide more treble-content and different overdrive characteristics.
  • Bridged-T notch filter. Produces the 400Hz notch found in the P10R response. Please note that the extra resistor in parallel with the capacitor to ground is not only for biasing the op-amp, but it also produces a high-pass shelving effect. High frequencies are then 6dB above low frequencies, as shown in the speaker response curve. To reduce the notch depth, you can replace the 1n and 15n caps with 1n5 and 10n, or even 2n2 and 6n8, respectively.
  • High-pass filter. This is the network between the first two op-amp stages. It simulates the low frequency rolloff of the speaker cabinet, plus the 90Hz peak. For deeper bass, you can replace the two 47nF caps with 56nF or 68nF. For less bass, use 39nF or 33nF.
  • Low-pass filter. These are the two identical networks between second and fourth op-amp stages. They simulate the steep high frequency rolloff of a guitar speaker, in addition to the peak near 3kHz. For more high frequency content, you can reduce the four 22k resistors to 18k or less. To cut more highs, increase them to 27k or more

 


 

 

 

 Bulldog Cab sim - [Ambrose Chapel]

 

Schematic posted over at diystompboxes. Can't find site for original and creator. More mods for Bulldog - Here 

 

 

 

 

 

 

Cab sim / Headphone amp - [Aron Nelson

Headphone amp &  Cab sim With mid scoop  by Aron Nelson - Found at diystompboxes

Taken from the diystompboxes by Liquids - "In the end mine looked a heck of a lot different than this generic schem, but it's got most of the concepts I used.  Speaker sim alone is not enough for most ears...you need a pre-amp sim too, which most forget, as most guitar amps have serious mid scoop and are never flat...flat EQ with guitar sounds like Pat Metheny, but muddier"

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

Commercial Cab Sim - Schematics


 

 

 

Marshall Cab sim  (On all XLR equipped Marshall amplifiers)  

 

Schematic from - (http://www.diystompboxes.com/smfforum/index.php?topic=44939.20

 

 

Schematic from -  (http://diy-fever.com/effects/marshall-cab-sim/)

 

 

 

 

 

 

 

From Hexe Guitar - Cab Sim Article  

 

 

From Hexe Guitar cab sim article - As an electronic circuit a Cabsim is a set of active or passive filters that create a desired frequency response. In its simplest form it's a typical low pass filter with roll off at about 4-5kHz, since this is the usual bandwidth limit of a guitar speaker (-3dB roll off).

Such a simple solution was implemented in TRIAMP amp by H&K (below) - Frequency Response here 

 

 As one can see from the schematic, the cabinet simulator is composed of two low pass filters connected in series.

 Similar idea (but based on LC elements) can be found in Record Out device in Triaxis preamp by MesaBoogie.

 

 

 

Cream machine - [Hex Guitar]

One step further; if we examine the frequency response of a guitar cabinet, we will notice another important feature - low frequency roll off starting usually at 100-200 Hz. This is achieved by using a high pass filter. Many amps contain various devices based on these two filters. Few examples - again by H&K - are cabinet simulators available in discontinued preamp series: Cream MachineCrunch MasterMetal Master  - Fequency Response here

 

 Other Schematics, frequency Graphs and sound samples here - (http://www.hexeguitar.com/diy-cabsims_e)  

 

 

 The next preamp by Mesa Boogie - V-Twin - has a simple filter, which can deliver purring blues tones. As usual for Mesa, the filter is based on inductance. Its schematicsound sample, and frequency response: