Jul 222013

Burnt Speaker

Frequently at Uncle Sam’s Jamms Service Center, we receive speakers for warranty repair in which the woofer voice coils are frozen and charred. The voice coil that drives the cone has overheated and burnt. Many times the owner believes that these failures will be covered by the manufacturer’s warranty. Unfortunately, in nearly all cases, the failure is due to excessive power applied to the speaker, which is specifically excluded from coverage by manufacturer warranties.

There are two main causes of speaker failure:

1 – Continuous power output from the amplifier is greater than the speaker is designed to handle.  This usually burns up the voice coil in the woofers.

2 – Excessive distortion caused by trying to get more volume (power) from the amplifier than it is capable of delivering. (Underpowered system.)   This causes the amplifier to go into clipping, which produces excessive high frequency energy and burns up the tweeters or horns.

So … you’re telling me that I can do damage to my system by having either too much power, and also not enough power? Exactly.   Let us explain.

Cause #1:   Excessive power to speakers. Usually, the reason for these occurrences is because the amplifier power rating was much greater than the speaker was designed to handle. We see this situation frequently in speakers used in DJ applications, where the program material contains continuous heavy bass signals. Even worse, often the amplifiers are over-driven into clipping due to the desire for even more bass.

Unfortunately, amplifier ratings and speaker power ratings as described by the respective manufacturers are not directly related to each other, and the consumer often chooses a speaker with a rating that appears to match the amplifier, when in fact the speaker is severely underrated. That is, an amplifier rated at 100 watts does not necessarily match a speaker rated at “100 watts”.

In order to properly understand the rules for speaker selection, a certain amount of technical background is necessary. Boredom alert! But try not to skip over this section. Understanding the technical details can save you a ton of money.

First, a definition:  POWER is the rate at which energy is converted from one form to another. The basic international unit of energy is called a JOULE and is defined as the amount of work energy performed by applying a certain amount of force (called a NEWTON) through a distance of one meter. The number of Joules of energy converted each second defines the amount of Power in WATTS. 1 Watt = 1 Joule/second; 5 watts = 5 Joules/second, etc. For instance, if your exercise bike had a generator connected to a lamp, the brightness of the lamp would depend on how hard and fast you pedaled. The faster and harder you pedal, the more power you generate and the brighter will be the lamp.

The purpose of an amplifier and speaker system is to convert the raw electrical energy from the AC power line into acoustic (sound) energy that you can hear. During this process, energy is converted from matter (such as coal or gas) into heat, then mechanical energy, then (possibly through magnetic energy) into electricity. From your AC power outlet the electrical energy is controlled by the amplifier in order to build a weak electrical signal into a strong signal, which is fed to the speaker to create magnetic energy, which pushes the speaker cone back and forth (mechanical energy), which moves air molecules in order to convert some of that mechanical energy into sound energy. The more energy (per second) is converted into sound, the louder the sound will be. So volume depends on the amount of electrical POWER used to create the sound by the speaker.

There are a number of methods used to describe the power of an electrical signal. These methods are given names such as “instantaneous power”, “average power”, “RMS Power”, “Peak Power”, “Music power”, “Program Power” and others.

In order to move the speaker cone in the complex manner required to accurately reproduce sounds, the power must be continually and rapidly varying. The power being used at any instant during the process is called “instantaneous power”. However, since that is always changing, it is not useful for describing the capability of an amplifier or speaker.  In other words, if you see or hear the term, “instantaneous power” … move along.  Nothing to see here.

“Peak power” is the maximum amount of instantaneous power present at the highest level during the signal. In a speaker, the peak power would occur at (approximately) the instant the cone reaches its most forward (or rearward) extended position. In an amplifier, the maximum peak power output to a speaker is limited by the amplifier power supply. If the level controls are increased beyond the point where the amplifier reaches the limits of the power supply, a severe form of distortion known as “clipping” occurs. As an analogy, think of tying a weight to a string and whirling it up and down in a circle in the room. If the string is short, the circle is small. This is like an amplifier delivering a low volume tone to a speaker. If you lengthen the string, the circle gets bigger. An amplifier would be delivering more power to the speaker, and the sound would be louder. If you lengthen the string too much, the weight will hit the ceiling and/or the floor, and the weight no longer moves in a circle. The size of the circle (and the amount of sound) is limited by the ceiling and the floor (the power supply in the amplifier). In an amplifier, the peak power rating is useful for describing the maximum instantaneous limit of its capability for pulse sounds such as drumbeats and bass notes.

“RMS power” is practically the same as “Average power”. Since an audio signal is constantly changing, mathematical methods were developed to accurately compare the voltage and current in an audio signal or AC power source to an equivalent DC level such as that produced by a battery. The method of comparison was based on the amount of heat each form of power would produce in a heating element such as a lamp bulb. So the values of voltage and current producing 10 watts of AC electricity would have the same average power as a battery supplying a steady 10 watts to a lamp, and would light it to exactly the same brightness. The Average Power (often called “RMS” power) is the most consistent method of comparing power levels between two devices. Most audio power amplifiers are rated for their Maximum Continuous Average (“RMS”) power output capability of an essentially undistorted signal to a specified load (speaker) impedance. Under certain conditions, a relationship exists between the peak power and the “RMS” power rating of an amplifier. (The term “RMS” stands for “Root-Mean-Square” and describes the mathematical steps required to calculate the effective values of voltage or current which determine the average power of an AC sine wave, which is the graph of instantaneous signal in a pure tone. Strictly speaking, “RMS” applies only to voltage and current, not power. “RMS power” is technically meaningless, but has become commonly used to refer to Average Power because it is usually calculated using the RMS values of voltage and current, and also to distinguish it from other “types” of power.)
“Music power” and “Program power” are rather nebulous terms that are often used in speaker ratings. They are explained by the manufacturers by saying that speakers are almost never used to produce pure tones (where average power is most easily measured) and that the power distribution in most music is erratic and of many frequencies. A closely related industry standard method of rating speakers is called the IEC power rating. The description usually given is that a complex waveform is used to evaluate the power capacity of the speaker. No exact formula or other mathematical relationship appears to be available, but from our research it appears that these ratings are approximately twice the equivalent average power.

Herein lies the problem. How can you match a speaker system to an amplifier when the amplifier is rated in “RMS” watts (average power) and the speaker is rated in Music Power or Program Power (which is definitely NOT average power)? Ultimately there must be some method of comparing apples to apples when it comes to the two.

Remember that average power was defined in terms of the “heating effect” produced by the AC power source. When audio power is applied to a speaker, much of the energy is converted to magnetism and then to sound. However, a significant portion of it gets converted to heat in the wire of the voice coil. When too much power is applied, the heat damages the insulation of the wires. They come loose and start to rub inside the speaker, or the insulation starts to burn. This causes a buzz or rattle from the speaker. If the wire gets even hotter, it fails like a burnt-out lamp bulb filament and the speaker quits entirely. Since it is heat that causes the speaker failure, you can see that it is the average power that is important in determining the failure limits of the speaker.

Now let’s consider the amplifier power rating. If you examine a specification sheet of an amplifier, you will see that they are usually rated in Average (“RMS”) power for an UNDISTORTED output. This is like swinging your weighted string so it almost touches the ceiling and the floor. But keep in mind that you can overdrive the amplifier into clipping. When this happens the power amplifier can produce up to TWICE AS MUCH power to the speakers as it is rated for. (Long technical explanation required – take my word for it.) So an amplifier rated at 100 watts clean can actually put out as much as 200 watts when heavily over-driven into severe distortion.

One other note about amplifier power ratings: The output power per channel usually depends on the IMPEDANCE of the speaker(s). For example, an amplifier might be rated at 100 watts per channel to an 8 ohm load and 190 watts to a 4 ohm load. Be sure you are comparing amp and speaker ratings for the same impedance value. We will assume you are only using one speaker per channel in this discussion.

Now, if you want to keep from smoking your speakers, you should consider the relationships between the speaker ratings and the maximum capability of the amplifier. The first thing to do is find out what form the speaker power rating is. If the speaker rating is clearly stated in RMS WATTS or AVERAGE POWER, you can compare it directly to the amplifier rating. If you are buying an amplifier rated at 100 watts/channel, a speaker rated at 100 watts RMS is usually adequate. However, if there is a possibility that it might be over-driven, (such as with lots of bass or heavy metal) you might want to buy a speaker rated at up to twice the power output rating of the amplifier to allow for the distorted output power. (Distortion of bass notes is not as obvious as higher frequencies, and it’s easy for enthusiastic DJs to overdrive the system without realizing it.)

If the speakers are rated in Program power or Music power, or if it is not clearly stated, you should also consider the additional rule of thumb that their average power rating is probably about half of the Program power rating. If you are buying an amp rated at 100 watts clean (200 watts maximum), your speakers should have an RMS Average Power rating of 100 to 200 watts, so the Program power rating should be 200 to 400 watts each.

In practice, for an application where the program material contains high average power levels, (especially lots of bass) the speaker Program Power rating should be MORE than two times the amplifier RMS power rating. For maximum protection, select speakers with Program Power ratings of four times the amplifier power rating. Keep in mind that these rules are based on approximations and some assumptions, so no guarantee can be made that you still won’t blow the speakers, but using these rules will greatly lessen the risk.

The speaker in the photo above was rated at 240 watts IEC for a duration of 8 hours, or 200 watts IEC for a duration of 100 hours. This appears to translate to approximately 100-120 watts RMS. (The manufacturer doesn’t give an RMS rating for this speaker.) The user was driving it with an amplifier rated at 500 Watts RMS per channel (with both channels driven). This amp is capable of producing over 1000 watts per channel when heavily distorted. The system was being used in a DJ application. More than likely the DJ was cranking it as loud as he could…..

Case #2: Excessive distortion. As mentioned previously, excessive distortion burns up tweeters. This happens when the amplifier does not have enough power to produce the desired volume and the user has set the volume controls too high. The amplifier goes into clipping (described above), the sound becomes harsh and distorted, and produces lots of high-frequency harmonic energy. Also as described above, the amplifier can actually put out up to twice as much distorted power as it can clean power. A lot of this extra distortion power is contained in those high-frequency harmonics, which are directed to the horns or tweeters in the speaker. The result is a burnt tweeter voice coil.
Distortion can also be produced in devices feeding the amplifier, with the same result. A mixer channel that is being overdriven, or an overdrive signal from a guitar amp can produce a distorted signal that can burn out tweeters even though the power amp is not operating near its full power rating. (That’s why most guitar amplifiers don’t have tweeters!)
There is much conflicting information available on the web and from supposed ‘experts’ that is based on the idea that distortion must be avoided at all costs. Some even go so far as to say the amplifier power rating should be two times the speaker’s IEC power rating(*1). The rationale is that speakers can normally handle brief pulses well in excess of their average power rating. While this is true, such applications are typically found in home theater and studio monitoring, where the average power level of the program material is normally well below the speaker rating. It is NOT a safe practice for systems where high continuous average volume is required, such as dance clubs, arenas, theaters and outdoor venues. If you go back and examine the ratings for the burnt speaker and the amp that was being used, you will see that the user thought he was following the (erroneous) recommendations.
To avoid damage due to excessive distortion, be sure your system’s amplifier power rating is great enough for the application to ensure that you have enough volume without overdriving it into distortion. Select your speaker power ratings to match the amplifier using the guidelines above, adjusting for the type of program material. If you are amplifying speech, speaker Program Power rating = amplifier RMS power is probably adequate. For recorded music or live acoustic-style performances, speaker Program Power rating = 2x amplifier RMS power would be appropriate. For live rock or ethnic music, or heavy-bass DJ situations, you may wish to increase the safety margin beyond this. (*2)
A properly designed amplifier/speaker system for sound reinforcement will have enough amplifier power that clipping distortion of the amplifier will never occur at any desired volume level, AND speakers that can handle continuous power near the rated output of the amplifier. Using an amplifier with twice the power rating of the speaker (or more) is inviting a trip to the repair shop.

Hopefully you’ve been able to learn a bit about speaker power rating and amplifier power rating from this article. To sum it up, compare apples with apples – not apples with oranges. RMS power is the true rating that should be used. If both amplifier power and speaker power is given in RMS wattage, a good rule is that the speaker capacity should be at a bare minimum equal to the amplifier output power. If you can go to twice the power (1000 watts RMS speaker capacity for 500 watt RMS amplifier) – that would be even better. Make sure you note the amplifier power rating for the impedance at which you are loading the amplifier. If the speaker (or amplifier) rating is given in Music Power or Program Power, divide that number by 2 to obtain RMS Power. If it is given in Peak Power, divide that number by 4 to obtain RMS power. Avoid overdriving (clipping) at all costs. Bass signal is where this usually occurs. For DJ’s, if you need heavy bass buy a appropriately sized subwoofer with sufficient power, and with the use of an electronic crossover, run all signals below 100 to 125 Hz to the subwoofer. If you find yourself running the amplifier and mixer into clipping in order to achieve the needed volume levels – your system is too small. This frequently occurs when attempting to set up outdoors. It takes a tremendous amount of power to fill up an outdoor space with sufficient sound pressure, because there are no walls to contain the sound. In so many cases, when we receive blown speakers and we question the owner about the last venue where the speakers were used and subsequently failed – they tell us that it was outdoors.

Apr 172013
Proper use of a graphic EQ

Proper use of a graphic EQ

Let’s talk about equalization, and how EQ should be utilized in your system.

EQ’s should be used as a subtractive device. That means they are primarily for lowering certain frequencies in a pre-amp or mixer. (See the photo for a good example of an EQ being used by a knowledgeable engineer.)

The ‘smiley face’ you’ve all seen on EQ’s only means that . . . the ‘engineer’ has no clue as to how to use it. No slider should be very far above ‘0’! Anything over ‘0’ adds noise, and also increases the likelihood for feedback.

This is extremely important in P.A. and recording situations. Every mix or amp sound has too much of something to start with. “…Too much bottom end …”, “…too many high’s …”, “… mids are honkin’ …”. Use an EQ to lower those trouble spots.    Result . . .  Cleaner sound, cleaner mixes, better sound.    Simple, isn’t it?
EQ should be subtle – not drastic.

The same goes for individual channel EQ on your mixer. Start with all of the EQ knobs straight up (12 o’clock). Make minor changes to tweak the sound of each channel – perhaps as low as 10 o’clock, or as high as 2 o’clock. Between 11 o’clock and 1 o’clock is preferred. Just because the knobs go all the way up to 5 o’clock and all the way down to 7 o’clock doesn’t mean that you need to try them there! Again, EQ should be subtle – not drastic. Keep this in mind, and improve the sound quality of your mixes.

Until next time, I’m Frank the Tech Guy.

 Posted by at 11:27 pm
Oct 242012

Modding the Fender Hot Rod DeluxeModifications (or “mods”) is one of the most common requests I receive as a musical instrument electronics tech.   As musicians search for that elusive tone, many seem to be willing to tweak the features and tone of the instruments and amplifiers they already own.

Now, there are two schools of thought on modifications.     The first philosophy seeks to … well, leave things alone.    If the amp or guitar you currently have doesn’t offer the features or tone that you are looking for, then you need to look for a replacement guitar or amp that does.

Certainly, this is a valid approach.    However, it can be a bit self-defeating unless you have deep pockets and a lot of time on your hands for testing amplifiers in music stores.    And … you may never find the stock instrument or amplifier that satisfies all of your needs.

Probably a more popular concept among most musicians is to locate the gear that suits you best, and then consider modding it (if you are so inclined) or having it modded to fill any additional requirements.

So, let’s take a look at the mods that musicians today are wanting.     Let’s start with guitars.    Probably the most common modification for guitars with humbucking pickups is the addition of a coil tap and/or phasing switch.    The coil tap cuts out one side of the dual-coil humbucker, making it essentially a single coil pickup.     This allows your Les Paul to sound more like a Strat – a useful addition for certain songs.     The modification is often done with the addition of a pull-switch pot, so that no holes need to be drilled into your instrument.     On a two pickup guitar with individual volume and/or tone controls, the coil tap mod can be added on both pickups.   To switch to the coil tap sound, just pull the knob on the corresponding volume control.   This is a simple mod, and is completely reversible.    It does require humbucking pickups which bring out the interconnection of the two coils in the cabling (typically either a 3-wire with shield or 4-wire with shield pickup cable).   Cannot be done on vintage pickups (such as old Gibson humbuckers) with only the single wire plus shield cable.   Note that like single coil pickups, your guitar will be a bit more “hum-prone” when you engage the coil tap feature.

Another common mod is phase reversal and series/parallel.   These mods require the 4-wire with shield pickup cable.   Again, by using the pull switch, you can wire the two coils in series (that’s the standard), in parallel, in phase and out of phase.   Each gives a uniquely different tone to your pickup.   Slightly more difficult to wire than the coil tap, this mod is still not outside the ability of many guitar players with some basic electronics knowledge.

Another guitar mod is the blend pot.    Used on dual pickup guitars that generally have four controls, the blend pot can replace one of your tone pots (you would then have two volumes and one master tone), or you could go with one master volume and two individual tones.    The now available control spot becomes the blend control.   By switching to one of your pickups, the blend control allows you to bring in a bit of the other pickup, depending on where you dial in the control.    If you switch to both pickups on your selector switch, the blend pot is overridden and you have the standard dual pickup sound.  Again, not a very difficult mod to install for the talented do-it-yourselfer.   Or, have it done at your local guitar repair shop – we get a lot of requests for these mods at Uncle Sam’s Jamms.

If I talked about all of the guitar amplifier mods that are commonly requested, I could build a library.   So, I’ll touch on the most common.    One of the most requested modifications that I have is the “blackface” mod for silverface Fenders.    When CBS bought Fender Musical Instruments from Leo Fender in 1965, they continued the product line pretty much unchanged for two or three years.   But then they started to tinker with the designs as well as the cosmetics.  By the time they brought out the silverface versions in the late 60’s and 70’s, they had made some changes to the amplifier designs which most musicians today believe were not in the best interest of tone quality.    So, the blackface mod is designed to take a silverface Fender and change the electronics back to the Leo Fender design.     In a nutshell, this changes the tube balance control back to a true bias control, beefs up the bias supply filtering, changes the resistance network around the phase inverter tube, cleans up the dressing of the wires, and removes the snubber capacitors on the output tubes.     The result is an amplifier that sounds fuller, more sparkling, has a more pleasing breakup when driven, and can now be biased properly when new matched pairs or quads of power tubes are installed.

Speaking of Fender, other common modifications that I install regularly include a number of mods for the “Big Four” Fender retro amps – the Hot Rod Deluxe and Deville, and the Blues Deluxe and Deville models.    The first mod is to fix the overly sensitive volume controls in the clean and the drive channels.   Both the clean volume control and the master control on the drive channel are very touchy at the low end of their range.   It’s hard to adjust for a suitable volume in a smaller venue – the amp gets too loud too fast.    The volume sensitivity mod fixes that problem.

Additional mods for the Fender “Big Four” include power supply stiffening with the addition of another filter capacitor in the power supply.    Also popular is the conversion of the tone stack to the “Twin Tone Stack”.   This enhances the action of the bass, middle and treble controls, increasing the dynamics of the amplifier.

The Fender Blues Junior is also frequently modified with the power supply stiffening and the Twin Tone Stack mod.    Another mod for the Blues Junior involves adding an internal bias control.    The fixed bias on the Blues Junior generally runs the bias much too hot.    It sounds pretty good, but your EL84 power tubes are being roasted.     The bias control allows them to run cooler while still maintaining a good tone – and they will last considerably longer.

Other mods that we had to a variety of amplifiers include the addition of a master volume control (we only install the post-phase inverter type of master volume on tube amps).     We also install a power limiting switch on 100 watt tube amps to drop their power to 50 watts while automatically reconfiguring the output impedance correctly at the same time.

As I said, covering all of the amplifier mods would take a complete library.   But if you desire it, in many cases there is a solution for your amplifier issue.  Until next time, I’m Frank the Tech Guy.

 Posted by at 3:01 am
Apr 062012

The Father of Loud' … Jim Marshall, creator of the Marshall amp, in 2005. Photograph: Damien Maguire/Rex Features

Jim Marshall, founder of Marshall Amplification, died on April 5, 2012 at his home in Milton Keynes, Buckinghamshire, United Kingdom.      Here is a brief biography of his life, courtesy of Wikipedia (http://en.wikipedia.org/wiki/Jim_Marshall_(businessman) All rights to the holder of the Wikipedia copyright, as per the terms of the Creative Commons Attribution-Share Alike License.

James Charles “Jim” Marshall, OBE (July 29, 1923 – April 5, 2012), known as The Father of Loud or The Lord of Loud, was an English businessman, and pioneer of guitar amplification. His company, Marshall Amplification, has created kits used by some of the biggest names in rock, producing amplifiers with an iconic status.  Marshall received an OBE honour for “services to the music industry and to charity”.  Marshall has been listed as one of the four forefathers of rock music equipment along with Leo Fender, Les Paul and Seth Lover.

Early life

Marshall was born in Acton, West London, in 1923, into a family which included boxers and music hall artists. As a child he was diagnosed with tubercular bones, and spent many years in the hospital. His formal education suffered as a consequence. During WWII he was exempt from military service due to his poor health. He became a singer, and then, due to the shortage of available civilian musicians, doubled as a drummer. In his day job as electrical engineer he built a portable amplification system so his light, crooning vocals could be heard over his drums.[ “I was making 10 shillings (£0.50/$0.75) a night and because it was wartime, we didn’t have any petrol for cars, so I would ride my bicycle with a trailer behind it to carry my drum kit and the PA cabinets which I had made! I then left the orchestra to be with a 7 piece band and in 1942 the drummer leader was called into the forces and I took over on drums.”

In order to become more proficient on the drums and to better emulate his idol, Gene Krupa, from 1946-48 Marshall took weekly lessons from Max Abrams. In the 1950s, Marshall started teaching other drummers, including Mitch Mitchell (The Jimi Hendrix Experience), Mickey Waller(Little Richard) and Mick Underwood (Ritchie Blackmore). Marshall commented, “I used to teach about 65 pupils a week and what with playing as well, I was earning in the early 1950s somewhere in the region of £5,000 a year (eqv. 2012 to £108,000/$170,000), which was how I first saved money to go into business.”

Marshall Amplification

From 1960, Marshall owned a moderately successful music store in Hanwell, west London, selling drums and then branching out into guitars. His many guitar playing customers (including Ritchie Blackmore, Big Jim Sullivan and Pete Townshend) spoke of the need for a particular kind of amplifier and Marshall saw the opportunity.   He recruited an 18-year-old electronics apprentice, Dudley Craven, who was previously working for EMI and, with his help, began producing prototype amplifiers, resulting in the foundation of Marshall Amplification, in 1962. It took Marshall six attempts to create an amp he was happy with, creating what later became known as “the Marshall sound”.   As the company grew, Marshall expanded his products, and unveiled the Master Volume Marshall amps and the classic JCM800 split channel amps.  Soon after he started production, musicians including Jimi Hendrix and Eric Clapton were using his equipment.

In 1984 Marshall was awarded the “Queens Award for Export”, an honour bestowed by Queen Elizabeth II of the United Kingdom in recognition of Marshall Amplification’s outstanding export achievement over a three-year period.  In 1985, Marshall was invited to Hollywood to add his hand prints to the “Rock and Roll Walk of Fame”.   In 2003, Marshall received an OBE honour from Buckingham Palace for “services to the music industry and to charity”, and he has donated millions of pounds to worthy causes including the Royal National Orthopaedic Hospital in Stanmore, London, where he was treated for tuberculosis as a child.

Death and legacy

Jim Marshall died on 5 April 2012 at his home in Milton Keynes, Buckinghamshire. Former Guns N’ Roses guitarist Slash commented; “The news of Jim Marshall passing is deeply saddening. [Rock and roll] will never be the same without him. But, his amps will live on forever!”.   Mötley Crüe bass player Nikki Sixx also paid tribute, stating Marshall was “responsible for some of the greatest audio moments in music’s history – and 50% responsible of all our hearing loss.”   In 1967, Jimi Hendrix stated, “There’s nothing that can beat my old Marshall tube amps, Nothing in the whole world.”  Marshall has been named, along with Leo Fender, Les Paul and Seth Lover, as one of the four forefathers of rock music equipment.


1.  Jim Marshall, Founder Of Marshall Amps, Passes Away Aged 88″. Metal Hammer. Retrieved 2012-04-05.

2.  Guitar amp pioneer Jim Marshall dies aged 88 BBC News. Retrieved 5 April 2012

3.  Guitar amplifier pioneer Jim Marshall dies aged 88 Reuters. Retrieved 5 April 2012

4.  Jim Marshall, creator of the Marshall amp, dies aged 88 The Guardian. Retrieved 5 April 2012

5.  Saunders, William (2010) Jimi Hendrix London Roaring Forties Press ISBN 978-0-9843165-1-9

6.  The Jim Marshall Story MarshallArts.org. Retrieved 5 April 2012

7.  Marshall Amps Founder Jim Marshall Dies Planet Rock. Retrieved 5 April 2012

8.  Obituaries: Jim Marshall The Telegraph. Retrieved 5 April 2012

9.  Jim Marshall, founder of Marshall amps, dies at 88 San Francisco Chronicle. Retrieved 5 April 2012

10. Jim Marshall, guitar amp pioneer, dies aged 88 Montreal Gazette. Retrieved 5 April 2012


 Posted by at 2:25 am
Mar 292012

What are ohms, what is impedance?
Short answer: The ohm is the unit of measurement for impedance, which is the property of a speaker that restricts the flow of electrical current through it. Typical speakers have impedance ratings of 4 ohms, 8 ohms or 16 ohms. The impedance of a speaker is a physical property that (ideally) does not change value, although from an engineering standpoint, there are many complex characteristics that make up speaker impedance For this reason, the rating of a speaker is called its ‘nominal’ value, which pretty much means “in name only”. For the average audio user, the nominal impedance is the dominant characteristic and for the purposes of this discussion, we will use the nominal value of the speaker’s impedance.

Why are “ohms” and “impedance” critical?
Two reasons:
(1) If you connect your amplifier to the wrong speaker impedance, you greatly risk damaging the amp. In tube amps, too high a load impedance (or a disconnected load) can result in damage to the output tubes or output transformer. In solid state amps, if the speaker impedance is too low, the amplifier will tend to overheat and more power is used up in the amplifier than is delivered to the speaker. Too many speakers (meaning too low of an impedance) on a solid state amplifier will likely burn up the output section.

(2) The amplifier will deliver maximum power (volume) to the speaker when the speaker impedance matches (is equal to) the internal impedance (called the OUTPUT IMPEDANCE) of the amplifier. Too low of an impedance will result in weak output and poor tone. If the speaker impedance is higher than that of the amplifier, its power output will again be less than it is capable of.

Understanding Ohms and Impedance
In order to understand the reasons for the rules for speaker connection, we need a bit of electrical theory. In order to relate it to something you are more familiar with, let’s consider the garden hose. Go outside, hook up the hose (no nozzle) and turn on the water. Pretty soon, water should start flowing out the end of the hose. This flow of water through the hose is similar to electric current, which is usually described as the flow of electrons through the wire and is measured in Amperes, and generally shortened to the word “amps”.

Now put your thumb over the end of the hose and try to stop the flow of water. Feel the pressure? This water pressure is similar to Voltage. It is the force of electricity that pushes the electrons through the wire. Notice that if you succeed in plugging the water flow, the pressure is still there, but there is no water (current) flow. This is like an amplifier with no speakers attached, or an AC outlet with nothing plugged in. Voltage is present, but there is no current flow.

Finally, move your thumb a bit to allow some water to spray. By varying the position of your thumb, you can control how much water comes out of the hose. Your thumb is restricting the flow of water. In an electrical circuit, things that restrict or control the flow of current are said to impede current flow, and are described as having impedance. In a hose, we use a nozzle to restrict the flow. In an electrical circuit, the device that uses electrical energy and has impedance is called the LOAD. In a light circuit, the light bulb is the load. In an amplifier system, the speaker is the load.

It should be apparent by now that there is a relationship between pressure (voltage), flow (current) and restriction (impedance). Since voltage or pressure is what moves the current, increasing the voltage pressure should increase the current, assuming the impedance doesn’t change. Decreasing the voltage should decrease the current. On the other hand, increasing the impedance restricting the flow of current will cause the current to decrease, like turning the nozzle toward OFF. Lowering the impedance is like opening the nozzle to allow more flow. This relationship was analyzed by a fellow by the name of George Simon Ohm a long time ago, and he identified a simple formula that is extremely important in electricity and electronics which bears his name: Ohm’s Law.

Ohm’s Law states: In an electrical circuit, current flow is directly proportional to voltage and inversely proportional to impedance. Mathematically, this becomes: Current (in amperes) equals voltage (in volts) divided by impedance (in ohms).

As an example, if a (solid state) amplifier is producing 10 volts AC to an 8 ohm speaker, the current in the speaker will be 10 volts / 8 ohms or 1.25 amperes. If the amplifier output is increased to 20 volts to that 8 ohm speaker, the current becomes 20 Volts / 8 ohms or 2.5 amperes. So increasing the voltage increased the current. If the voltage decreases back to 10 volts, the current will decrease back to 1.25 amperes.

Now, if our amplifier with 10 volts output is connected to a 4 ohm speaker, the lower impedance will allow more current to flow. The amount will be found by 10 volts / 4 ohms = 2.5 amperes. If we use a 2 ohm speaker, even more current flows: 10V/2 ohms = 5 amperes.

Finally, if we can measure or in some other way determine the amount of current being drawn from the amplifier, we can calculate the value of the load impedance using Ohm’s Law. We will use this shortly to figure out what happens when we connect several speakers to the output of an amplifier. The formula for this is: Impedance (in ohms) equals Voltage (in volts) divided by Current (in amperes).

Let’s use an amplifier with banana jack terminals and connect the red terminal of the amplifier to the red or ‘+’ terminal of an 8 ohm speaker. Also connect the black terminal of the amp to the black or ‘-‘ terminal of the speaker. If you feed a pure tone through the amp so that it delivers 10 volts to the speaker, the current flow through the speaker (as we saw above) should be 1.25 amperes.

Next, let’s connect another 8 ohm speaker to the amplifier terminals in the same way, so you have two wires from the amp’s red terminal going to the ‘+’ terminals of the speakers, and two wires from the amp’s black terminal to the speaker ‘-‘ terminals. This is called a PARALLEL connection, because of the way it looks in an electrical schematic diagram.

The first thing to understand is that the voltage output from the amplifier ideally does not change. So it’s still 10 volts AC. And since each speaker is connected directly to the amp’s output terminals, each speaker will receive 10 volts from the amplifier. As we saw earlier, if 10 volts is applied to an 8 ohm speaker, it will draw a current of 1.25 amperes from the amplifier. And if each speaker needs 1.25 amperes, then the amplifier must supply a total of 2.5 amperes to the two speakers. If you add a third speaker, it will also draw another 1.25 amperes, (total 3.75 amperes) as will a fourth (which would total 5 amperes). If you keep adding speakers, at some point the speakers will demand more current than the amplifier can deliver, and it gives up – it smokes and dies. Too many loads is an overload.

Now, we are ready for impedance. As we said earlier, if you know the voltage and can figure the total current, you can calculate the total impedance of all the speakers together by dividing the voltage by the total current. A single speaker is simple: 10 volts divided by 1.25 amperes equals 8 ohms.

Remember that two 8 ohm speakers would draw a total of 2.5 amperes from a 10 volt output. So 10 volts divided by 2.5 amperes equals 4 ohms. Notice that adding a speaker in parallel DECREASED the total impedance. What about 3 speakers that draw 3.75 amperes? 10 volts divided by 3.75 amperes equals 2.67 ohms. Four speakers that draw 5 amperes from a 10 volt source have a total impedance of 10 volts divided by 5 amperes which equals 2 ohms. As more speakers are added, each one draws additional current from the 10 volt source, so there must be less total restriction of current. So the first thing to conclude is that ADDING SPEAKERS DECREASES THE TOTAL OHMS IMPEDANCE.

Well, what if the speakers have different impedances? Like an 8 ohm cabinet and a 4 ohm cabinet? The same method can be used. To make it simpler, remember that impedance was a physical property that doesn’t depend on the voltage. The speaker has the same impedance whether the source is 10 volts or 1 volt. So let’s use 1 volt to make it simpler. The 8 ohm cabinet would draw 1V/8 ohms or 0.125 amperes. The 4 ohm cabinet would draw 1V/4 ohms or 0.250 amperes. Both together draw 0.375 amperes. Total impedance is 1V/0.375 amperes, or 2.67 ohms. (Notice that the total is less than the lowest value speaker. That is always the case when connecting speakers in parallel).

A 4 ohm, an 8 ohm and a 16 ohm cabinet all connected to the same amplifier (1V out) would draw currents of 1/4, 1/8 and 1/16 amperes, for a total current of 0.4375 amperes. Impedance is 1/0.4375, or 2.286 ohms. (Using a calculator with a 1/x key makes this really simple. Key in: 4 (1/x) + 8 (1/x) + 16 (1/x), =, (1/x) and read the answer.)

While the calculations may seem complicated, examination of the results above reveals some patterns that make things much easier.
First, if all speakers (or cabinets) have the same impedance ratings, the total impedance can be found by using the impedance value of one speaker and dividing that by the total number of speakers. If you go back to our example of 8 ohm speakers, we found that a single speaker had a total impedance of 8 ohms, two 8-ohm speakers had a total impedance of 4 ohms (8/2); three speakers had a total impedance of 8/3 ohms, or 2.67 ohms, and 4 speakers totaled 8/4 or 2 ohms.

Second, the 2:1 relationship between typical speaker impedance ratings allows for some equivalents when mixing different ratings. A single 4 ohm speaker is the equivalent of two 8 ohm speakers in parallel. So a 4 ohm speaker combined with an 8 ohm speaker would have the same total impedance as three 8 ohm speakers in parallel. (See if you can figure out the equivalents for a 4, 8 and 16 ohm speaker combination.)*
So, if you see a speaker jack labeled “Minimum Load 4 ohms”, that means you can connect up to two 8 ohm speakers or a single 4 ohm speaker to that jack.

If you are mixing speakers with different impedance ratings, be sure to check the total impedance using the rules above to be certain the total is within the limits of the amplifier. Solid state amps typically have a ‘minimum load impedance’ indicated near the speaker terminals, and the total speaker impedance must be equal to or greater than that value. Tube amplifiers typically have a switch on the back to adjust for the speaker load impedance. Tube amps have different output characteristics than solid state amplifiers, and too low a load impedance will not normally damage them, but the total output will become weaker and muddy. So too little load impedance is still undesirable. Too high a load impedance on a tube amp can cause high voltages inside the amp that can damage power output tubes or the output transformer.

So, how do you tell what the impedance of a speaker is? On most cabinets, it should be printed on a label next to the jack. If the speaker is visible, it may be printed on the speaker label or stamped on the frame or magnet. To measure the true impedance of a speaker or cabinet requires a rather complex procedure involving signal generators, power amplifiers and high frequency AC voltmeters. However, with raw speakers and many cabinets, the ohmmeter function of a digital multimeter can help you identify what the impedance of the speaker should be. Generally, the reading given by an ohmmeter will be about 2/3 to 3/4 of the impedance of the speaker. So, a 4 ohm speaker will typically measure about 2.5 – 3 ohms, and an 8 ohm speaker will typically read about 5-6 ohms, while a 16 ohm speaker will measure around 12 ohms.

Another thing…. As a general rule, all speaker jack connections are considered parallel connections and will follow the above rules. So if you run a cable from the amp to a speaker that has two jacks, and run another cable from the second jack on the first speaker to a second speaker, it is still a parallel connection.

I hope this has helped clear up some misconceptions about speaker impedance and will help you to correctly connect the proper number and type of speakers to your amplifiers. Until next time, I’m Frank the Tech Guy.

 Posted by at 5:39 am
Oct 102011
Pedal board

In my daily work I receive a number of questions about the proper order in which to interconnect effects pedals – which ones should go first (closest to the guitar), which ones go last (closest to the amp), and which ones fit well in the middle.

Well, it must be said first that there is no definitive answer to this question, and neither should there be. Getting that unique and original sound is often about breaking the rules, rather than following them. However, this article provides a suggested order, and some explanation as to why that arrangement seems to work properly. However, the most important judge of whether your effects are in the right order would be your ears. If you like the sounds coming from your amp, then don’t listen to what anyone else tells you. Experiment by putting your pedals in all sorts of weird orders and you may stumble on something remarkably cool. The suggestions here, however, would be a good place to start. Rather than listing every single model of effect available, I have grouped effects into families. These should be recognizable to most guitarists.

As I mentioned, the “order” that we are using starts at the guitar, and ends at the amp. For some reason, most stomp boxes have the input jack on the right, and the output on the left. Therefore when you lay your effects out on the floor, the signal goes from right to left.
The theory behind this order is that first should shape your sound, then introduce effects that add color or modulation into the sounds, and then introduce effects that take away from the sound.

1. Distortion and overdrive.

These should come immediately after your guitar. The tone from your guitar and the overdrive form the basis of your tone. This goes first because you only want to be distorting or overdriving the sound of your guitar. The other effects in your chain are carefully sculpted, and if you distort them, you would end up with mush.

2. EQ and Wah.

Some people like to have an EQ immediately after their guitar, rather then after the distortion. Some people like to have EQ before and after the distortion. In an ideal world you would probably have an EQ after each effect, to fine tune the sound at each stage. In the real world that is not practical. Wah pedals are really a type of sweepable EQ that boost and cut the frequencies of the signal as you rock the pedal. In my experience, the wah pedal works best when placed after the distortion. This seems to be because the frequencies that are boosted and cut are those that contain the frequencies of the distortion. This means you get a very linear sweep, with one particular band of the distorted signal boosted at anyone time. Many people (Jimi Hendrix included) have the wah pedal in front of the distortion. This means you are feeding the distortion with different a particular frequency at any one time. The distortion pedal will react differently to differing incoming frequencies, so you will get a different “color” of distortion tone across the sweep of the wah. However, this seems to give the wah a much more binary, (on or off) feel, with a narrow section in the middle that does most of the wah-ing, and any movements of the pedal either side of that narrow band having little effect.

3. Delay (echo)

Some delay pedals come with reverb built in. Unless you will never ever buy any modulation type pedals, avoid them. You don’t want your delay and reverb to be happening at the same point in the chain. Get a delay pedal and a separate reverb pedal. Now that you have built, shaped, and colored your tone, you may want to repeat it using delay. Delay pedals simply repeat the sound that goes into them a number of times, normally getting quieter with each repeat. You want to have your delay before the modulation, because you don’t want your modulation effects to be repeated. Only sounds before the delay will be repeated. Sounds created after the delay will only be heard once.
Once the guitar tone has been created, overdriven or distorted and “shaped” with EQ and wah, and repeated as required with delay now is the time to start adding color and flair to the sound.

4. Modulation.

In this section, I include phasers, chorus pedals, flangers, and envelope filters. These are all effects that add to the sound. They add color, noise and often volume. These effects need to go after your distortion because if the distortion goes after then, you will then distort the subtlety of the effect. If you have a nice chorus effect for example, adding a lovely shimmer and sparkle to your sound, do you really want to then distort that shimmer and sparkle? Probably not.

5. Noise Gate

The noise gate normally goes near the end of the chain, because it is designed to remove background noise which comes from the electronics in the guitar (if it has active pickups) and in the other effects pedals. A noise gate would be pretty ineffective at the beginning of the chain. My choice is to put it at this point, close to the end of the chain.

6. Effects that take away from the sound

There only seem to be two types of pedals that fall into this category: Tremolo pedals and volume pedals. Tremolo pedals slice up the sound much like someone quickly turning the volume up and down on your amp while you are playing. Volume pedals don’t add volume, they reduce it. A volume pedal on full volume would be the same as not having a volume pedal. These types of effect should come last, because you want the complete sound to be cut. If you put the volume pedal after the guitar, only the volume of the guitar would be cut. The volume of all the subsequent effects would stay the same. This would mean that when you reduce the volume on your pedal, the guitar sound would cut out, leaving all sorts of hissing and whooshing from your other pedals. By having the volume pedal last, you can control the overall volume of the signal going to the amp. The same applies to tremolo pedals. By having them at the end, they cut out all the noise, leaving you with silence in “off” periods. (Assuming the depth on your tremolo pedal as at max.)

6. Reverb

Reverb breaks the rules I’m afraid. I just said that effects that add sound should come before effects that take away from the sound. Reverb is the exception, and I can justify this by imagining what happens to a guitarist playing in a big concert hall without a reverb pedal. (He doesn’t need one, the concert hall provides the reverb.) All the sounds are created and effects are added and blast out of the amp. Then the sounds bounce around inside the hall and get jumbled up. Any silent periods in the music are filled with the sound of the previous section still bouncing around. Therefore to emulate this, reverb pedals go after effects that cut the sound. However, if you want your tremolo and volume pedals to completely cut the signal, put your reverb pedal first.

Note on Effects Loops

The last thing to mention is effects loops. Effects loops are a “send” socket and a “return” socket on the back of your amp. These are normally used to add in modulation, delay and reverb effects. These exist because many people like to use the natural over drive of their amp to get their distortion, and therefore don’t want to line up modulation, delay or reverb pedals before the overdrive happens, for reasons I have already mentioned. The effects loop is a circuit that is in between the actual amplifier in the amp and the speaker, so is similar to having your modulation effects after an external overdrive or distortion device.

So, there you have it … as I said, a starting point. Experiment, and most of all enjoy all of the variations and endless possibilities with your effects pedals. Until next time, I’m Frank the Tech Guy!

Jun 132011

Last time we discussed compressors, expanders, limiters and gates – devices that do not make drastic alterations to your sound, but are nonetheless very important. Next, we discuss effects that make “drastic” changes to your sound, namely Gain, Overdrive and Distortion. First, let’s dispel the myth that these three effects are the same thing. Similar, yes – identical, no.

First, let’s talk about gain. Gain by itself is technically not an effect, but is a parameter used to describe the amount of boost in your signal without changing tonal qualities. Essentially, gain is a function of the level of a preamp. Sometimes you will see “gain” as the label on the control of a stomp box or perhaps a channel on your amp. In the purest context of gain, lead players use the gain function to lift their guitar above the other instruments in the band and let it stand out in the mix. You will want to be careful however if you have multiple stomp boxes with gain settings. If you add more gain to every effects box and use multiple boxes at once you can end up blowing your audience out the back wall of the room, as well as drowning out the rest of the band.

Next, we’ll talk about overdrive. Overdrive is also a parameter that you may see on some effect boxes. This is generally described as the warm distorted sound that you get from cranking a tube amp’s volume up. You can change the dynamics of overdrive just by the way that you play. If you play softly on the strings the overdrive does not show up much but if you start to play harder the overdrive starts coming through. This makes overdrive a very dynamic effect. It is a slightly distorted sound that can really add to the overall tone you are looking for in your playing style. So you don’t have a tube amp, and even if you did, you can’t turn it up loud enough in your apartment complex to get the overdrive sound. What do you do? There are stomp boxes on the market today that allow you to get an overdrive sound at lower volumes. I have used the Tube Screamer stomp box and really enjoyed the sounds I got from it. This is probably one of the most widely used overdrive effects units on the market today.

Finally, distortion. Distortion can be one of the most difficult effects to choose. There are hundreds of different distortion effects on the market today and a new guitar player looking for his new sound is going to have a brain aneurysm trying to decide which one to get. What makes distortion stand out from the other effects we have dealt with so far is the fact that there is no standard in distortion stomp boxes. For example, if you buy two different chorus stomp boxes and set the parameters the same, you are going to get almost the same sound with very little variance in effect. Sure, you may notice some subtle differences, but the standards for these types of effects are very close from manufacturer to manufacturer. This is not the case with distortion stomp boxes. Not all distortion pedals are created equally. Distortion can be hugely different from manufacturer to manufacturer, and even from model to model in the same manufacturer’s line. Do not despair. There are some ways to sort through this crazy maze of sound. Let’s break down how to select the right distortion for you.

The first thing you need to do is figure out what kind of distortion you are looking for. This is based on the kind of music you are going to play, whether it be grunge, classic rock, heavy metal, modern rock, country (yes even country has distortion), or any other of the many styles of guitar playing out there. Once you figure this out you are on your way to choosing the right distortion for you. The best way to find your dream distortion is to start weeding-out the ones that are not right for your style.

What is in a name? The best place to start your weeding-out process is by reading the descriptive name on the units you are looking at. For example, if you are a blues or country player you may want to avoid stomp boxes that have words in their description such as “Death,” “Metal,” “Grind,” “Grunge,” “Atomic,” “Nuclear,” and “Atom Splitter.” These are definitely not going to shape your sound to your taste and would more than likely be a waste of time to even check out. Check them off your list. If you are a metal or grunge player you will want to avoid units with descriptive words like “Blues,” “Warm,” “Fuzz,” and even “Overdrive” unless it is preceded by a word like “Death,” “Metal,” “Grind,” etc. Now keep in mind that all the different pedal manufacturers are trying to outdo each other in their descriptions so some pedals may not even give you a clue as to where they fall in this mix. Those you are not sure of, you should check out. By following this simple step of name analyzing you can probably cut your selection process in half or better.

Another way to help in your selection process is to remember that most tube distortions will have a warmer and smoother tone while solid state distortions have more edge to them. This does not mean to discount them for this property, but it is another reference you can use while weeding the less desirables out.

When you get around to trying a different pedal always start with the unity gain control at “0.” This will allow you to hear what the pedal does to your sound without an increase in volume. This is important because you want to hear the pedal’s nuances before you crank it up. This is what will set the bar. Now start playing with the controls and see what you can come up with. Do not change your amp settings for different pedals. Set it up once to get your favorite tone before you start and then don’t touch it. If you do, you will be skewing the results of your testing. The distortion unit should sound good with your clean tone. If it does not sound good, you will have to sacrifice your favorite clean tone for the distorted tone. You need to find something that will work with the clean tone you like to use so that everything sounds good when you are switching between the stomp box in the on position and off position.

Now at this point you have probably weeded out the units that you did not like. I suggest that you make a list of two or three that you really like and one or two “possibles”. If you’re impatient and need a pedal to take home right now, select the one you like best and buy it. If you have a little more patience, I suggest that you take a break for at least an hour or so and go do something else that is away from a lot of noise. I suggest this simply because your hearing degrades over time when being assaulted with sound. If you take a break for awhile you can come back with fresh ears to try your favorites again. This will help you to select the one that sounds best to you.

Wait, what about those of us who play many different styles of music? There are two choices here. You can select a couple of different pedals for the different styles or you can go with a multieffect unit that has a couple of distortion choices – such as many rack-mount units on the market today.

Can you use two distortion stomp boxes together? If you get the tone you want by doing this, go for it. In music, the end always justifies the means when it comes to the tone you like and want to play.

I hope these discussions of effects and how to properly use them has been helpful. Meanwhile, I’m still Frank the Tech Guy – keep on playin’ that music!

 Posted by at 4:44 am
Mar 292011

In Part 1 we talked about reverb, delay, chorus, and flange. These are the “echo” or “delay” effects. In Part 2 we will discuss what compressors, limiters, expanders, and gates do and how they can be used. Note that these effects are what we call “control” or “behind the scenes” effects in that they do not intentionally modify the sound for variety or the enjoyment of the listener … rather, they control the basic sound so that it behaves itself. Let’s see what we mean by this …

What does a compressor do? Compression is arguably the effect most misused by guitarists. This misuse is probably because many players do not understand what compression is or what it does. When a new guitar player gets hold of a new effects unit they expect that it is going to alter the sound of the guitar in a very noticeable way. A compressor is not this kind of effect. Think of it as a ghost in the shadow of the effects chain. If you weren’t looking very hard for it you would not even notice that it was there.

So what exactly is a compressor? A compressor is a signal processor that is used to reduce the dynamic range between the softest and loudest parts of the audio signal. Think of the sound of your guitar as a wave. There is a peak where it is at its highest level and a trough where it is at its lowest. Sometimes these highs and lows get out of hand; that is, the highest levels start clipping, and the lowest levels are drowned out and are inaudible in the mix. A compressor reduces the highes and brings up the lows so that your dynamic range is, well … compressed. This stops the clipping of the louder passages and improves the sound while preventing costly damage to your equipment. It also brings out your quieter passages so that the listener can hear them.

Compression can be used on any single instrument or on the mix as a whole. Many bands and studios use compression on vocals because of the dynamic range of the human voice. Very high notes take more power to belt out and will sound louder than the low notes that sound softer because of the lack of power used to hit them. Using the compressor will make the singer have a more consistent range of volume. It also works great for live applications where the singer is constantly moving his head toward and away from the microphone. Bass players use compression to make a smoother-sounding transition between notes.

One of the benefits of compression on a guitar is the longer sustain of notes or chords. As the signal is starting to dip below the floor, the compressor will open up and let more signal through. This will allow the signal to be audible longer resulting in more sustain.

You will need to set up four main parameters on a compressor. These parameters are the compression ratio, threshold level, attack time, and the release time. The compression ratio sets the level of compression that will take place once the signal reaches the threshold. A 3:1 compression ratio would mean that for every 3dB of signal above the threshold there would only be a 1dB increase at the output. The threshold level is the level the signal needs to reach in order for the compressor to kick in and start working. The attack time is the amount of time it will take the compressor to react to the incoming signal. The release time is the amount of time it takes for the compressor to allow the signal to return to a normal level.

There are two main types of compression—hard knee and soft knee. The knee is the moment that the compressor starts to reduce the gain when the signal reaches the threshold. Hard-knee compression cuts the signal off abruptly when it reaches the threshold point. Soft knee is a smoother gain reduction that lets the sound taper off at the threshold point. Most good compressors on the market today let you switch between hard knee and soft knee.

It is very important to understand that compression can be overused and cause your signal to sound flat and weak. Use it with care, experiement with it, and improve your sound. Remember, a compressor is a tool to tweak your sound – use it carefully. Just as seasoning adds to the flavor of a good dish but too much can ruin an entree, so goes compression.

Limiters are quite similar to compressors. As we talked about earlier, a compressor reduces louder signals of your instrument. Unlike a compressor, a limiter completely cuts the signal off at the threshold. It is a great device to save your sound system from hitting levels that can damage or destroy it. It will not let your signal go above the maximum level you set. So unlike a compressor, the limiter is only going to deal with the louder levels of your signal.

Expanders are completely the opposite of compressors. Whereas a compressor is going to reduce your signal into a set parameter, the expander is going to widen that signal within set parameters. Why would you want to do that? An example of the use of an expander would be finger-sliding sounds on an acoustic guitar. Let’s say that you have the finger-sliding sound coming through your mix on the lower end of the audible signal. If you expand or exaggerate the signal dynamic width, it will push this sound further down into the mix so it will not be so pronounced. Another example would be breathing sounds from a singer at the microphone. Again if the signal is expanded it pushes unwanted sounds further down the audible signal. An expander is usually used to reduce unwanted background noise in the mix.

A gate works like a limiter but at the other end of the signals dynamic range. A gate cuts the signal off below the set parameters level. Unlike the compressor which pulls the lower level sounds up, a gate chops it off completely. Like an expander it is used to keep unwanted background noise out of the mix. One very common use of gates is miked drums. When the drums are miked up individually, you do not want other drums sounds bleeding into the wrong microphones. The gates cut off the signals bleeding over from other drums and clean up the mix. This is effective in keeping the cymbal sounds from bleeding over onto the tom tracks. It is used most dynamically on the snare drum mic and hi-hat mic because of their close proximity. Gates work well with guitar for eliminating the hiss and unwanted noise heard when the instrument is not being played.

So where should these effects be put in the effects chain?
There are two schools of thought on where to put compression in the effects chain. The first is that it should be at the beginning, so a smooth clean signal hits all of the other effects. I do not buy into this method personally because I know that some effects are going to cause the signal to clip again. The method I like is to put it after all the effects except the echo effects like reverb, delay, chorus, and flange. I want to compress all the effects in the signal but let the reverb-type effects fade out naturally. Both methods work for different people so you will want to try them both and see which works best for you. These methods also apply to limiters and enhancers. As far as a noise gate goes, I like them at the end of the effect chain so they cancel out any residual noise caused by the effects unit, especially flange. Flange has a tendency to cause a lot of background noise when the guitar is not being played.

Next time, we’ll look at Overdrive, Gain and Distortion. Aren’t these all different names for the same effect? Actually, no. They’re similar, but not the same thing. Stay tuned! Until next time, I’m Frank the Tech Guy.

 Posted by at 2:09 am
Mar 202011

I just want to give you a quick note about effects before we get started. You can’t set up your effects in your practice area and expect them to sound the same in the club this weekend. Room acoustics change the dynamic sound of your effects. Allow yourself plenty of time to set up your equipment and tweak your effects at the club before you start playing. If you don’t do this, you can have a disastrous show and may not be invited back to that club again. Remember that every club is a different room environment. It is best to keep a notebook on how you set up your effects at each club in case you play there again.

Now let’s talk about specific effects.

Echo Effects:


“Reverb” is a good place to start since it is built in to most guitar amplifiers on the market today (and for the past fifty years!). Reverb is an echo effect reminiscent to sound you get when you are overlooking a canyon and can’t resist the urge to shout “hello” and hear it echo back at you. If you want an example of reverb, turn the reverb knob all the way up on your guitar and strum it. Immediately mute the strings with your hand and you should hear the echo reverberate from the speaker.

In the early days of recording reverb was done in different ways. One way was to place a microphone at one end of the room and another close to the speaker cabinet. You would then record the guitar on two tracks and play them back together giving a sort of echo effect. If the effect needed to be tweaked or changed the engineer would move the microphones or speakers around the room until the desired effect was achieved. Another way of achieving reverb is to place the microphone and amp in a bathroom. We have all sung in the shower before and thought we sounded pretty good, right? Bathrooms are small rooms with hard tile walls that reflect sound instead of absorbing it. Weird Al Yankovic recorded many of his early albums in a bathroom.

There are many types of reverbs that you may run across in your search for the best sound for you. Three of the most common are spring reverb, plate reverb, and digital reverb.

As a general rule of thumb too little reverb is always better than too much. Too much reverb in the mix can make the sound muddy and drown out vocals and other instruments.

Delay is probably one of the most valuable effects. It is the building block that many other effects such as reverb, flange and chorus are based on. A delay is basically what the name says. It is a delay of the original signal of the guitar that plays at a set time after the original notes or chord is sounded. It can range from milliseconds to several seconds depending on how you set the time parameters. When set at several seconds you can actually solo over yourself.

Slapback delay is probably the most commonly used and can range from 30 milliseconds to 100 milliseconds.

Chorus is a version of delay and is my favorite effect for clean sounds. Chorus gives the impression of multiple instruments playing the same part. The unit puts a very small delay in the signal and (depending on the amount of delay) it detunes the echo to give the effect that another guitar is playing with you. This effect adds a sparkle and clarity to your sound. Chorus is best and most often used with slower tempo songs.

Flange and Phase Shifters:
Flange was created by accident in a studio. It was found that if you played back a reel-to-reel tape of the guitar track, held up the reel (the flange of the reel, to be exact) with your hand, and then let it go, it would catch up with the other tracks causing what would become known as flanging. The best way I can describe this effect is that it’s like riding a roller coaster. You go up the hill slowly (the engineer holds the reel back with his hand) and when you hit the top you pause for a millisecond and then you rush to the bottom very quickly (the engineer lets go of the reel to let it catch up with where it should be). Then you start up the hill again slowly (the engineer holds it with his hand again) to do it all over again. The other instrument tracks would be played normally and the guitar track would be held up and let go to catch up at regular interverals throughout the track. This created a “whooshing sound” on the guitar track kind of like a jet engine.

Thanks to electronic and digital technology we can reproduce this effect with stomp boxes and multi effects called flangers. Flange moves in and out at a constant and steady predetermined speed that you set.

Phase shifters are like flangers except they have multiple flanges going on at the same time and sometimes with no predetermined speed. Both are great additions to any guitar player’s setup.

A couple of things you need to remember about using effects in general. There are no set rules on how to use these effects. You can use just one or you can use a combination of all of them. While the echo effects are similar in ways they have their own distinct differences that can complement one another in a mix. Play around with them and have fun. You never know what you will come up with.

All the effects I have mentioned have their own parameters that can be adjusted giving you a full range of variations to play with. You can find your own signature sound by using what you like from each one.

Effects can improve the sound of your guitar and sometimes can make you think you are playing better. They can cover small mistakes, but I urge you not to use them for this. That is not the intended purpose of effects. You should be able to play a part cleanly with good technique before you start adding effects. If you can play it clean, it will sound even better with effects.

If you use compression and gates always put your reverb and delay behind the gates in the effects chain so they fade out naturally instead of being cut off by the gate.

In Part II of ” ‘Effective’ Use of Effects”, we will look at compressors, limiter, expanders, and gates.

 Posted by at 11:59 pm
Oct 182010

For those of you who do work on your own amplifiers, there are a few resources on the internet which offer schematics for many different makes and models.      A few of those include Dr Tube, at www.drtube.com/guitamp.htm, the Free Information Society, at www.freeinfosociety.com/media_index.php?cat=13&subcat=10&start=0, and up until recently Larry’s Schematic Heaven, at www.schematicheaven.com.

Unfortunately, Schematic Heaven seems to have finally made it to heaven, as the website has been down for several weeks now. Fortunately, a new resource for schematics has been opened, and will hopefully be able to reintegrate all of the schematics which were formerly housed on Schematic Heaven.

The new site is part of www.amprepairparts.com and so far has cataloged all of the Fender schematics which were formerly available from Schematic Heaven. The Fender schematics are found at www.amprepairparts.com/fenderschem.htm. There are also a few other schematics available on the site, and the company has promised to try and catalog all of the schematics which were formerly on Schematic Heaven within next few weeks.

If you work on amplifiers, you know how vitally important schematics can be. They are the road maps of the circuit design. Yes, simple problems can be resolved without the use of schematics, especially if the technician is familiar with the particular amplifier … much as you don’t need a road map to drive around familiar roads near to where you live. But if the problem is difficult or unusual, or the technician is not familiar with the amplifier, having the correct schematic is as vital as having that road map for a road trip.

One last thing: www.amprepairparts.com also carries a nice selection of parts for amplifier repair, and the folks there are very helpful. It’s handy to be able to locate the correct schematic for the amp under repair, and then purchase the parts you need from the same website.

Until next time, I’m Frank the Tech Guy. Keep on fixin’ those amplifiers!

 Posted by at 1:50 am