Posts Tagged ‘Theorie’

Comparison of the Frequency Spectra and the Frequency Decay between Telecaster & Stratocaster

In electrics, Theorie on April 14, 2017 at 12:08 pm

from Johannes Husinsky, October 2002

The project’s goal was to experimentally determine the tonal differences between three different electrical guitars and different pick-ups on one guitar. To do so, I used a computer interface and data acquisition program based on LabView, which computes the frequency spectra using Fast Fourier Transform software.
Connecting a guitar to a computer and picking a string is easy, whereas the analysis of the spectra can be challenging to analyze. First, the frequency spectrum depends on the location of picking the string. For example, if one picks a string exactly in its middle, theoretically just the fundamental is excited; whereas if a string is picked next to the bridge, a high number of harmonics can be observed. Furthermore, if the string is picked on a node of a certain harmonic, this harmonic is not excited.

Second, the output of an electrical guitar also depends on the location of the pick-up. The closer a pick-up is located next to the bridge, the brighter the sound of the guitar will be, whereas a pick-up nearer to the neck will output a more mellow and bassy sound. This is caused by the magnitude of the harmonics amplitudes in different locations on the guitar. For example, the fundamentals maximum magnitude is in the middle of the string and the second harmonics ones are, ideally, on the first and the third quarter of the string. Since the neck pick-up is located closer to those maxima than the bridge pickup, it registers a higher amount of these harmonics than the bridge pick-up does.

Third, picking a guitar with a pick (plectrum) results in a brighter sound than picking a guitar with fingers. The reason is on the one hand the width of the picking element, which causes the string to oscillate, on the other hand the different rigidity characteristics of a pick and a finger. The width of a finger plucking the string is often larger than the width of a pick. This results in a sine wave-like shape of the string, whereas a string picked with a pick generates a shape comparable to a sawtooth wave, which has a higher harmonic content than a sine wave-like one. Furthermore, because of the plasticity of a finger, the high harmonics of the string are absorbed if a string is plucked with a finger. It takes more time until the finger is in is initial shape again,
whereas the pick reaches its initial shape quite quickly.
Set up for the measurements:
Since the voltage output of higher harmonics was very small, I just plotted the frequency spectrum up to the 4th harmonic. To avoid a node of any of these harmonics being over one of the pick-ups, a capo was set on the third fret. In addition, the string (the d-string was used for all measurements) was picked only on positions where harmonics up to the fourth were excited. I made measurements of three different guitars: A Fender Telecaster, a Fender Stratocaster, and an electrical guitar built by Mako.
The guitar built by Mako has a humbucking neck pick-up and a humbucking bridge pick-up, therefore the signal output is higher compared to the Fender guitars. For these measurements, both pick-ups were used together.
Prior to the measurements, Prof. Errede suggested to make measurements on the one hand with the non-picked strings damped, on the other hand with them free, to evaluate, if the vibrating string affects the others and these again re-affect the originally vibrating string. Unfortunately this cannot be seen in the charts, although it is very likely that this happens. Therefore I will only display the charts of the first group.

  • d-string with capo in 3rd fret, picked with finger, other strings were not damped


  • d-string with capo on 3rd fret, picked with plectrum


It can be seen that the higher harmonics are excited moreso using a pick than using one’s finger:


The lack of higher harmonics between 700 and 1400 Hz is due to the location where the string was picked.

Fender Telecaster

The Fender Telecaster has one single-coil neck and one single-coil bridge pickup.
For the following measurements both pick-ups were again used together.

  • d-string with capo on 3rd fret, picked with finger


It can be seen in the frequency decay chart, that the tune of a picked string is initially slightly higher. Whereas the 350 Hz and 352 Hz slopes are initially almost equal, the 350 Hz slope is significantly higher as time goes on.

  • d-string with capo on 3rd fret, picked with plectrum


Fender Stratocaster

The Fender Stratocaster has three single-coil pick-ups, a bridge, a middle, and a neck pick-up. Since the switch allows, among others, to combine either the bridge and the middle, or the middle and the neck pick-up, both switch positions were used for the measurements.

  • d-string with capo on 3rd fret, string picked with finger, bridge and middle pick-up together


  • d-string with capo on 3rd fret, string picked with pick, bridge and middle pick-up together


  • d-string with capo on 3rd fret, string picked with pick, bridge and middle pick-up together


  • d-string with capo on 3rd fret, string picked with plectrum, neck and middle pick-up together


As expected, the frequency spectrum of the neck and middle pick-up is different from the spectrum of the bridge and middle pick-up. Because of the position of the bridge pick-up, the output of higher harmonics is higher than the output of the fundamental, as can easily be seen in the charts, whereas the spectrum of the neck and middle pick-up verifies the bassier sound, since the fundamental has the highest signal output.
Another characteristic of the frequency decay of the Fender Stratocaster can be seen, if the decay charts are looked at. Independent of the pick-up combinations and the way the string was plucked, there is a blip in the decay of the 2nd and the 3rd harmonic which can only be observed at the Fender Stratocaster:


Why different different electrical guitars sound different can be seen in their corresponding frequency spectra. Plucking a string with a plectrum results in the excitement of higher harmonics, whereas the frequency spectrum of a finger-picked string mainly consist of the fundamental and a few higher harmonics. A difference in the frequency spectrum can also be seen for different pick-ups. The spectrum depends on pick-up location. The harmonic spectrum includes only the fundamental and a few harmonics (neck pick-up) or even very high harmonics (bridge pick-up). The audible result of different pick-ups is either a bright and sharp sound (bridge) or a bassy and mellow one (neck). On order to gain a sound comparable to an acoustic guitar, the combination of as much pick-ups as possible is recommended.



Basic electric guitar circuits

In electrics, manufacturing, Theorie on June 12, 2016 at 4:10 pm

(Pickups) by Kurt Prange

Passive (i.e. battery-free) electric guitar circuits are relatively simple and the possibilities for customization are endless. A basic understanding of pickups, potentiometers, capacitors and switches is all you need to get creative and take more control of your instrument’s voice on an electronic level.
Where does the electric guitar signal come from?
Pickups are transducers that convert the mechanical energy of a vibrating guitar string into electrical energy by way of electromagnetic induction. It is a fundamental concept studied in physics and electronics that a changing magnetic field will generate a current through a coil of wire. The electric guitar pickup uses permanent magnets and pole pieces to form a steady magnetic field in the vicinity of each individual guitar string. An opposite magnetic polarity is induced in the metallic (steel core) guitar string when mounted above its respective pole piece and when the string moves, the otherwise steady magnetic field changes accordingly. Wire is wrapped around the poles thousands of times to form a coil within the magnetic field to pick up an induced current and voltage.


The output signal from the pickups is AC (alternating current) because the direction of the current alternates, producing a positive voltage when the string moves in one direction and a negative voltage when the string moves in the opposite direction.
The previous drawing illustrates the electrical and magnetic function of a single-coil pickup. Some pickups might use six permanent magnets in place of the six pole pieces to create the magnetic field, but the idea is the same: create a steady magnetic field around a coil in proximity to the guitar string. The name “single-coil” pickup becomes more significant when compared to the humbucker or “dual-coil” pickup.


Pickups: Single-Coil vs. Humbucker
The first successful guitar pickup was developed in the early 1930’s by Rickenbacker® to help amplify Hawaiian lap steel guitars which were popular at the time. The first pickups were single-coils and while they do a good job of picking up the guitar signal they are also susceptible to picking up interference from nearby electrical devices. The Gibson® humbucker (US Patent 2896491) was developed in the 1950’s to eliminate the “hum noises” resulting from electromagnetic interference. The humbucker uses two coils and a pair of pole pieces (having opposite magnetic polarities of each other) for each string. The coils are wound and connected to each other in such a way that the current produced by the moving guitar string in the two coils adds up (in-phase), while the current produced by electromagnetic interference in the two coils cancels (out-of-phase). Not only does the humbucker drastically reduce noise from interference, but it also has a different characteristic sound. The single-coil pickup is commonly considered to have a thin, clear and bright (more treble) sound, while the humbucker is known to have a full, but dark (less treble) sound with more overall signal output.
Connecting Multiple Pickups
When connecting more than one pickup, it’s important to follow the manufacturer’s color codes and wiring diagrams so that the phase relationship is correct. The phase relationship of a pickup is determined by the winding direction of the coil and the polarity of the magnets. The two coils of the traditional humbucker are connected in series with the phase relationship shown in Fig. 1. Most modern Stratocaster® style guitars with three single-coil pickups are supplied with a reverse wound/reverse polarity middle pickup for a parallel hum canceling effect when the guitar is switched to a two pickup position (e.g. neck & middle pickup together) as shown in Fig. 2.


Pickup Specs
Most replacement electric guitar pickups have limited electrical specifications given on the packaging or on-line which can give you a basic idea of the relative output level and how bright or dark a similar pickup will sound.
• DC Resistance: This can be measured directly with an ohm meter and gives you an idea of how many turns of wire the coil has. If the same gauge of wire was used for two pickups, then the pickup with fewer turns to the coil will have a lower resistance which, in general, makes for a lower output level and a brighter sound.
• Inductance: Inductance is the ability of an inductor (or coil) to store energy in a magnetic field. A higher inductance makes for a higher output level and a darker sound.
• Peak Frequency: This is the frequency beyond which the output level begins to fall dramatically. A higher peak frequency would make for a brighter pickup.
Variety is the spice of tone.
Guitar pickups are a vital component of your tone and replacing them is something that most guitarists can learn to do themselves. Using high quality pickups can go a long way to bringing new life and excitement to your playing experience. There are hundreds of pickup manufacturers and thousands of pickups to choose from. Whether you’re looking for a hotter pickup, trying to capture a beloved vintage tone or seeking single-coil sound in a noiseless package, brands like DiMarzio®, Seymour Duncan®, Lace®, Porter®, Fender®, Gibson® and many others offer a solution.

Potentiometers and Tone Capacitors
What is a Potentiometer?
Potentiometers, or “pots” for short, are used for volume and tone control in electric guitars. They allow us to alter the electrical resistance in a circuit at the turn of a knob.


It’s useful to know the fundamental relationship between voltage, current and resistance known as Ohm’s Law when understanding how electric guitar circuits work. The guitar pickups provide the voltage and current source, while the potentiometers provide the resistance. From Ohm’s Law we can see how increasing resistance decreases the flow of current through a circuit, while decreasing the resistance increases the current flow. If two circuit paths are provided from a common voltage source, more current will flow through the path of least resistance.



We can visualize the operation of a potentiometer from the drawing above. Imagine a resistive track connected from terminal 1 to 3 of the pot. Terminal 2 is connected to a wiper that sweeps along the resistive track when the potentiometer shaft is rotated from 0° to 300°. This changes the resistance from terminals 1 to 2 and 2 to 3 simultaneously, while the resistance from terminal 1 to 3 remains the same. As the resistance from terminal 1 to 2 increases, the resistance from terminal 2 to 3 decreases, and vice-versa.
Tone Control: Variable Resistors & Tone Capacitors
Tone pots are connected using only terminals 1 and 2 for use as a variable resistor whose resistance increases with a clockwise shaft rotation. The tone pot works in conjunction with the tone capacitor (“cap”) to serve as an adjustable high frequency drain for the signal produced by the pickups. The tone pot’s resistance is the same for all signal frequencies; however, the capacitor has AC impedance which varies depending on both the signal frequency and the value of capacitance as shown in the equation below. High frequencies see less impedance from the same capacitor than low frequencies. The table below shows impedance calculations for three of the most common tone cap values at a low frequency (100 Hz) and a high frequency (5 kHz).


When the tone pot is set to its maximum resistance (e.g. 250kΩ), all of the frequencies (low and high) have a relatively high path of resistance to ground. As we reduce the resistance of the tone pot to 0Ω, the impedance of the capacitor has more of an impact and we gradually lose more high frequencies to ground through the tone circuit. If we use a higher value capacitor, we lose more high frequencies and get a darker, fatter sound than if we use a lower value.
Volume Control: Variable Voltage Dividers
Volume pots are connected using all three terminals in a way that provides a variable voltage divider for the signal from the pickups. The voltage produced by the pickups (input voltage) is connected between the volume pot terminals 1 and 3, while the guitar’s output jack (output voltage) is connected between terminals 1 and 2. From the voltage divider equation below we can see that if R1 is 0Ω and R2 is 250kΩ, then the output voltage will be equal to the input voltage (full volume). If R1 is 250kΩ and R2 is 0Ω, then the output voltage will be zero (no sound).


Potentiometer Taper
The taper of a potentiometer indicates how the output to input voltage ratio will change with respect to the shaft rotation. The two taper curves below are examples of the two most common guitar pot tapers as they would be seen on a manufacturer’s data sheet. The rotational travel refers to turning the potentiometer shaft clockwise from 0° to 300° as in the previous visual representation drawing.


How do you know when to use an audio or linear taper pot?
It’s really a matter of personal taste when it comes to volume control. Notice how the rate of change is much more dramatic on the audio taper pot when traveling back from 100% to 50% rotation. This means that the same amount of rotation would give you a more intense volume swell effect with an audio taper than with a linear taper. Using a linear taper volume pot would give you a more gradual change in volume which might feel like you have more fine control with which to ease back the volume level.
For tone control, it’s basically standard practice to use an audio taper. The effect of the tone circuit is not very noticeable until the resistance gets pretty low and you can get there quicker with an audio taper.
How do you know what value of potentiometer to use?
The actual value of the pot itself does not affect the input to output voltage ratio, but it does alter the peak frequency of the pickup. If you want a brighter sound from your pickups, use a pot with a larger total resistance. If you want a darker sound, use a smaller total resistance. In general, 250K pots are used with single-coil pickups and 500K pots are used with humbucking pickups.
Specialized Pots
Potentiometers are used in all types of electronic products so it’s a good idea to look for potentiometers specifically designed to be used in electric guitars. If you do a lot of volume swells, you’ll want to make sure the rotational torque of the shaft feels good to you and most pots designed specifically for guitar will have taken this into account. When you start looking for guitar specific pots, you’ll also find specialty pots like push-pull pots, no-load pots and blend pots which are all great for getting creative and customizing your guitar once you understand how basic electric guitar circuits work.

(Switches and Output Jacks)
Now let’s take a look at how pickup selector switches and output jacks work.
Pickup Selector Switches
Most guitars have more than one pickup and each one has unique tonal characteristics depending on its placement, construction and materials. The pickup selector switch allows the guitar player to choose between different pickups or a combination of them. The pickup placed close to the guitar neck has a warm, smooth tone with more bass content and is frequently referred to as the “rhythm” pickup, while the pickup placed close to the bridge has a sharper, biting sound with more treble content and is frequently referred to as the “lead” pickup. Of course, these are just generalizations. You might find that the neck pickup sounds sweeter for your leads or maybe you get more rhythm crunch from the bridge pickup. The subjective nature of tone is one of the main reasons it’s empowering to be able to customize your own instrument.


People are often confused by the switch terminology of “poles” and “throws”, but it’s actually quite simple. The switch allows us to change the electrical continuity between its terminals. The “pole” is the name of the terminal whose continuity is switched between one or more throws. As shown in the DPDT (double pole double throw) switch drawing above, in position “1” there is continuity between “Pole A” and “A Throw (1)”. In position “2” there is continuity between “Pole A” and “A Throw (2)”. This A-side alone could be thought of as an SPDT switch because it has a single pole with two throws, but because we have an additional B-side the entire switch has two poles with each pole having its two respective throws (i.e. DPDT). The standard Telecaster switch could be considered DP3T because it has two poles with each one having three throw terminals. The standard modern Stratocaster switch adds two intermediate switch positions “2” and “4” (as shown below) where each pole has electrical continuity with two of its respective throw terminals at once.


The standard Les Paul switch is shown below. In position “1” P(A) has continuity with T(A), but P(B) is disconnected from T(B). In position “2” both P(A) and P(B) have continuity with their respective throws T(A) and T(B). In position “3” P(B) has continuity with T(B), but P(A) is disconnected from T(A). The ground terminal is used to connect to the common ground along with the potentiometers, output jack and the bridge in order to eliminate popping and buzzing noises.


The Output Jack
The output jack allows us to connect the signal from the guitar to an amplifier. The standard guitar output uses a ¼” mono jack having two terminals (as shown below) which make contact with the mono ¼” plug end of the guitar cable. The “tip” terminal is connected to the output signal and the “sleeve” terminal is connected to the guitar’s common ground. This is standard for amps and effects pedals, too.
Wiring Diagrams
It’s easy to find electric guitar wiring diagrams on-line through the websites of guitar and pickup manufacturers. There are also a lot of popular modifications out there that you might like to try out. Once you understand the basics of how these circuits work, you can even get creative and customize an original circuit that suits your style best. You won’t have to feel locked into your standard set up ever again. If you come across a new trick that you think you might like, heat up your soldering iron and try it out.


article from:
Kurt Prange (BSEE) is the Sales Engineer for Amplified Parts in Tempe, AZ. Kurt began playing guitar at the age of nine in Kalamazoo, MI. He is a guitar DIY’er and tube amp designer who enjoys helping other musicians along in the endless pursuit of tone.


Document for print here: Basic electric guitar circuits

The Frequency Spectrum of all Instruments

In Sonorisation, Theorie on May 27, 2016 at 9:49 am

frequency instruments rangeInstrument-Sound-EQ-Chart

The Purpose of Power explained

In Amps, Gears, Theorie on May 20, 2016 at 10:14 am


I cannot resit to publish an excellent white paper 
from Acoustic Image, Rick Jones, which explains the various 
parameters to take into consideration regarding Amps power.
More informations and other white papers are available on the 
Acoustic Image Web site.


“More power, more volume.”

It’s an assumption many musicians make. And it’s partially true.

But, its only one chapter in a complex story. A more thorough understanding of the fundamentals of musical instrument amplification can help you as a player better select and utilize amp and speaker systems. Hence these few paragraphs.

I’ll start by explaining what “loudness” is and what creates it as far as the ear is concerned. I’ll then explain the role that amp power plays in creating “loudness.” Following that, I’ll discuss how amp power affects the reproduction of music signals and then finish with a discussion of the trade-off between increasing amp power and adding more speakers. Finally, I’ll summarize what all this means in the store and on the gig.

What is Loudness?

Humans hear changes in sound intensity caused by the compression and rarefaction of sound waves (think of the ups and downs on a sine wave or the waves created when you drop a rock in a still pond). Interestingly, the sound pressure our ear responds to is a non-linear function— expressed as pressure squared. This is an important aspect of hearing that gives our ears a wide dynamic range and it affects how amp power relates to perceived loudness. Typically, the level of sound is stated as Sound Pressure Level or SPL, which is calculated as a ratio of the square of the pressure level measured to the square of the smallest pressure level that can be heard (aka the threshold of hearing). SPL is measured in decibels (or dB for short), and is calculated using logarithms. If you would like more detail on this subject, check out any book on acoustics or simply google “definition of SPL.”

You have probably seen tables that tell you the SPL of various sounds. For example, the threshold of hearing is 0 dB SPL, the threshold of pain (where permanent damage to your ears happens in a very short time) is 135 dB, typical conversation is 60 dB, and a jet plane taking off is 120 dB. This doesn’t mean that a jet is twice as loud as people talking…it is a thousand times louder. And polite conversation itself is a thousand times louder than the threshold of hearing. Obviously, the ear is both pretty sensitive and versatile.

How does all this relate to amplifier power? Basically, a loudspeaker converts electrical watts (the rating on your amp) into acoustic watts which are in turn perceived as changes in sound intensity (or SPL) by the ear. An acoustic watt measures sound intensity per unit of area. So, when converting from acoustic watts to SPL, the size of the room has to be taken into account. To put this in the real world, the same amp with the same wattage rating will sound much louder in your living room than in Madison Square Garden.

It may surprise you that the typical loudspeaker is pretty inefficient, with a conversion efficiency of a few percent, often less than 1 percent. Fortunately, a small value of acoustic watts will create a high SPL in a moderate sized room. For example, 0.5 acoustic watts will produce an SPL of 100 dB in a 100 cubic meter room (a typical living room) and 4 acoustic watts will produce an SPL of 90 dB in a 10,000 cubic meter room (the size of a largish club).

Given all this, how much amplifier power do you need? The calculations are complicated, so you will have to take my word for this, but let’s assume a loudspeaker that has an efficiency of 1% and the need to produce an SPL of 90 dB in a moderate sized club. That’s a situation that most club-playing musicians face. It turns out that about 0.4 acoustic watts would be needed. So a 40 Watts amp would suffice. Right?

Unfortunately, it’s not that simple. The above calculations are for an average SPL, so an average of 40 watts would be required. And a music signal is far from average. Let’s take a look.

What Does A Music Waveform Look Like?

A music signal has an average level but it also has many peaks in it that correspond to transients coming from the instrument being amplified. A pop or slap from a bass or the sound of the pick on a guitar string can produce a momentary increase is loudness from the instrument. The transient signal typically lasts a fraction of a second and it does little to increase the average loudness. Here’s an example:

sgnThis is several second clip of a piano trio. As you can see, there is an average component to the waveform and there are many transient signals with amplitudes that are considerably higher than the average, in some cases up to ten times.

An amplifier reproducing this complex signal has to instantaneously deliver power to recreate the transients without compressing or clipping them. Otherwise, what results is a loss of fidelity and transparency of the amplified signal. So, in the example mentioned above, the amp outputting 40 Watts to reproduce the average level of the signal (90 dB SPL in a moderate sized club) has to output 400 Watts for a fraction of a second to reproduce the full fidelity of the music signal. If the transient is larger than the assumed ten times, more power is required.

So, one important purpose of power output capability is to reproduce the full dynamic range of the complex signal being amplified without damaging it (or as we put it at AI, to create “power without corruption”).

What If You Need More Volume?

So, if you need more volume, you should get an amp with more power, right? Not necessarily.

There are two ways to get more volume. One is to increase the output power of the amp and keep the speaker the same, the other is to keep the power output the same and add an identical speaker.

So if you double the power output of the amp, the result is 3 dB more acoustic output from the speaker and 3 dB higher SPL. If you double the number of speakers, the result is double the sound radiating area and doubling of the pressure level coming from the speakers. Since SPL is proportional to pressure squared, the output of the system is increased by a factor of 4 or 6dB. So, doubling the output power results in 3 dB higher SPL while doubling the speaker radiating area results in 6 dB higher SPL.

As you can see, adding speakers is a much more efficient way to increase volume in an amp/speaker system.

The Purpose Of Power

So what does this mean to you, the performing musician? At the risk of over generalizing, let me say this: the most important purpose of power is to increase fidelity. If you need more volume, add more speaker radiating area. When shopping for an amp and speaker combination, don’t just go for the power rating of the amp, take into account the speaker configuration that will give you the volume level that is appropriate for your playing situations. Overall, consider the amp as the means for getting the quality of sound you want and consider the speaker as the means for getting the quantity of sound you need.

In considering amp power, you should be aware of the fact that not all manufacturers measure and rate their amps the same way. Some use an rms (root mean square, don’t ask) rating, some use a music power rating, some use a peak rating and some just make up a number. What’s more, some may use a single frequency to measure the power level and some may specify the power over a range of frequencies. Other important factors are the distortion level at the rated power and the load used for the measurements. The resulting difference in power ratings from these approaches can be large. For example, a peak power rating can be more that two times the rms rating and a rating at 5% distortion can be 25 to 50% higher than a rating at 0.5% distortion. So, as a buyer and user, you should be aware of how your amp’s power rating has been obtained and, when comparing power ratings, you should make sure that they are obtained under the same circumstances. Otherwise, you’re comparing apples to oranges.

So, bottom line, how much power do you really need?

The example I gave is a pretty good one: 40 Watts on average will result in adequate SPL in moderate sized rooms for “acoustic” amplification. Allowing for clean reproduction of transients (10X more power) brings you to 400 Watts. This should be an rms power rating at a distortion of less than 3% and should be over the frequency range of 40 to 10k Hz. Having more power than that only improves the transient reproduction for a marginally cleaner sound and doesn’t substantially increase the volume at which you can play. If you need more volume, add more speakers.

A Final Word

Of course, even if you find two different amplifiers with the same power, headroom and distortion ratings and play them through the same speaker and enclosure, they will not sound the same (I did say that this was a complex issue and we will need to deal with this and other matters in future articles). So, ultimately, you will have to use your ears to decide what sounds best to you.

But if you will excuse a commercial moment, here is a simple statement that provides a solid basis of comparison: at Acoustic Image, we custom design our portable, full-featured amplifiers without off-the-shelf modules…to produce a high-fidelity, transparent signal that doesn’t color the sound…and to assure tons of headroom. We rate power output conservatively—and carefully (and tortuously) test ever amplifier to make sure it meets precise specifications before it leaves the shop. Most of all, we use our ears in testing to make sure the amps sound good.

We also equip every combo (and extension cab) with a down firing woofer which makes a relatively small speaker sound louder by increasing its effective radiating area through something called the “dipole effect” (more about that too later).

If you have questions, shoot me an email.

Rick Jones


Diapason, positionnement des frettes et du chevalet

In réglages, Theorie on July 11, 2014 at 10:27 am

Le diapason de la guitare

Le diapason de la guitare est la longueur de la corde vibrante quand on l’a joue à vide. Elle est calculée entre deux points précis : le sillet de tête et le sillet de chevalet. Cette mesure est prise sur la corde de Mi aiguë.

Par exemple sur une guitare acoustique, le diapason varie entre 61 cm et 66 cm.
La tension des cordes augmente relativement à la longueur du diapason (logique), par conséquent les guitares avec un diapason court ont des cordes moins tendues et requierent moins de force dans les doigts.

Comment calcule-t-on le positionnement des frettes ?

frettes1Le positionnement des frettes est calculé grâce à une formule mathématique portant le doux nom de L/17,817 



frettes2Et en clair, ça veut dire quoi ?
Tout d’abord on détermine la longueur du diapason. Pour trouver la longueur de la première case (du sillet de tête jusqu’à la première frette), il faut diviser la longueur du diapason par 17,817.


On prend ensuite la longueur restante et on divise à nouveau par 17,817, nous trouvons alors la longueur de la deuxième case, et ainsi de suite… Ce calcul tombe rarement juste car il faut arrondir à la décimale. Pour éviter d’accumuler les erreurs, il existe d’autres repères. Par exemple on sait que la douzième frette doit se trouver exactement au milieu du diapason.

Sillet de chevalet

doigtPour jouer une note (sauf pour les notes jouer sur une corde à vide), il faut presser la corde contre la touche de manière à ce qu’elle soit appuyée contre la frette, ce qui fait augmenter la tension de la corde.


Cette légère augmentation de la tension entraîne une augmentation de la hauteur de la note, et de ce fait la note n’est plus tout à fait juste. Selon cette logique, la fausseté s’accroît au fur et à mesure que l’on se rapproche de la dernière frette.


Les luthiers doivent donc compenser ce phénomène. Pour cela ils modifient la position du chevalet pour augmenter le diapason des cordes basses. Cet angle du chevalet varie selon les fabricants et l’action de la guitare.



Sur les guitares fchevalet2olk (à cordes aciers), l’angle donné au sillet de chevalet crée un écart de 4,8 mm à 6,4 mm entre la corde de Mi grave et la corde de Mi aiguë. Cela permet de compenser la tension et de rétablir la justesse.

Quand on achète un paquet de corde, le Mi aiguë à un tirant moins important que le Si, le Si que le Sol, et ainsi de suite…


Plus le tirant de la corde est élevé, plus la note est fausse dans les aiguës sans la compensation du diapason. C’est pourquoi les sillets de chevalets ne sont pas perpendiculaires mais légèrement inclinés.




Comment régler une guitare?

La justesse, connue souvent aussi sous le nom d’intonation

Si lorsque vous vous accordez parfaitement à l’accordeur ou à l’oreille, que vos cordes sont neuves, et que lorsque vous jouez un accord, cela ne sonne pas juste, cela provient de la mauvaise position du sillet ou du chevalet.

Comment y remédier ?

Jouez à la 12eme case l’harmonique puis la note frettée,

  • Si la note frettée à la 12ème case est plus aiguë que l’harmonique, c’est que le diapason (longueur de corde vibrante) est trop court, donc il faut allonger le diapason en déplaçant ou en grignotant à la lime, ciseau à bois le haut du devant du sillet du chevalet.
  • Si la note frettée à la 12ème est plus grave que l’harmonique, c’est que le diapason est trop long, donc déplacer ou grignoter le haut de l’arrière du sillet de chevalet.

Notez également que la justesse souffre d’une hauteur trop importante des cordes par rapport à la touche ou de cordes dont le tirant serait trop faible.

Quel est le réglage correct d’une guitare classique?

Voici un exemple de réglage correct/grand confort pour une guitare classique:

Vérifiez que le manche est plat, mesurer avec un réglet métallique de précision à la 12ème case, du dessus de la frette au dessous de la cordes, l’idéal est: pour la mi aigue : de 3mm pour la mi grave: de 4mm

Quel est le réglage correct d’une guitare cordes acier?

Un autre exemple de réglage correct/grand confort:

Vérifiez que le manche est plat, ou très égèrement concave, mesurer avec un réglet métallique de précision à la 12ème case, du dessus de la frette au dessous de la cordes, l’idéal est: pour la mi aigue : de 1.5mm pour la mi grave: de 2mm

Tips for writing Songs

In Solfège, Theorie on March 16, 2014 at 6:51 pm
Have you ever stopped for a moment to imagine just how many songs, in total, have been written? Consider… many thousands of years of songwriting, countless millions of songwriters during that period… there must literally have been billions of songs penned. What aspiring songwriters need to do is stop and ask themselves this question: “What can I do to make my songs stand out from all the others?” In this multi-segment feature, we’ll try to go about answering that question.
Types of Songs
Most songs written in the last one hundred years can be loosely grouped into one of several categories; songs written around a chord progression, songs written around a melody, or songs written around a riff.

Songs Written Around a Chord Progression – A favored method of songwriting by musicians like Stevie Wonder, the concept of writing around a chord progression involves initially creating an interesting series of chords, and then basing the vocal melody on that chord progression.

Songs Written Around a Melody – This is probably the most common method of songwriting for pop writers. The composer starts with a vocal melody, and around that melody creates a chord progression and song arrangement.

Songs Written Around a Riff – The emergence of the guitar as a “lead” instrument helped create this method of songwriting. These songs are born out of a guitar (or other type of instrumental) riff, after which a vocal melody (which often mimicks the guitar riff) and chord progression are added. “Sunshine of Your Love” is a perfect example of a riff-based song.

Let’s examine songs written around a chord progression.

Writing Songs Around a Chord Progression
Writing Better Songs

To begin writing songs based on chord progressions, we first need to understand that each key has a series of chords that “belong” to it (referred to as a key’s “diatonic chords”). What follows is an explanation of how to find out which chords belong to which key.

Diatonic Chords in a Major Key

(Don’t know how to play diminished chords? Here are some common diminished chord shapes.)

The above is an example of the chords in the key of C major. We arrived at these chords by beginning with a C major scale, and using the notes from that scale to create a series of chords that belong in the key of C major. If this flies way over your head, don’t get stressed. It is NOT neccessary to fully understand the above in order to write a great song.

Here is what you should try to bring away from the above:

    • in every major key, there are seven different chords. The order of these chords are: major, minor, minor, major, major, minor, and diminished. The order is the same for whichever major key you are in.
    • the space between each of these chords is as follows: between chords 1&2: tone, 2&3: tone, 3&4: semitone, 4&5: tone, 5&6: tone, 6&7: tone, 7&1: semitone (now we’re back to where we started).

So, you’ll need to memorize this: tone tone semitone tone tone tone semitone, and major minor minor major major minor diminished.

Now you know the order of chords in a major key, let’s figure out the diatonic chords in the key of G major. To get the notes, start with the note G, then follow the tone tone semitone tone tone tone semitone rule. If this is tricky for you, start by finding the note G on your sixth string. Count up two frets for a tone, and one fret for a semitone. Hopefully, you come up with the notes G A B C D E F# G.Now, just tack the chord types from our other memorized list above (major minor minor major major minor diminished) onto these note names, in order, and we come up with the chords in the key of G major. They are: Gmajor, Aminor, Bminor, Cmajor, Dmajor, Eminor and F#diminished. Try using these rules to figure out the diatonic chords in a bunch of different keys. With this knowledge, you as a songwriter now have armed yourself with a powerful tool; a means of analyzing other people’s songs, in order to dissect them, and use some of their techniques in your own songwriting. Next, we’ll analyze some great songs to find out what makes them tick.

Analyzing Brown Eyed Girl
Now that we’ve learned what the diatonic chords in a major key are, we can use this information to analyze some popular songs, and try to figure out why they’re so successful.We’ll begin with an easy and very popular tune, Van Morrision’s “Brown Eyed Girl” (get tab from Here are the chords for the intro and first part of the verse, which comprises a large part of the song: Gmaj – Cmaj – Gmaj – Dmaj. By studying the above progression, we’ll can surmise that the song is in the key of G major, and that the progression is I – IV – I – V in that key. These three chords, the I, IV, and V chords (all of which are major), are by far the most widely used of all chords in pop, blues, rock, and country music. Songs like “Twist and Shout”, “La Bamba”, “Wild Thing”, and many others use these three chords almost exclusively. With this in mind, we can conclude that it is not the chord progression that makes “Brown Eyed Girl” so special, as these chords are used constantly in pop music. Rather, it is the melody, the lyrics, and the arrangement (which includes the song’s very famous guitar riff) which make the tune so distinct.

Analyzing Here, There, and Everywhere
Now, let’s look at a slightly more involved chord progression; the first part of the verse to Paul McCartney’s “Here, There, and Everywhere” (get tab from from the classic Beatles’ album Revolver: Gmaj – Amin – Bmin – Cmaj. This song also happens to be in the key of G major, which we can establish by analyzing the chords. The above progression, when analyzed numerically, is: I – ii – iii – IV (which then repeats). After this part is repeated, the song continues: F#dim – Bmaj – F#dim – Bmaj – Emin – Amin – Amin – Dmaj. (Don’t know how to play diminished chords? Here are some common diminished chord shapes.)Continuing to analzye in the key of G major, the above progression is vii – III – vii – III – vi – ii – ii – V. There is one pesky detail about this progression, though; in the key of G major, the third (iii) chord should be Bminor, when, in this case, it’s Bmajor. This is our first example of a songwriter’s use of chords that fall outside of the major key that he/she started in. Exactly why the above progression works, and sounds good, is beyond the scope of this article, but it is important to note that many songs use chords other than just the seven chords in it’s key. In fact, one of the factors that makes a chord progression sound interesting is it’s use of chords that don’t directly belong to it’s key.

Analyzing Pachelbell’s Canon in D / Basketcase
Lastly, let’s have a look at two songs that have much more in common than you might at first think: Pachelbell’s Canon in D Major. Dmaj – Amaj – Bmin – F#min – Gmaj – Dmaj – Gmaj – Amaj. Green Day’s Basketcase Emaj – Bmaj – C#min – G#min – Amaj – Emaj – Bmaj – BmajAt first, you might think these two tunes couldn’t be more different, right? The chords looks totally different. If you analyze each tune numerically, though, it paints a different picture. Here are the numerical progressions for each, Canon in D major being in the key of D major, and Basketcase being in the key of E major:Canon in D Major

I – V – vi – iii – IV – I – IV – V


I – V – vi – iii – IV – I – V – V

The two songs are almost identical. Yet, they obviously don’t sound anything alike. This is a great example of how different a chord progression can sound, when you alter the way in which it is played. I suggest doing what Green Day may, or may not have done here; try taking the chord progresssion to the verse, or the chorus of a song you like, fiddle with a couple of the chords, change the key, change the “feel” of the tune, and write a new melody with different lyrics, and see if you can’t come up with a completely new song.

With this article, we’ve just started to scratch the surface of analyzing the art of songwriting. For further study, you might want to read writing songs in minor keys.


Writing in Minor Keys
In the previous feature, we examined the basics of writing songs in major keys, and before you tackle Part II of this feature, it’s advised that you familiarize yourself with that aspect of songwriting.Sometimes, the theme or mood you wish to create with a song doesn’t suit the generally “happy” sounds that a major key tends to provide. In these situations, a minor key is often the best choice for your song. Which isn’t to say that a song written in a minor key has to be “sad”, or that a song written in a major key need be “happy”. There are thousands of songs written in major keys that certainly not uplifting (Ben Folds Five’s “Brick” and Pink Floyd’s “Wish You Were Here” are two examples), just as there are many tunes written in minor keys that reflect positive, happy feelings (like Dire Straits’ “Sultans of Swing” or Santana’s “Oye Como Va“). Many songwriters will use both major and minor keys within their songs, perhaps choosing a minor key for the verse, and a major key for the chorus, or vice versa. This has a nice effect, as it helps break up the monotony that sometimes results when a song lingers in one key. Often, when switching to a major key from a minor key, writers will choose to go to the Relative Major, which is three semitones up (or, on the guitar, three frets up) from the minor key the song is in. So, for example, if a song is in the key of E minor, the relative major of that key would be G major. Similarly, the Relative Minor of a major key is three semitones (or frets) down from that key; so if a song is in D major, it’s relative minor key would be B minor.We’ve got lots more to discuss, but before we do, we need to learn what chords we can use in a minor key.

Diatonic Chords in a Minor Key
Writing Better Songs: Part II - Writing in Minor Keys
(Don’t know how to play diminished chords? Here are some common diminished chord shapes.) We have a lot more chord choices when writing songs in minor keys than we do if we’re writing in a major key. This is because we compile two scales to create these chord choices; both the (ascending version of the) melodic minor, and the aeolian (natural) minor scale.It is not necessary to know or understand these scales in order to write good songs. What you need to summarize (and memorize) from the above illustration is when writing in a minor key, chords can be found starting on the root (minor), the 2nd (diminished or minor), the b3rd (major or augmented), the 4th (minor or major), the 5th (minor or major), the b6th (major), the 6th (diminished), the b7th (major), and the 7th (diminished) of the key you’re in. So, when writing a song which stays in the key of E minor, we could use some or all of the following chords: Emin, F#dim, F#min, Gmaj, Gaug, Amin, Amaj, Bmin, Bmaj, Cmaj, C#dim, Dmaj, and D#dim.Phew! Lots of stuff to worry and think about. You might want to keep this in mind too: in most “popular” music, diminished and augmented chords really don’t get used a whole lot. So if the above list looks daunting, try sticking to the plain major and minor chords for now.In many traditional harmony books, you’ll see the above series of chords, accompanied by a diagram that illustrates “acceptable” progressions of these series of chords (eg. V chord can go to i, or to bVI, etc). I have chosen not to include such a list, as I find it to be rather restrictive. Try combining various chords from the above illustration of the chords in a minor key, and decide for yourself which sequences you do, and don’t like, and develop your own “rules”.

Next, we’ll analyze some great songs to find out what makes them tick.

Minor Key Signatures
Now that we’ve learned what the diatonic chords in a minor key are, let’s analyze a few songs. Here is a song with a relatively simple chord progression: Black Magic Woman (made famous by Santana): Dmin – Amin – Dmin – Gmin – Dmin – *Amin* – Dmin* OFTEN PLAYED AS AmajAll of the chords (including the Amaj possibility) fit into the key of D minor (which contains the chords Dmin, Edim, Emin, Fmaj, Gmin, Gmaj, Amin, Amaj, Bbmaj, Bdim, Cmaj, and C#dim). If we analyze Black Magic Woman numerically, we come up with i – v – i – iv – i – v(or V) – i. There are just a few simple chords here, but the tune is very effective – a song doesn’t have to contain ten different chords to be great.

Now, let’s look at a slightly more complex song. Most people will recognize the very famous Eagles tune Hotel California. Here are the chords for the intro and verse of the song: Bmin – F#maj – Amaj – Emaj – Gmaj – Dmaj – Emin – F#maj. By studying the above progression, we’ll can surmise that the song is in the key of B minor (which contains the chords Bmin, C#dim, C#min, Dmaj, Daug, Emin, Emaj, F#min, F#maj, Gmaj, G#dim, Amaj, A#dim). Knowing this, we can numerically represent the chord progression of the song as i – v – bVII – IV – bVI – bIII – iv – V in that key. Hotel California is a great illustration of a tune which more fully takes advantage of all the chords available in a minor key. To more fully comprehend minor keys, and how to write songs in minor keys, I highly recommend analyzing dozens more songs, in the same manner as illustrated above, until you get a better idea of what chord movements sound best to you, etc. Try “borrowing” parts of chord progressions from songs you like, and adapting them into your own songs. Your efforts should pay off in no time, and you’ll find yourself writing better and better chord progressions for your original songs. Good luck!from About.Com Guitar

Balanced audio

In Accessories on March 14, 2014 at 11:53 pm

From Wikipedia, the free encyclopedia

Balanced audio is a method of interconnecting audio equipment using balanced lines. This type of connection is very important in sound recording and production because it allows for the use of long cables while reducing susceptibility to external noise.

Balanced connections use three-conductor connectors, usually the XLR or TRS phone connector. XLR connectors, for instance, are usually used with microphones because of their durable construction, while TRS jack plugs are usually used for mixer inputs and outputs because of their smaller profile.


Many microphones operate at low voltage levels and some with high output impedance (hi-Z), which makes long microphone cables especially susceptible to electromagnetic interference. Microphone interconnections are therefore a perfect application for a balanced interconnection, which cancels out most of this induced outside noise.

If the power amplifiers of a public address system are located at any distance from the mixing console, it is also normal to use balanced lines for the signal paths from the mixer to these amplifiers. Many other components, such as graphic equalizers and effects units, have balanced inputs and outputs to allow this. In recording and for short cable runs in general, a compromise is necessary between the noise reduction given by balanced lines and the cost introduced by the extra circuitry they require.

Interference reduction

Balanced audio connections use a number of techniques to reduce noise.

A typical balanced cable contains two identical wires, which are twisted together and then wrapped with a third conductor (foil or braid) that acts as a shield. The two wires form a circuit carrying the audio signal; one wire is in phase with respect to the source signal, the other wire is reversed in polarity, which is also referred to as being 180° out of phase at all frequencies. The in-phase wire is called non-inverting, positive or “hot” while the out-of-phase wire is called inverting, phase-inverted, anti-phase, negative or “cold”. The hot and cold connections are often shown as In+ and In− (“in plus” and “in minus”) on circuit diagrams.


The term “balanced” comes from the method of connecting each wire to identical impedances at source and load. This means that much of the electromagnetic interference will induce an equal noise voltage in each wire. Since the amplifier at the far end measures the difference in voltage between the two signal lines, noise that is identical on both wires is rejected. The noise received in the second, inverted line is applied against the first, upright signal, and cancels it out when the two signals are subtracted.

This differential signal recombination can be implemented with a differential amplifier. A balun may also be used instead of an active differential amplifier device.

The wires are also twisted together, to reduce interference from electromagnetic induction. A twisted pair makes the loop area between the conductors as small as possible, and ensures that a magnetic field that passes equally through adjacent loops will induce equal levels of noise on both lines, which is canceled out by the differential amplifier. If the noise source is extremely close to the cable, then it is possible it will be induced on one of the lines more than the other, and it won’t be canceled as well, but canceling will still occur to the extent of the amount of noise that is equal on both lines.

The separate shield of a balanced audio connection also yields a noise rejection advantage over an unbalanced two-conductor arrangement (such as used in typical home stereos) where the shield must also act as the signal return wire. Any noise currents induced into a balanced audio shield will not therefore be directly modulated onto the signal, whereas in a two-conductor system they will be. This also prevents ground loop problems, by separating the shield/chassis from signal ground.

Differential signalling

Signals are often transmitted over balanced connections using the differential mode, meaning the wires carry signals of opposite polarity to each other (for instance, in an XLR connector, pin 2 carries the signal with normal polarity, and pin 3 carries an inverted version of the same signal).

Despite popular belief, this is not necessary for noise rejection. As long as the impedances are balanced, noise will couple equally into the two wires (and be rejected by a differential amplifier), regardless of the signal that is present on them.A simple method of driving a balanced line is to inject the signal into the “hot” wire through a known source impedance, and connect the “cold” wire to the signal’s local ground reference through an identical impedance. Due to common misconceptions about differential signaling, this is often referred to as a quasi-balanced or impedance-balanced output, though it is, in fact, fully balanced and will reject common-mode interference.

However, there are some minor benefits to driving the line with a fully differential output:

  • The electromagnetic field around a differential line is ideally zero, which reduces crosstalk into adjacent cables, useful for telephone pairs.
  • Though the signal level would not be changed due to nominal level standardization, the maximum output from the differential drivers is twice as much, giving 6 dB extra headroom
  • Increasing cable capacitance over long cable runs decreases the signal level at which high frequencies are attenuated. If each wire carries half the signal voltage swing as in fully differential outputs then longer cable runs can be used without the loss of high frequencies.
  • Noise that is correlated between the two amps (from imperfect power supply rejection, for instance), would be cancelled out.
  • At higher frequencies, the output impedance of the output amplifier can change, resulting in a small imbalance. When driven in differential mode by two identical amplifiers, this impedance change will be the same for both lines, and thus cancelled out.
  • Differential drivers are also more forgiving of incorrectly wired adapters or equipment that unbalances the signal by shorting pin 2.

Internally balanced audio design

Most professional audio products (recording, public address, etc.) provide differential balanced inputs and outputs, typically via XLR or TRS phone connectors. However, in most cases, a differential balanced input signal is internally converted to a single-ended signal via transformer or electronic amplifier. After internal processing, the single-ended signal is converted back to a differential balanced signal and fed to an output. A small number of professional audio products have been designed as an entirely differential balanced signal path from input to output; the audio signal never unbalances. This design is achieved by providing identical (mirrored) internal signal paths for both pin 2 and pin 3 signals (AKA “hot” and “cold” audio signals). In critical applications, a 100% differential balanced circuit design can offer better signal integrity by avoiding the extra amplifier stages or transformers required for front-end unbalancing and back-end rebalancing. Fully balanced internal circuitry has been promoted as yielding 3 dB better dynamic range, as explained above.


Introduction to Audio

In Sonorisation, Theorie on March 11, 2014 at 8:50 pm

SoundThis beginner-level tutorial covers the basics of audio production. It is suitable for anyone wanting to learn more about working with sound, in either amateur or professional situations. The tutorial is five pages and takes about 20 minutes to complete.

What is “Audio”?

Audio means “of sound” or “of the reproduction of sound”. Specifically, it refers to the range of frequencies detectable by the human ear — approximately 20Hz to 20kHz. It’s not a bad idea to memorise those numbers — 20Hz is the lowest-pitched (bassiest) sound we can hear, 20kHz is the highest pitch we can hear.

Audio work involves the production, recording, manipulation and reproduction of sound waves. To understand audio you must have a grasp of two things:

  1. Sound Waves: What they are, how they are produced and how we hear them.
  2. Sound Equipment: What the different components are, what they do, how to choose the correct equipment and use it properly.

Audio Tutorials

IntroIntroduction to Audio; The basics of sound theory, sound equipment and audio work.
Audio ConnectorsConnections; Audio cables and connectors, wiring instructions, etc.
MicrophonesHow Microphones Work; Basic microphone technology, examples of common types, characteristics, etc.
Using Microphones, How to choose the correct microphone and use it properly.
Audio Mixing DeskSound Mixers; An introduction to sound mixers, from small portable units to studio consoles.
Balanced Audio TechnologyBalanced Audio, How balanced audio works and how to use it in your systems.
Sound Quality, Controlling sound levels and quality.
Audio NoiseNoise Types & Colours, White noise, pink noise, etc.

How to Wire an Unbalanced Microphone To A Balanced XLR Input

In Gears, Public Address, Theorie on March 11, 2014 at 8:23 pm
Sometimes it is necessary to wire unbalanced consumer microphones to a balanced professional input. This article describes how to connect a TS jack to XLR.

Using consumer microphones into professional audio equipment doesn’t often come up, but sometimes it has to happen. The problem is that pro audio gear usually has balanced inputs presented on XLR sockets, while consumer signal sources come presented as an unbalanced signal, usually a 1/4 in or 3.5mm TS jack for a microphone. Grounded consumer sources can cause hum by introducing ground loops into the system, but microphones usually float free of mains earth.First, if the consumer mic jack plug is not molded on, or will be cut off anyway, examine the microphone cable. If the cable has two cores and a shield, then it is a balanced cable wired to an unbalanced jack, and the cable can be wired to an XLR3 connector as a balanced connector as described in the article how to wire a XLR plug. The mic then becomes a regular balanced microphone. If the cable has a shield and just one core, or the plug is molded on and has to be kept, then proceed as follows.

Beware P48 Phantom Power on Professional Balanced Audio Inputs

The first thing to be aware of is P48 power. This is a 48V supply in series with about 6.8k ohms fed to XLR pins 2 and 3 relative to pin 1 (which is usually at ground potential). It is a common way to power professional microphones. Many mixers can be switched in sensitivity between microphone level and line level, but the phantom power supply may be present in both modes unless explicitly switched off.P48 power has the capacity to seriously harm the consumer microphone and must be switched off before a consumer source is connected to a professional input. On mixers there is usually a switch and associated LED to indicate phantom power on, but some items may have the phantom power enabling set in a menu somewhere. Before proceeding be absolutely positive that phantom powering is switched off.

Set The Mixer to Microphone Level Not Line Level

Consumer microphones output at low levels, anything from 25mV for 96dB SPL (loud!) for a sensitive electret down to a tenth of that for a dynamic microphone. The mixer should be set to microphone level sensitivity, and double checked for phantom power set to off. Some mixers automatically disable P48 powering on line level; it can be re-enabled on switching to mic level.

How To Wire a Balanced Consumer Microphone To a Professional Microphone Input


Wire the consumer microphone signal ground to XLR pins 1 and 3 (mixer ground and signal -ve) and wire the consumer signal core to XLR pin 2 (signal +ve).This can either be done as shown in the diagram, with a tip-sleeve line jack socket going via a short section of unbalanced audio single-core coaxial cable to a 3-pin XLR plug to go into the mixer.Alternatively the consumer mic jack plug can be cut off. If the microphone cable has a single center core and shield then it can be wired as follows:

  • XLR pin 1 and XLR pin 3 to microphone shield
  • XLR pin 2 to microphone center

it is a good idea to MARK the XLR connector as unbalanced, and a warning about the danger of phantom power!

Alternative Methods of Wiring Unbalanced Consumer Signals to a Balanced Professional Audio Input

An audio transformer can be used to interface the unbalanced microphone to the balanced input, however the cost of a good transformer can be prohibitive, and cheap audio transformers can impair the sound quality.Unbalanced equipment at line level can be interfaced to balanced inputs using a direct-inject (DI) box. If one is handy, it can be worth trying with a microphone but some types of DI box will be noisy with the low impedance weak microphone level signal.A different technique is needed if the requirement is to wire a balanced microphone to an unbalanced consumer microphone input.

from Richard Mudhar


In Gears, Sonorisation, Theorie on March 7, 2014 at 1:05 am

Les décibels sont l’unité par excellence dans le monde l’audio. Celle- ci permet d’exprimer le rapport entre deux grandeurs, entre une grandeur et une référence, un gain en tension, en puissance,… Les décibels résultent du logarithme du rapport de deux grandeurs.

Cette unité a été créée dans le but de simplifier les calculs, ou plus simplement de comprimer l’énorme étendue de l’échelle des intensités sonores audibles par un être humain par exemple. En effet, celle- ci s’étend de 10-12 W/ m² à 10² W/ m²…. Le rapport entre ces deux valeurs est donc de 1 à mille milliards!

En décibels, cela donne une échelle de 0 à 140 dB, 0 dB correspondant au seuil d’audition et 140 dB au seuil de douleur. Largement plus simple d’utilisation que nos W/ m²!

Et lorsqu’on se base sur une grandeur de référence, cela va permettre de calculer une valeur absolue. Nous verrons ci- dessous quelques références. Mais avant d’entrer dans le monde de l’audio, voyons à quoi correspondent les logarithmes.

* * *

Les logarithmes

Progression arithmétique

 Voici une suite de nombres dans laquelle chaque terme est obtenu en additionnant au précédent une valeur constante. Un exemple:

 0 => 1 => 2 => 3 => 4 => 5 => …

 Dans ce cas, chaque terme est égal au précédent + 1.

 Progression géométrique

En progression géométrique, chaque terme est obtenu est multipliant le précédent par une valeur constante. Un exemple:

 1 => 10 => 100 => 1000 => 10 000 => …

Chaque terme est donc égal au précédent multiplié par 10.


Lorsqu’on fait correspondre les deux progressions en alignant le 0 de la progression arithmétique au 1 de la progression géométrique, cela donne ceci:

géométrique 1 10 100 1000 10 000
Progression arithmétique 0 1 2 3 4

Le logarithme d’un nombre de la progression géométrique sera donc le nombre correspondant de la progression arithmétique.

Effectivement, le logarithme de 1 correspond à 0. Le log 10 à 1, le log 100 à 2,…

Remarque: Seuls les nombres positifs possèdent un logarithme.

Quelques formules:

Log A . B = Log A + Log B

Log A/B = Log A – Log B

1. Les dB et les logarithmes

Le décibel (dB) est égal à 10 fois le logarithme en base 10 du rapport de deux puissances :

 n dB= 10 log P2/P1

 Ce calcul s’applique également au calcul de l’intensité (W/ m²)

 N dB= 10 log I2/I1

 Lorsqu’il s’agit de tension, le décibel est égal à 20 log V2/V1.

Ce calcul s’applique également au calcul de pression acoustique (Pa) :

 20 log p2/p1

 2. Les dB acoustiques 

– dB SPL: Il s’agit tout simplement de la mesure de la pression sonore, ou tout simplement du niveau sonore.
Niveau provenant d’une enceinte par exemple, niveau de bruit dans une rue,…
0 dB étant considéré comme le seuil d’audition et 120/ 130 dB comme le seuil de douleur.

On parlera aussi de dBA ou de dBC. Il s’agit simplement de pondération permettant la mesure du niveau de pression sonore selon certains critères.
Le dBA par exemple prend en compte la sensibilité de notre oreille qui est différente suivant la fréquence.

3. Les dB électriques

– Les dBm: L’existance du dBm est dûe à l’apparition du téléphone. Les lignes étant au départ d’une impédance de 600 ohms, 0 dBm correspond à un signal de 0.775 volt soit une puissance dissipée de 1mW.
0 dBm correspond donc à 0.775V (pour autant que l’impédance de charge soit de 600 ohms).
(Remarque: avec le temps, cette notion de 600 ohms a “disparu”. On garde alors 0 dBm = 0.775V qque soit la charge).

– Les dBu et dBv: En “oubliant” cette histoire d’impédance, les dBu et dBv sont apparus. 0 dBv = 0 dBu = 0.775 volt.

– Le dBV: 0.775V n’étant pas forcément le plus facile à manipuler lors de calculs, le dBV (grand V !!!) est apparu. 0 dBV correspnd à un niveau électrique de 1 volt.

0 dBm = 0 dBu = 0 dBv = 0.775 V
0 dBV = 1 volt

– Les dB fs: La notion de dB fs est utilisée en numérique. Les signaux dépassant le niveau de 0 dB fs seront écrêtés. Ce qui veut dire que ces signaux ne pourrant pas être échantilonnés…

Quelques calculs:

La tension mesurée à la sortie d’une table de mixage est de 1 volt.

Quelle est sa valeur en dBu et dBV?

En dBu, la référence est de 0,775 V pour 0 dB. Le calcul est le suivant: 20 Log 1/0,775 soit 2 dBu environ.

En dBV, la référence est de 1 V pour 0 dB. Le calcul est le suivant: 20 Log 1/1 soit 0 dBV.

A l’inverse, on connaît la valeur en dBu et on souhaite connaître la valeur en volt. La formule est la suivante: [ 10^ (x/20) ] * 0,775

10^ correspondant à 10 exposant.

Pour une valeur de 6 dBu, cela correspond à une tension de [10^ (6/20)] * 0,775  soit (10^ 0,3) * 0,775 = 1,55 V.

En dBV, une valeur de 6 dBV correspond à [ 10 ^ (6/20)] * 1 soit 2 V.

Explications sur les logarithmes issues de « Rappel de mathématique » de Dr Sc. B. Mahieu.

Didier Pietquin © 2006, Mise à jour 2008

reference: décibels