TalkBass TechTalk

 

Written by Joris van den Heuvel, TalkBass name "Joris", e-mail: Joris.bass@planet.nl

 

This document is a vast collection of technical facts a bass guitar player might want to know about his or her amplification equipment. It does not cover the bass guitar itself.

 

Publication of this document - or parts of it - prohibited without prior written permission of the author.

 

Parts printed in blue are being edited.

 

1.      Types of bass amplifiers

 

Below is a schematic of a basic bass amplifier system in its simplest form. All setups incorporate these subsystems.

 

bass

-

preamp

-

tone controls

-

master volume

-

power amp

-

loudspeaker

 

1.1.            Combo

 

The most commonly used type of bass amplifier is the combo. One unit contains all of the above subsystems:

        Preamplifier, also known as input amplifier, often with gain knob. The term "preamplifier" is often used to describe both the preamplifier and the signal processor. Technically, these are separate parts. See 2.1

        Signal processor: tone knobs or equalizer, effects, master volume knob. See 2.2

        Additional features may of may not be built in: effects loop (2.3), compressor (2.7.1.2), tuner output, DI output (2.4).

        Power amplifier. See section 5

        Loudspeaker cabinet. See section 6

A combo is the least flexible way to amplify a bass. If you don't like one specific part about it, you have no choice but to replace the whole unit.

 

1.2.            Head

 

A head is everything that's in a combo, except for the loudspeaker cabinet. A head may have its own casing or may be in a 19" chassis, leaving it fairly unprotected. If you play gigs on a regular basis, it would be wise to put it in a 19" rack case for protection.

 

1.3.            Cabinet

 

Obviously, when you want to use a head, you'll need a separate loudspeaker cabinet. The cabinet is connected to the head with a loudspeaker cable (see 7.3), which differs from a signal cable in many ways. Some cabinets have a built-in rack space, in which you can put a 19" chassis head. You could say you built your own combo. But with one advantage: you'll be able to interchange parts.

 

1.4.            Rack system

 

One step further. Preamplifier/signal processor and power amplifier are separate units mounted in a rack case, stacked on top of one or more loudspeaker cabinets. Hence the nickname "stack". Countless combinations exist, and components can be added to suit individual needs. Signal processors, effects units, tuners, power conditioners, multiple power- and preamplifiers, compressors, crossovers, you name it. Often functions are combined into one unit, like with multi effects processors, which could incorporate a compressor, equalizer, chorus, overdrive, delay, reverb etc. More on this in section 2.

 

2.      Signal processing and effects

 

2.1.            Preamplification

 

The first thing in every bass amplification system is the preamplifier. It takes the signal from the bass, which is low-current (in the 10 μA range), low-voltage (in the 100 mV range), and boosts it to medium-current (apx. 10 mA), medium-voltage (apx. 2 V). This is necessary to keep noise and interference minimal. The term "preamplifier" is often used to describe the unit that houses the preamplifier and the signal processor. Technically, these are separate parts. But in case if a tube preamp, the preamp section itself has its own "sound", so an important part of the tone shaping takes place there. Some transistor amps may have an input section with a saturation circuit, to simulate the characteriscs of a tube preamp.

 

2.2.            Equalization

 

This is where part of the tone shaping takes place. The term "equalization" originates from the PA field where equalizers are used to obtain an even frequency response in a given room. Because every room is different and sounds different, a device is needed to correct these differences, in other words: to equalize the response. For bass guitar use, the device should really be named "tone controller" instead of equalizer, but that aside. The working principle is simple: certain frequency ranges can be cut or boosted. Mainly three types exist:

        Tone controls: rotary knobs labeled "bass", "mid" and "treble" control fixed frequency ranges. Other labels may exist and more knobs may be present. Some equalizers have "pre-shape" or "tone matrix" or other controls. In essence they are equalizer presets. They preset an equalizer setting that has proven to be the preference of many bass players.

        Graphic EQ: sliders labeled with frequencies control cut and boost (aka "gain") of the frequency range with the labeled frequency as a center. Each slider controls a socalled "band". The frequencies are fixed. As is the width of each band ("bandwidth").

        Parametric EQ: this type lets you control all aspects of each band through separate controls. Frequency, bandwidth (also designated "Q"), and gain. Some parametric equalizers have additional shelving bands. They control everything below (low shelf) or above (high shelf) their center frequency.

 

2.3.            Effects loops

 

When using a head or combo together with additional effects pedals or rack units, it is desirable to first have preamplification and equalization, then the effects kick in, and then the whole thing is power amplified. Heads and combos don't have a separate preamp and power amplifier unit, in between which you would connect your effects. Instead, many of them provide an "effects loop", which consists of an output (send) and an input (return). The signal from the preamp is tapped and sent out, in other words you "send" it to your effects, and then "return" it back to the power amplifier section. Some heads and combos have switchable signal levels, usually you have a choice of 10 dB (line) or +4 dB (pro/studio). A rule of thumb is: pedals use 10 dB and rack units use +4 dB, but some rack units are also switchable. Always choose the higher level, but everything has to match.

 

2.4.            DI or Direct Inject

 

In order to connect your amplifier to a PA or a studio mixer, a DI is used. A DI device (which is often built into the amplifier) provides a level adjusted, electrically isolated, balanced connection. Signal levels and balanced signals are discussed in 7.1. The electrical isolation is a way to prevent ground loops and related dangers. With many different devices, some with very high power consumption, connected to the mains system in a building, voltages appear across the mains wiring. These voltages contaminate the ground (zero volts) level, for instance when a signal cable (very low power) goes from the main mixer to the bass amplifier on stage, along with the ground conductor from the mains, where the power amplifiers get their juice from (very high power). This leads to a difference in ground level, and this is called a "ground loop". A very loud, system-wide hum is the result. The electrical isolation has the additional advantage of safety. When something blows, chances of you getting mains voltage across your body (and your equipment) are reduced.

 

2.5.            MICing to a PA

 

Of course, YOUR bass sounds the way YOU want, played through YOUR amp (else you wouldn't have bought it, duh!). You want your audience to hear YOUR sound. You could use a DI to go directly to the mixing table. This is a very reliable way to get your signal to be amplified by the PA. But A DI taps the signal before it reaches your amplifier, so your sound stays with you only, as the audience only hears the sound of the bass without your amplifier. Putting a microphone in front of your amplifier is the only way. But there are drawbacks. Mics also have a sound of their own; they alter the sound. And in general, 1 close mic is used for your whole rig. That single mic hardly captures the sound (that doesn't mean it can't sound good, though). A 2x10 bass cabinet has, say, 2 10 inch speakers, a horn tweeter and a bass port. To capture everything, you would have to mic 2-3 feet away, but the influence of other musicians' sounds may seriously affect sound quality, and feedback from the PA can easily occur. Or use 3 mics close up (one for one of the speakers, one for the horn and one for the port). If you put a 1x15 underneith, get ready for 2 more mics (one for the speaker and one for the port). Only pros will have these kinds of demands, as it takes a lot of time to sound check the rig this way. Time you don't have at a medium gig.

 

2.6.            Analog/digital

 

The processing of signals can be done in two ways: analog or digital. Real-world signals are analog, so processing them with analog electronics is the most obvious, and before digital signal processing (DSP) was possible, the only way. When DSP became available, possibilties grew immensely, and are still growing, because the limit is determined only by the speed of the applied digital processor(s), while computing speed doubles every 18 months nowadays. Digital processing requires highly complex designs, and still some analog circuitry, because the real-world signals have to be converted to digital signals (sampling or ADC) first, then processed by the actual processor, and afterwards converted back to analog (DAC). ADC, in short, is meauring the signal voltage many times per second, too fast to be descerned by the human ear, and describing each measurement with a number. DSP, in short, is applying complex math to the acquired digital stream. DAC, in short, is putting all the digital values back in sequential order, while each number represents a voltage.

 

Charateristics of each type are in the table below:

 

Analog

Digital

Extremely simple designs are possible to achieve the end result.

Designs require a minimum level of (high) complexity.

More complex designs generally require more electronics.

More complex designs hardly need more electronics. The opposite seems to be true: as science progresses, systems become more compact, but with improved capability.

Susceptible to noise and interference

Once digital, much less noise is added, but strong interference may lead to hard failure.

Capabilities are limited

Only the designer's imagination and the current computing speed limit capabilities. There's a trade-off between sound quality and capability.

 

2.7.            Signal processors and effects

 

2.7.1.        Dynamics based

 

These are signal processors that, in one way or another, react to and/or alter the playing volume. The basis of all dynamics controllers is the envelope follower. What it does is take a look at the input signal, and then output a signal which perfectly tracks the volume of the input. With that signal the actual effect is controlled, as if you were turning an effect knob while you're playing. In advanced multi-effects processors it can be found as a separate module with its own controls, and can be linked with different effects, like a wah-filter or a phaser.

 

2.7.1.1.    Noise gate

 

As the name implies, it's main use is prevent noise from being heard. This is accomplished by muting the sound when the volume is below a certain level (called "threshold"). The idea is: if you're not playing, then why listen to the internal processes of your equipment? (Hmm, sounds like a good idea after a good Mexican meal ;-) ) More sophisticated noise gates may have several controls:

 

        Threshold level: the volume level at which the gate opens

        Attenuation: how much the volume is being cut when below the threshold level

        Attack: the time it takes for the gate to go from the attenuation value to fully "on"; prevents popping

        Hold time: the minimal "on" time; prevents constant opening and closing of the gate, for instance when playing short, separate notes.

        Decay or release: the time it takes for the gate to go from "on" to the attenuation value; prevents popping

 

Note: a noise gate can't block all noise. If the power amplifier section puts noise on the speakers when no signal is present, a noise gate can't suppress that noise, because a power amplifier can't be muted. Also, when there's noise while you play, obviously the noise gate won't have any effect on that.

 

2.7.1.2.    Compressor

 

This device is used to make volume changes less dramatic. It gradually turns down the volume when playing louder and resets it when playing with less intensity. Controls are:

 

        Theshold: the volume level at which the compressor will engage.

        Compression ratio: how much the volume is cut. Say you have a given rise of input volume. A 2:1 ratio will increase the ouput volume only half of that, as soon as the volume level exceeds the threshold level. A 4:1 ratio will only increase the output for a quarter of that.

        Attack: the time it takes for the compressor to adjust the volume. With a larger attack time, you can keep sudden peaks in the signal. For slap playing this may be desirable.

        Release: the time it takes to return to the normal volume.

        Gain: an extra gain stage is used to bring back the compressed signal to counteract the cut volume.

 

2.7.1.3.    Limiter

 

A variation of the compressor. A limiter usually has a fixed high compression ratio, a high threshold, and a very fast attack and release. Its purpose is to avoid very high peaks that would, for instance, overload an amplifier or recorder. Professional amplifiers have this feature built-in. An advanced compressor can be used as a limiter with the proper control settings.

 

2.7.1.4.    Envelope filter

 

A filter changes its frequency at command of the envelope follower. The most common type is the touch-wah, which is an envelope-controlled low-pass filter with a peak just before cut-off. When playing softly, the wah is closed, giving a very muffled sound, and playing louder gradually sweeps the wah to a higher cut-off frequency, giving a more "ah"-type sound. The touch-wah is best used with very dynamic playing, causing the device to continuously shift from low to high frequency cut-off and back.

 

2.7.2.        Time based

 

A lot of different effects can be obtained by recording a signal, storing it for a period of time, and then playing it back. The recording time can vary from .001 seconds to a few seconds, or actually virtually infinite, in which case you have a recording device - hardly a sound effect. With very short recording times, time based effects begin to overlap with frequency based effects.

 

2.7.2.1.    Delay

 

Back in the early days, delay machines consisted of a tape recorder with a circular (i.o.w. endless) tape. The signal was recorded onto the tape, spun round and played back just before being recorded again. Delay time could be varied by changing the tape speed or length. This type of delay gave a characteristic sound, mainly due to saturation (overdriving) of the tape, comparable to tube saturation. For some time, solid state analog delays have been around. They apply a socalled bucket brigade delay (BBD). An analog memory array of up to a few thousand cells is constantly cycled. Compact chorus and flanger pedals still use this technology.

 

Repeating, fading echoes can be created by feeding the attenuated output of the delay back into the input ("feedback").

 

A delay is the most basic application of a digital processor. The first fully digital sound processors were delays (Lexicon were the first). A simple delay doesn't require a DSP, which were incredibly expensive at that time. Due to the nature of digital systems, storing and recalling information without loss is a piece of cake. If we simply continuously record, store, and playback, we have an outstanding delay. Maybe too good. That's why nowadays, modern delays have some sort of built-in signal degradation to make them sound more like good-ol' tape machines, only 10 times as cheap, and with much less noise

 

2.7.2.2.    Reverb

 

A reverb creates the sense of room acoustics and reverberation. The spring reverbs of old are still quite popular, because of their low cost. More modern reverbs use digital technology: the signal is input in a mathematically created virtual room, and the sound of the "room" is sent out. Before life-like reverbs like these were possible, the only way to get the desired reverb was to actually record in a room that had that desired sound. Many close harmony groups recorded their LPs in bathrooms.

 

Digital reverberation requires very complex mathematical functions to be performed by the DSP. In the table above, there was mention of "a trade-off between sound quality and capability". Because reverb requires a fast and accurate DSP, only costly reverb units that meet those requirements, provide life-like reverb. A good example is the legendary Lexicon PCM-90, which uses all of its processing power, just to create reverb. Inexpensive units, and especially most multi effect units, may sound artificial and cold, and may lack definition.

 

Reverb is rarely used for bass.

 

2.7.2.3.    Chorus / flanger

 

A chorus is a short delay (5 50 milliseconds) with its delay time slightly varied over time ("modulation"). This creates a light shimmering effect, because the pitch of the delayed signal is constantly changing. The delay creates a doubling effect, while the modulation makes it seem as if two different instruments are playing the same notes, instead of just one original and an exact copy.

 

A flanger is essentially the same device, but part of the delay output can be fed back into the input (just like a normal delay), causing a phaser like sound. At extreme settings, a flanger can sound like you're playing in a pool or a rotating tunnel. When the feedback control of a flanger is set to 0, you have a chorus device.

 

Flangers used to be created by applying finger pressure to the flange of the reel of a tape delay machine. Hence the name "flanger".

 

2.7.3.        Frequency based effects (filters)

 

These effects alter the frequency characteristic of the sound they process. Below is a detailed explanation of many different kinds of basic filters and variations. In the graphs, both the gain and the frequency axis are logarithmic.

 

Low pass. The most basic of all filters is the low pass. It rolls off all frequencies above its cut-off frequency with a certain slope. In today's electronic music, this filter, with a steep slope, is often used on the total mix to create a kind of varying fidility.

 

 

 

 

 

 

 

High pass. The inverse of the low pass is the high pass. Mostly used to keep unwanted low frequencies out of a signal. Very low frequencies can't be heard, but can damage loudspeakers and amplifiers, as they will effortlessly try to reproduce them.

 

 

 

 

 

 

 

Band pass. A band pass filter only puts out a narrow range of frequencies. The loudest frequency is called the "center" frequency, the sharpness of the top is called "Q", and the rate at which the surrounding frequencies fall off is called "slope".

 

 

 

 

 

 

 

Band stop /notch. The band stop filter cuts a certain frequency range. This is not the inverse of the band pass, but a low pass and a high pass together. The notch filter is a relative of the resonance filter. It's a special type of band stop, and filters one frequency or a very narrow band out of the signal. For instance to radically cancel hum a 60 Hz notch filter can be applied.

 

 

 

 

 

 

Resonance. The purpose of the resonance filter is to isolate one frequency or a very narrow band. An example of this filter is the tuning section of a radio receiver. You (usually) only want to hear one station, and isolating its frequency is therefor necessary.

 

When applied on an audio signal, a resonance filter sounds like a phaser, but less apparent.

 

 

 

 

 

High shelf. Often parametric equalizers have additional shelving filters. Much like a high pass filter, a high shelf deals with everything above a certain frequency, but doesn't cut everything. There's a minimum gain, which is the actual gain setting. A high shelf can also be set to boost the high range, but this is hardly ever needed.

 

 

 

 

 

 

Low shelf. The inverse of the high shelf.

 

 

 

 

 

 

 

 

 

2.7.3.1.    Equalizer

 

Although not really an effect, the creative mind could use it as such. It is described in section 2.2. The graph shows different settings:

 

Solid lines each band separately boosted.

Dot/dash line all bands boosted (bands influence each other)

Dashed line each band separately attenuated

Dotted line all bands attenuated (bands influence each other)

 

 

 

2.7.3.2.    Phaser

 

A phaser consists of a socalled comb filter. The graph clearly shows how it got its name. You could think of it as a short delay (0.1 2 milliseconds) which causes ripples in the frequency characteristic, when mixed back with the original sound. You can create phasing effects while moving your flat hand closely towards your mouth while making an "FFFFFF"-like sound. Because of the delay-like setup in a phaser, it falls in between the time based and frequency based effects. Digital effects processors will often create a phaser with a slightly modulated delay (like a flanger), analog phasers almost always use a comb filter (which is actually a delay. In fact, every filter is a delay, but an explanation is beyond the scope of this document).

 

2.7.3.3.    Wah-wah

 

A foot controlled steep low pass filter, sometimes with adjustable resonance. This resonance is put just before the cut-off point of the low pass filter, and moves along with that cut-off frequency. The sound of this effect does its name honour.

 

 

 

 

 

 

 

2.7.3.4.    Touch filters

 

An adjustable filter is controlled by an envelope follower. The most common type is the T-wah or touch-wah. It sounds of course much like the wah-wah, but the filter is controlled by playing volume.

 

2.7.3.5.    Triggered filters

 

An adjustable filter is controlled by a trigger, which, in turn, is controlled by playing notes. When triggered (a note is played) the filter opens or closes at a preset rate. In a way a trigger is an envelope follower with just an "on" and an "off" state.

 

2.7.4.        Wave form based effects

 

The last group of effects are the socalled "unlinear effects". They all alter the waveform one way or another, or create a totally different one, controlled by the original waveform.

 

2.7.4.1.    Distortion

 

The most obvious waveform effect is distortion. Originally distortion was created by just turning up an amplifier too far, so the power section ran out of headroom (hit the ceiling) and chopped off (clipped) the tops of the signal. Slight distortion is often described as "growl". Some bass amplifier brands are known for this sound. For electric guitars, usually a lot more than a growl is sought for, and hence clipper circuits are used. A clipper circuit deliberately chops off most of the signal producing sounds from rumbling to roaring to squeeking to fuzzing. Nowadays, countless amplifiers and stomp pedals exist, each claiming to have their own distinct sound, but they all rely on the same principle: clipping.

 

For bass amps, distortion is often introduced in the loudspeaker cabinet. The drivers have a limited linear excursion (see 6.13), and when driven beyond their linear range, they will distort gradually as volume increases. This is also part of the "growl" mentioned above.

 

2.7.4.2.    Rectifier

 

Another form of distortion is rectifying, a method of converting AC to DC, applied in practically every power supply. The negative half of the signal is flipped over to the positive side.

 

2.7.4.3.    Wave shaping

 

When a transfer function is applied to a musical signal, sounds ranging from brass-like to ring modulation can be created. This is called wave shaping. A lot of different transfer functions exist, and these effects are mainly incorporated in small digital pedal boards. Often it creates a distortion-like sound.

 

2.7.4.4.    Harmonic enhancer

 

Another variation of the distortion. The signal is high-pass filtered at a few kHz, then distorted, and mixed back to the signal. It creates the sense of increased brightness and clarity. It is often applied on female vocals to give them a breath-ey sound.

 

2.7.4.5.    Octave doubler

 

By half-cycle rectifying the signal, its frequency can be divided by two, creating a sound one octave below the original pitch. This is often used to fatten up solos and other parts played high up the neck. This effect is known to overload amplifiers when used on low bass lines. Usually, the effect can't be used on it's own, but can only be mixed in.

 

2.7.4.6.    Pitch shifter

 

While the octave doubler can only divide the input signal's pitch by two, this effect can change the pitch over a large range, usually 2 octaves to +2 octaves.

 

2.7.4.7.    Ring modulator

 

The amplitude (envelope) of the signal is modulated (controlled) with a high pitched frequency. The resulting sound is an unmusical, bell-like ringing, hence the name. Some ring modulator features a controllable frequnecy, be it from an envelope follower, or from a rocker pedal.

 

2.7.4.8.    Synthesizer

 

This is the most radical form of altering the sound of a musical instrument. The pitch of the signal is determined. With this information, a synthesizer is controlled, creating its own sounds. Some synthesizers feature MIDI note out capability, so you can hook up MIDI syntheziser modules to create virtually every synth instrument. Ever dreamed of playing the piccolo on a bass guitar? With this kind of stuff you can.

 

 

3.      Necessary physics

 

3.1.            Voltage, current, power and resistance

 

If we're gonna talk about electric basses, some basic knowledge about electricity is essential. Although we could stick to the absolute essentials, we don't. For all electric equipment to operate, a voltage is needed. For example, a wall outlet supplies 120 volts (US) or 240 volts (Europe), and a car battery 12 volts. Once a device is connected (and switched on) current will start to flow, and power is generated. The amount of power is dependent on the voltage of the source and the resistance of the load. Before we start to describe everything in detail, we first we have to make some conventions:

 

 

Symbol

Unit

Unit symbol

Voltage

V

volt

V

Current

I

ampre

A

Power

P

watt

W

Resistance

R

ohm

Ω = Greek capital Omega, rarely the symbol W is used

Frequency

f

hertz

Hz

 

In the equations these standard mathematical operators are used:

xy multiplication

x/y division

xy x raised to the power of y

10log x the 10-base logarithm of x (scientific calculators have this function built in under the "log" key)

 

Right, now we've established that, here we go. The relationship between voltage, current and power is best illustrated by an equation:

 

P = VI Power equals Voltage times Current. Math is easy ain't it?

 

Example 1: What's the current through an electric heater that's 115 volts, 1800 watts? Filling in the equation P = VI gives 1800W = 115VI, I = 1800W/115V = 15.65A (Note: this is very close to the trip point of the circuit breaker)

Example 2: What's the maximum power generated by a car generator, rated 14.4V, 70A? P = VI = 14.4V70A = 1008W. Must be serious car, then.

 

Resistance is another basic aspect of electricity. It determines how much current will flow though a certain load at a certain voltage level. To be more precise: 1 Volt is generated when 1 Ampere flows through a 1 Ohm load. In an equation it looks like this:

 

V = IR Voltage equals current times resistance.

 

Example: What's the current through a 10Ω resistor connected to a 12V battery? V = IR gives 12V = I10Ω, I = 12V/10Ω = 1.2A

 

If we combine the two above equations we get:

 

P = I2R (P = VI, substitute V by IR)

P = V2/R (P = VI, substitute I by V/R)

 

A useful help, when working with these equations, is to put them in triangles, like this:

 

P V P V2

 

V I I R I2 R P R

 

Knowing two variables out of three, you can find the missing one by multiplying or dividing them. If the two known numbers are horizontal, multiply them. If they're vertical, divide the top number by the bottom one.

 

When values become very large or very small, we can decide to put a prefix in front of the unit symbol, to indicate the magnitude of the number. That way we don't have to write down all the zeros that precede or trail the value. The most common prefix is kilo, as in kilogram (kg), kilowatt (kW), kilohertz (kHz). Below is a list of all prefixes and their multipliers.

 

Prefix

Symbol

Magnitude multiplier

Power

Absolute

atto

a

10-18

,000000000000000001

femto

f

10-15

,000000000000001

pico

p

10-12

,000000000001

nano

n

10-9

,000000001

micro

μ

10-6

,000001

milli

m

10-3

,001

centi

c

10-2

,01

deci

d

10-1

,1

-

-

100

1

deca

D

101

10

hecto

h

102

100

kilo

k

103

1,000

mega

M

106

1,000,000

giga

G

109

1,000,000,000

tera

T

1012

1,000,000,000,000

peta

P

1015

1,000,000,000,000,000

exa

E

1018

1,000,000,000,000,000,000

 

Example: What's the current through a 10kΩ resistor connected to a 250mV (line level) source, and how much power is generated?

 

V = IR P = VI

I = V/R P = .25V.000025A

I = .25V/10,000Ω P = .00000625W = 6.25μW

I = .000025A = 25μA

 

3.2.            Calculating resistances

 

Calculating series resistances isnt very hard. You can just add up their values to find the substitute value Rs:

 

Rs = R1+R2+R3+.+Rn (n resistors in series)

 

With parallel connections it gets more complicated: say you have two resistors R1 and R2. Their substitute value Rs will be:

 

Rs = (R1R2)/(R1+R2) (2 resistors in parallel)

 

The following is a more general equation, in which you can put as many resistors as you like in parallel:

 

Rs = 1/(1/R1+1/R2+1/R3+.+1/Rn)) (n resistors in parallel)

 

3.3.            Decibels

 

Due to of the characteristics of our hearing (explained in more detail in chapter 4), sound pressure levels (SPL) are expressed in decibels, symbol dB. Initially, this scale was introduced as the Bel, but the deciBel seemed more practical (1 B = 10 dB). The dB is not a unit, like, for instance, the volt. Rather, it describes the magnitude ratio of two numbers along a logarithmic scale, meaning adding dBs actually means multiplying the numbers. This is in such a way that adding 10 dB means multiplying the sound pressure (and the necessary amplifier power) by 10. Consequently, adding 20 dB is a multiplication by 100, 30 dB is times 1,000 etc. As said before, only power ratios (P1/P2) can be expressed. There's always need for a reference level. For instance, when calculating the maximum SPL of a loudspeaker, the reference level would be the efficiency rating (sometimes called "reference efficiency"). In mathematical terms (told you so, we don't stick to the basics):

 

PdB = PdBref+1010log(P) and back: P = 10((PdB-dBref)/10) Reference in dB

 

PdB = 1010log(P/Pref) and back: P = Pref10(PdB/10) Reference in watts

 

Example 1: you have a speaker with a reference efficiency of 96 dB (1W at 1m), and a maximum input power of 250 watts. The maximum achievable SPL is: SPL = 96+1010log(P) = 96+1010log(250) = 96+24 = 120 dB. So, the speaker produces 96 dB at 1W of input power. This is the reference. We want to multiply the power by 250. This adds 24 dB to the total SPL.

Example 2: an amplifier's maximum output is 300 Watts. You need more power and decide to purchase a 500 watts amplifier. That way you have 500/300 = 1.67 times as much power available. SPL will increase by: SPL = 1010log(500/300) = 2.2 dB.

 

A list of dB conversions:

 

Increase in dB

Power multiplication factor (rounded)

3

2

6

4

9

7.9

10

10

12

15.8

14

25.1

20

100

24

251

27

501

30

1000

100

10,000,000,000

 

It was mentioned before: power equals voltage square divided by resistance (P = V2/R). Because of the square in the equation, signal levels, which are voltages, require another method of determining dBs. With signal levels, a rise of 20 dB (instead of 10) is multiplication by ten. This is how we recognize the squared voltage on a logarithmic scale. The number in dBs increases twice as much, compared to power dBs.

 

VdB = 2010log(V/Vref) and back: V = Vref10(VdB/20) Reference in volts

 

VdB = VdBref+2010log(V) and back: V = 10((VdB-dBref)/20) Reference in dB

 

For audio signal levels, the reference (Vref in the equation) is .775 V. This is a standard value, called 0 dBm.

 

Example 1: An effects pedal accepts a signal level of 10 dB. This is a very common value and is called "line" level. Converting this value to a voltage gives: V1/V2 = .77510(-10/20) = .775 .316 = .25 V

Example 2: A rackmount effects unit accepts a signal level of +4 dB. Again, this is a very common value, sometimes called "studio" or "professional" level. In volts this is: V = .77510(4/20) = .7751.585 = 1.23 V

Example 3: When measuring a test sinewave with a digital voltmeter, the display reads 1.00 volts. Determining the signal level in dBs: VdB = 2010log(1.00/.775) = 2.21 dB

 

4.      The human hearing

 

4.1.            Sound perception

 

Especially for bass players it's important to know how the brain interprets low frequencies. The fact that our hearing is insensitive to low frequencies is somewhat counteracted by the brain. If the brain detects two different pitches, the difference is perceived as a third pitch. If, for instance the two first harmonics of a bass guitar note are heard, but the fundamental frequency of the string is omitted, the brain will "fill in" the missing fundamental. The difference with the real thing is feeling. A very low frequency can be felt, rather than heard. But we do perceive it, even if it isn't there.

 

This is common knowledge to bass cabinet designers. The lowest frequencies are not very important, as opposed to a high sound pressure level. More on this in section 6.

 

4.2.            Sound pressure

 

The human hearing is a piece of art from Mother Nature. It is capable of processing frequencies that are a factor 500 to 1,000 apart, and sound pressure levels that are a factor 10,000,000 apart! There's no audio technology on this earth that can beat the human ears. So you better be careful with them!

 

Techtalk. Because our ears can determine vast changes in volume, sound pressure is measured in decibels, a logarithmnic scale (see 3.3). If you double power (twice the watts) the sound becomes 3 dB louder. However, an increase of as much as 10 dB is perceived as a doubling of volume. This equals ten times the power. So in order for you to play twice as loud, you need an amplifier with 10 times as much power.

 

More on this in section 5.2

 

4.3.            A-weighting

 

Our ears don't have the same sensitivity for each frequency. The lower the frequency, the less sensitive, and the same goes for higher frequencies. The midrange, which is around 1 kHz, is the most sensitive range. This is where most of the human voice is. Scientists have measured the hearing sensitivity for different frequencies of thousands of people. The cross-section of these measurements is called the A-curve. Applying this curve is called A-weighting.

 

4.4.            Hearing damage and protection

 

Playing on a live stage with a band almost always means high sound pressure levels. Especially when there's PA support for the whole band including drums. At close range (3 feet), a medium powered bass amplifier (200 watts) produces an average sound pressure level of around 100 dB, crest factor (see 5.2) considered. Every other musician adds roughly 2 dB to that. The drummer could easily add another 3-6 dB. Then there's the backfire from the PA - especially the bass range.

 

Reason enough to wear ear protection. The simplest form is foam rubber plugs. Squeeze, put in, throw away after use. Cost less than a buck. But often these simple plugs will hurt when worn for long periods of time. And often they seal your inner ear from the outside air. For a little more cash ($10-25) special musician's ear plugs are made that adapt to the shape of your inner ear, often with one or several ventilation channels, allowing your inner ear to breathe and sweat. The most advantageous, and of course the most expensive ($50 and up), are customly molded silicone plugs.

 

5.      Power Amplifiers

 

A power amplifier is in essence an AC to AC power converter. It draws power from the wall socket with a fixed voltage and frequency, and converts it to a variable voltage with variable frequency, exactly mimicing the input signal (to a certain extent). In theory, only the power level will have increased.

 

5.1.            Power amplifier specifications

 

There's much more to a power amplifier than its power rating. On top of that, power can be measured in many ways. Not every company will specify the maximum output power of their amps the same way. There are no laws that dictate how power should be measured. Some will specify max power at 1% THD, some at 10% THD (true for most car amps), some will measure short-term power, others will specify long-term burst power, etc. etc. There's no right or wrong, they're just ways to measure. Often companies that offer low budget equipment choose a method that boosts their figures.

 

Term

Unit

Description

Good

Bad

Sensitivity

dB

Input voltage needed to achieve full output power. Useful for level matching.

-

-

Noise

dB

Noise produced by the amplifier itself with master volume set to 0

-80dB

-50dB

SN ratio

dB

Signal to Noise ratio

100dB

60dB

THD

%

Total Harmonic Distortion; describes how much the output signal resembles the input signal

0.05%

0.25%

IMD

%

InterModulation Distortion; describes how large signals (low frequencies) influence smaller signals (higher frequencies)

0.05%

0.25%

Slewrate

V/μs

The maximum rate at which the output voltage is able to change

50V/μs

20V/μs

Channel

separation

dB

Crosstalk between channels of a stereo amplifier

60 dB

30 dB

Voltage gain

dB

Input-to-output voltage gain with master volume at maximum

-

-

Power consumption

W

Mains power consumption. Typically an amplifier requires twice its output power from the mains when operating at full power .

1.5x output power

3x output power

 

5.2.            Understanding amplifier power

 

5.3.            Minimum load impedance

 

There's only so much an amplifier can drive. It is designed for a specific minimum load impedance. This is not the actual output impedance of the amp itself (discussed in 5.4), but a rating for the load that can be safely connected to the amplifier. This has to do with the maximum output current the amplifier can deliver, as a low-impedance load draws more current than a higher-impedance load. It may be hard to understand why it's called minimum load, but as described above and in 3.1, amplifier load increases when impedance decreases.

 

5.4.            Output impedance and damping factor

 

As with all electricity supplying components, amplifiers have an output impedance. You could look at it as a resistor in series with the output of the amplifier, representing the imperfection of that amplifier's output stage. Directly related to output impedance is the damping factor. This is a measure for the control an amplifier has over a speaker cabinet. A given transistor amplifier may have an output impedance of .02 ohms. If the connected load has nominal impedance of 4 ohms, the damping factor is calculated by dividing the load impedance by the output impedance: 4 / .02 = 200. Not bad. Obviously, the higher this factor, the better. Especially the low bass response will be tighter. This is exactly why it's important that loudspeaker cables should be made of large copper diameter wire. You don't want to lower the damping factor by connecting the speaker cabinet with a thin cable with a relatively high series resistance. The speaker cabinet's internal crossover may also add to this series resistance. The woofer is usually crossed over by one or two series inductors, that have a resistance of their own. However, usually, the performance of the cabinet won't be affected because the designer should have taken the crossover losses into account.

 

Tube amplifiers usually have a much higher output impedance. This is where part of their characteristic sound comes from. Because the amplifier has less control over the connected load, the sound will have one or two pronounced resonance peaks around the port frequency of the cabinet. This is often described as the "warm" sound of tube amplifiers. Other factors are at play here, but this is definitely one of them.

 

5.5.            Frequency response

 

Frequency response is not much of an issue for solid state amplifiers. A typical range is 10 Hz 100 kHz. However, if not explicitly mentioned in the specifications, this is measured at low power. The curve could change significantly when measuring at full power. This is called "power bandwidth". Low frequencies demand more from the power supply capacitors, since low frequencies draw current for longer periods of time than high frequencies. High frequencies demand more from the electronics' slew rate (see 5.1), since fast changes of the output level are required. Both these limitations result - obviously - in distortion, but even then, a good amplifier will perform its duties over far greater frequency range than that of the human ear.

 

5.6.            Distortion, clipping

 

When a power amplifier is forced past its maximum output power, it will clip. This means the output voltage of the amp is equal to the supply voltage from the power supply and can't go any further. The peaks (positive as well as negative) of the signal are clipped off, hence the term "clipping". Clipping in itself isn't bad, neither for the amplifier nor the connected speaker(s). It's the compression of the signal that causes problems. When a signal is compressed (by clipping it) its average power increases, and may be over the amplifier's maximum thermal capabilities. Furthermore, a clipped signal loses low frequency content, due to the fact that low frequencies usually have a larger amplitude. Low frequencies get clipped first, so to speak. This means the high frequency content increases relatively, and this can damage tweeters (horns).

 

Now a more technical story. A side effect of clipping that is often overlooked is the DC decoupling instability. Most amplifiers can't amplify DC signals (why would they need to?), so they are DC decoupled. This has a huge advantage for designers: the output offset adjustment can be done automatically by the amplifier itself, instead of adjusting it by hand at the time of manufacture. This self-adjustment relies on the fact that the amplified signal always has an average value of zero (equal power in both the positive and negative half of the signal). If the signal is non-symmetrical, as is the case with many musical instrument signals, clipping will occur at only one half (positive or negative) of the signal. As a result, the output offset adjustment starts to shift, as the amplifier will keep its output at a zero average. In turn, this causes a very low frequency (that of the adjustment cycle) at the output. This is often visible as a strong waving of the speaker cones. Needless to say, a speaker cone that moves this much, is prone to damage.

 

5.7.            Volume control

 

It's a common misconception to believe the master volume control of the power section to control the output power from 0-100% when it is turned from 0 to 10. The volume control of a power amplifier is an attenuator and thereby attenuates the (obviously) too strong input signal. It gives you the possibility to turn down from maximum power. If the input level (from the preamp/signal processor) is too low, maximum power will not be achieved. If the input level is very high, maximum output power is reached long before the master volume control is set to its maximum.

 

5.8.            Amplifier topologies

 

Due to the nature of loudspeakers, amplifiers have to be designed so that a loudspeaker can move in (negative) and out (positive) the cabinet. In practise, this means separating the amplifier in two halves: one for each half of this movement. A problem occurs when the two halves have to take over from each other. At near-zero current, both tubes and transistors are non-linear. It means they will not reproduce the signal well (aka "crossover distortion"). This problem returns every half cycle of the waveform, as a waveform crosses zero twice each cycle.

 

Another problem occurs when the output devices of the amplifier are not fully "on". Because a music signal is of a constantly changing amplitude, this is practically always the case. The connected load of the amplifier receives part of the output voltage, while the output section gets the remainder of the supply voltage. This remainder is converted to waste heat. When an amplifier is working somewhat below its maximum power, more heat than output power is produced, even if the amplifier is theoretically ideal.

 

There are several ways to address these problems. They're called "classes". Not every amplifier class is suited for audio (there are more purposes for amplifiers). Only those who are, are listed below.

 

        Class A: Maximum current flows through the output stage at all times. This way the near-zero current is avoided, and thereby crossover distortion eliminated. An unavoidable side-effect is, when no signal is present, power consumption is at maximum, and the amplifier will run hot when no sound is produced. Better still, the amplifier will cool down when operating at moderate to high output power.

        Class B: The opposite of class A. No current flows through the output stage when in rest. Stand-by power consumption is nearly non-existent, but crossover distortion is eminent, be it acceptible for some applications (like speech or sirens).

        Class AB: The best of both worlds. A small stand-by current keeps the crossover distortion at a low level, and when silent, power consumption is only a fraction of the maximum power. Nearly all conventional power amps are class AB.

        Class D: As mentioned above, heat is produced when an amplifier output device is not fully "on". Class D amplifiers use digital technology to rapidly and constantly switch the output devices on and off, effectively avoiding the "in-between" state. By filtering (averaging) the switching frequency out of the output, the intended amplified signal appears on the output. This class is a.k.a. switching amplifiers. It won't be before long when every amplifier uses class D topology ('cept for them good-ol' tube amps, but then again, ya never know). When combined with a switching power supply, instead of a conventional heavy mains transformer, weight, mains power and cooling requirements can be drastically reduced.

        Class G: This topology uses two sets of output transistors and two supply voltages. One set controls low-to-medium power signals, keeping power consumption and heat at a moderate level. When high power is needed (during signal peaks), the second transistor set takes over and provides the higher voltages, fed by the higer supply voltage. As soon as the peak is over, the first set gets back to work.

        Class H: Much like class G, this system uses two stages. Only now the supply voltage is temporarily increased (switched) to deal with the peaks. The advantage is: you only need one set of (expensive) output transistors, and the switching can be done by much cheaper electronic switches.

 

5.9.            Transistor or solid state amplifiers

 

Most amplifiers use solid state technology, a.k.a. semiconductors. A transistor is a semiconductor. This means its conduction can be controlled. A small current at the input will result in a higher current at the output, supplied by the power supply. This current gain is the building block for an amplifier. A transistor's current gain is limited. This is especially true for high power transistors. This is why several stages are used to increase the input current about 1 million times and the input voltage about 100 times. Each stage amplifies the signal a bit more, and the final stage drives the loudspeaker. A more recent development is the MOSFET (=Metal Oxide Semiconductor Field Effect Transistor). A semiconductor also, but with a different working principle. Its resistance can be controlled by an input voltage. This makes for more efficient (read: cheaper) amplifier design, because high power MOSFETs have a higher gain and can be driven with less stages. Another development is the IGBT (Insulated Gate Bipolar Transistor): a regular power transistor is driven by a smaller MOSFET, combined in the same housing.

 

In a transistor amplifier, the mains input is first transformed to a lower AC voltage (and at the same time electrically isolated from the mains) by a transformer, then rectified by diodes, and then buffered (to get an almost-perfect DC voltage) by one or more pairs of very large capacitors. These are the energy storage units for when the AC voltage is crossing zero and thus not able to supply current. These large capacitors are of huge importance. With a high current, a small capacitor will easily be drawn almost empty every cycle. The voltage drops and the amplifier will clip more easily. Another side-effect of this is hum, because even when running idle, the supply voltage will drop every rectified half-cycle.

 

5.10.         Transistor amplifier maintenance

 

Apart from keeping the amplifier dust free, there's not much you can do. The main cause of failure of transistor amps is overheating, so cooling air must be able to flow freely. Obviously, overloading an amplifier will certainly overheat it, as the heat sink(s) and, if applicable, the cooling fan(s) weren't designed to be overloaded.

 

5.11.         Tube or valve amplifiers

 

5.12.         Tube amplifier maintenance

 

5.13.         Bi-amping

 

When powers increase, passive crossovers (see 6.6) become very expensive, and even then, power losses are unacceptable. This is the point where one decides to bi-amplify. It means that low frequencies and high frequencies are separated by an active crossover at signal level, and then each frequency range is amplified by a separate amplifier and separate speaker(s). This is especially advantageous for subwoofer setups as a passive subwoofer crossover requires very large and expensive inductors.

 

A schematic of a bass guitar bi-amplified setup:

 

 

 

 

 

 

crossover

high output

-

amplifier

-

cabinet

bass

-

preamp / eq / effects

-

 

 

 

 

 

 

 

 

 

low output

-

amplifier

-

cabinet

 

5.14.         Stereo amps and bridging

 

Most power amplifiers have two channels. The intended application for a 2-channel amp is a stereo setup, as it is desirable to have two absolutely identical amplifiers for left and right. However, amplifying a monaural bass guitar in stereo is hardly ever needed. Several possibilities exist:

 

Setup

Application

Dual mono

The same signal is fed to both amps, each amplifier drives a cabinet

Bridged

Most 2-channel amps provide a possibility to link up both channels into one channel with double power. However, the minimum load impedance (see 5.3) is also doubled

Bi-amped

An active crossover is put in the signal chain; one channel amplifies the low frequencies, the other channel gets the high frequencies (see 5.13)

Stereo

Obviously for use with two (identical) cabinets. Some effects processors create stereo effects from a monaural source, and in this case a stereo setup may be interesting

 

6.      Loudspeaker cabinets

 

6.1.            Loudspeakers

 

Below is a schematic cross section of a loudspeaker:

 

a permanent ring magnet

b back pole plate

c front pole plate

d vented pole piece

e voice coil former

f voice coil inside flux gap or air gap

g basket

h mounting flange

k spider

m spider support

n cone

p dust cap

q surround

r voice coil wires

s cone punch tubes

t connection terminal support

u cone wires

 

 

 

 

 

 

 

And this is how it all works: the permanent magnet (a) provides an extremely strong magnetic field (symbol B, a.k.a. as flux, typically 1 Tesla) inside the air gap (f). The magnetic field is directed there by the pole plates (b/c) and the pole piece (d). The better part of the voice coil (f) is inside this circular slot, and is held there by the tubular voice coil former (e), which is held in place by the spider (k). Electric current through the voice coil will induce a magnetic field, and, by Lorentz' law, a force is the result. The coil puts the cone in motion, which will, with its large surface, move air. The spider's second task is to provide most of the cone suspension (symbol Cms). It makes the cone move back to its resting position. The cone itself is centered by the surround (q), which is usually made of rippled impregnated cloth. The surround also provides a part of the suspension and damps the unwanted surface vibrations of the cone.

 

A close-up of the motor section:

 

In this case, the voice coil is overhung, and the coil overhang (the part of the coil that extends out from the air gap) is the distance the coil can move linearly, give or take 10%. This figure is known as xmax. Once the xmax is crossed, any further movement will be increasingly non-linear, as a smaller part of the coil is covered by the air gap, and thus motor force is gradually reduced. Distortion is the result. When the coil has moved fully out of the front pole plate, motion almost stops, and clipping occurs, comparable to amplifier clipping. With an unproperly constructed spider and surround, the coil former could end up slamming onto the pole piece and front pole plate or onto the back pole plate. This will obviously lead to severe damage of the coil, possibly short-circuiting the connected amplifier. This phenomenon is known as over-excursion or Xdamage (see 6.7.2).

 

 

6.2.            Cabinets

 

Bass loudspeakers can't operate in free air. They need a cabinet to absorb the sound radiated from the back of the loudspeaker, or else it would cancel out the sound radiated from the front. A properly designed cabinet drastically improves the bass response of the loudspeaker. Below is a list of common ways to achieve this:

 

Principle

Description

Closed

A closed system is a cabinet with nothing more than an enclosed air volume, in which the compliance of the driver and the damping of the air interact to improve the bass response.

Open-backed

The driver is mounted in a cabinet, but the back of the cabinet is left open for the better part. This type of cabinet won't provide much bass. Most guitar cabinets are constructed this way.

Ported

This type is defined by the use of a port, a hole in the cabinet wall; often a tube or rectangular channel, extending into the cab, is attached. This port will make the cabinet resonate at a certain frequency. This frequency is chosen just below the point where the loudspeaker's response starts to roll off, effectively improving the bass response. The far majority of (bass guitar) cabinets apply this principle. It has numerous advantages over a closed system. It is, however, much harder to design.

Horn loaded

 

Transmission line

 

 

6.3.            Configurations

 

Bass speakers come in different sizes. Standardised sizes are 8, 10, 12, 15 and 18 inch. Often smaller speakers are used in multiples. Below is a table of common configurations.

 

Size

Common multiples

8"

2 4 8

10"

1 2 4 8

12"

1 2 4

15"

1 2

18"

1

 

A general rule is: the larger the speaker, the lower the bass response, the smaller the speaker, the better the treble response.

 

6.4.            Frequency response

 

The frequency response of a cabinet is the range where it supplies an almost even loudness. The roll-off points are those frequencies where the loudness drops considerably. However, "considerably" is a relative term. Most cabinet manufactorers will specify their limits at -3dB, a clearly noticable drop in volume, which also equals half the acoustical output power. Others may use -6, -8 or even -10dB. This rather specifies the "usable range" of a cabinet: the range at which it produces significant output. Neither way is good or bad, it's just ways to measure. Just like amplifier output power. As a consumer, however, you will have to constantly pay attention to how specifications are obtained.

 

6.5.            Impedance

 

Resistance was discussed earlier in section 3.2. The calculations given are also applicable to loudspeaker impedances. Connecting two cabinets on one amplifier (channel) always means parallel connection, unless explicitly mentioned. Connecting different cabs in series is generally not recommended, as it may give unpredictable results. Identical cabs can be series connected without any problems. A special "series-Y" cable may be needed.

 

Impedance, however, is a different phenomenon. It describes the resistance of a load connected to AC (Alternating Current) instead of DC (Direct Current). Resistance should be seen as impedance at a frequency of 0 Hz. Adding the variable of frequency complicates matters. If a loudspeaker were a simple resistor, its impedance would be a constant throughout its frequency range. But this is not the case. Its impedance changes with different frequencies. Around its resonance frequency (see 6.13) the graph shows a high peak, and with higher frequencies, the impedance gradually rises. The specified impedance is really a nominal impedance, measured at 1 kHz. That's exactly why an ohmmeter will never show the true impedance of a cabinet. An ohmmeter measures (DC) resistance. Typically for loudspeakers, their DC resistance is about 80% of their rated impedance.

 

6.6.            Crossovers

 

There are few speakers that can reproduce the entire audible range. The ones that can are likely to be fit into high-end audiophile systems. Normal people resort to using different drivers optimized for a certain frequency range, and fitting them into the same cabinet. Two-way systems are the most common, as they are relatively easy to design and give good results at low cost.

 

Using optimized drivers means having to split up the frequency range into bands (much like equalizer bands). The separation range is called the crossover frequency. One driver's output rolls off, while the next one gradually takes over. The rate at which this takeover is done is called slope. Slopes are 6, 12, 18 or 24 dB per octave.

 

6.7.            Power handling

 

There are two types of power handling for a speaker. Thermal and mechanical:

 

6.7.1.        Thermal power handling (Pth)

 

The only electrical part of a loudspeaker is the voice coil. It is usually made of thin copper wire on a high-temperature resistant coil former. The thermal power handling (Pth) is the maximum average power the coil can dissipate (heat conduct), before the coil wire or the glue, with which the coil is attached, will melt. Because the rise in temperature due to high power is fairly slow, short, high power peaks (in the order of tenths of seconds) will be averaged by the thermal sloth. A rule of thumb: peak power (Ppeak) equals twice the continuous power. Professional drivers are often able to withstand peaks of four times their rated continuous power. Some even 10 times.

 

6.7.2.        Mechanical power handling (Xdamage)

 

A loudspeaker is an electro-mechanical device. The mechanical part can, just like the electrical part, be overloaded. The cone suspension system is limited in its excursion (movement). Please note: this is not the figure of Xmax, which is the maximum linear excursion of the cone (see 6.1). The cone should be able to move considerably more than Xmax.

 

6.8.            Damage

 

6.9.            Protection

 

6.10.         Efficiency and sensitivity

 

6.11.         Phasing and distortion

 

6.12.         Placement

 

6.13.         Thiele/Small parameters

 

In the past, two engineers, Thiele and Small, have developed a measurement standard for loudspeaker characteristics. The Thiele/Small parameters, T/S parameters in short, are what defines a loudspeaker. These parameters are a list of figures of mechanical and electrical constants:

Para-meter

US

unit

Eur.

unit

Description

Interpretation

Fs

Hz

Hz

Free air resonance frequency

Specifies the the top of the resonance peak of the driver in free air, floating in an infinite space.

Qms

-

-

Mechanical Q factor

These three figures describe how the driver resonates. The bandwidth of the resonance is split into motional resonance (Qms) and electrical resonance (Qes). Their value is combined in Qts.

This value is of great importance to cabinet design.

Qes

-

-

Electrical Q factor

Qts

-

-

Total Q factor

Vas

ft3

dm3

Equivalent air volume

See 6.1

Cms

in/N

mm/N

Suspension compliance

See 6.1

B

T

T

Magnetic field strength

Strength of the magnetic field inside the air gap

BL

N/A

N/A

Force factor

Specifies how much force the voice coil will apply to the cone at a given current

Xmax

in

mm

Maximum linear cone displacement

See 6.1

Xdmg

in

mm

Maximum cone displacement

See 6.7.2

μ0

dB

dB

Reference efficiency

See 6.10

Sens

dB

dB

Sensitivity

See 6.10

Pth

W

W

Thermal power

See 6.7.1

Ppeak

W

W

Peak power

See 6.7.1

 

6.14.         Cabinet DIY

 

6.14.1.     Tools

 

6.14.2.     Materials

 

6.14.3.     Design trade-offs

 

6.14.4.     Measurements

 

6.14.5.     Cost

 

6.14.6.     Literature and software

 

7.      Signals, connectors, cables

 

7.1.            Signal types

 

Signal type

Description

Microphone

Low impedance (150Ω), low level (10mV)

Instrument

High impdeance(1MΩ, low level (100mV)

-10 dB line

Medium impedance (15kΩ), medium level (250mV)

+4 dB line

Medium impedance (15kΩ), high level (1.25V)

 

All of the above signals can be balanced. This means that the signal is split in two, and one of them is inverted (put in counterphase). The original signal is called positive or hot, while the inverted signal is called negative or cold. Both signals are sent across the cable, each through their own conductor. At the reception side, the negative signal is inverted back and both signals are added together, effectively cancelling any noise, hum or interference that both signals may have picked up in the cable.

 

Although not really a signal type as such, this is what an amplifier puts out:

 

Amplifier output

Very low impedance (0.01Ω), very high level (50V)

For instance:

100 Wrms, 8Ω load = 28.3Vrms

600 Wrms, 4Ω load = 49.0Vrms

 

7.2.            Connector types

 

Connector type

Signal type

Application

" jack

signal, speaker

Instrument, speaker

TRS (3-contact " jack)

signal

stage, studio

XLR

signal, speaker

stage, studio, microphone

Cinch

analog / digital signal

home stereo, digital audio, video

Speakon

speaker

professional speakers

Banana / 4mm

speaker

home/semi-pro speakers

Binding posts

speaker

home/semi-pro speakers

5-pole DIN

digital control signal

MIDI

 

7.3.            Cable types

 

signal

A small diameter core, shielded by copper mesh, along with metal foil or carbon screen. Shielding prevents interference.

balanced signal

A conductor pair, shielded by copper mesh, along with metal foil or carbon screen. Shielding prevents interference. Balanced signals see 7.1.

balanced multi signal

Many pairs of conductors, each with their own shielding provide fast and flexible connection of a mixing console to multiple signal sources on a stage.

speaker

Large copper diameter conductor pair, unshielded.

digital

75 ohms coaxial cable of low capacitance. One cable for stereo connection.

digital optical

Plastic light conductor. One cable for stereo connection.

 

8.      Frequencies of note pitches

 

Open string frequencies for an 8-string bass guitar tuned F# B E A D G C F are bold. Other standard string tunings are a subset of those. The rectangle indicates middle A.

 

Octave

Note

-5

-4

-3

-2

-1

0

1

2

3

A

 

27,50

55,00

110,00

220,00

440,00

880,00

1760,0

3520,0

A#

 

29,14

58,27

116,54

233,08

466,16

932,33

1864,7

3729,3

B

 

30,87

61,74

123,47

246,94

493,88

987,77

1975,5

3951,1

C

 

32,70

65,41

130,81

261,63

523,25

1046,5

2093,0

4186,0

C#

 

34,65

69,30

138,59

277,18

554,37

1108,7

2217,4

4434,9

D

 

36,71

73,42

146,83

293,66

587,33

1174,7

2349,3

4698,6

D#

 

38,89

77,78

155,56

311,13

622,25

1244,5

2489,0

4978,0

E

20,60

41,20

82,41

164,81

329,63

659,26

1318,5

2637,0

5274,0

F

21,83

43,65

87,31

174,61

349,23

698,46

1396,9

2793,8

5587,7

F#

23,12

46,25

92,50

185,00

369,99

739,99

1480,0

2960,0

5919,9

G

24,50

49,00

98,00

196,00

392,00

783,99

1568,0

3136,0

6271,9

G#

25,96

51,91

103,83

207,65

415,30

830,61

1661,2

3322,4

6644,9