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The Audio Pages

Elliott Sound Products Frequency & Amplitude Explained

Frequency, Amplitude & dB

Rod Elliott (ESP)
With Thanks to Lenard Audio [1]

Articles Index
Main Index

Contents

Introduction
1.0 - Frequency
1.1 - Musical Notation
1.2 - Wavelength
2.0 - Amplitude - dB
2.1 - dB Reference Levels
2.2 - Weighting Curves
3.0 - Frequency & Amplitude
4.0 - Crossover Networks & Filters
Conclusion
References

Introduction

Sound is carried from the source to our ears or a microphone by means of minute vibrations,
which are passed through the air. Sound has two primary components, frequency and intensity.
The frequency refers to the pitch of the tone or other sound, and typical sounds have many
different frequencies all happening at once. Frequencies are measured in Hertz (Hz), named
after the physicist Heinrich Hertz. The old standard (now discontinued almost everywhere) used
Cycles per Second (cps) as the standard measurement. Hz and cps are the same thing - both
refer to the number of complete cycles of a waveform in one second.

Sound intensity (or amplitude) is measured in decibels (dB). The prefix 'deci' means one tenth.
The Bel was invented by engineers of the Bell Telephone Laboratory to quantify the reduction in
audio level over a 1,600m (1 mile) length of standard telephone cable, and was originally called
the transmission unit or TU. It was renamed in around 1923-4 in honour of the Bell Laboratory's
founder Alexander Graham Bell. Because the Bel is too large for general use, the dB became
the preferred unit.

1.0 - Frequency

The range of frequencies we humans can hear is generally taken as being from 20Hz to 20,000
Hz (20kHz), but the conditions are not usually specified. As we get older, the first to suffer is the
high frequencies, and by around 50 years of age, most males will be limited to around 14-
15kHz, with females usually suffering less loss. Frequencies below 25Hz are felt rather than
heard, but the conditions under which we experience such low frequencies make a big
difference to how they are perceived.

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Figure 1 - Typical Human Hearing Range

Our hearing is most sensitive at around 3.5kHz, as shown in Figure 1. Our hearing, eyes and
sensitivity to touch or pain, are all logarithmic functions. This enables us to experience a vast
variation with each sense. As the intensity of the sense increases, we automatically compensate
by reducing our sensitivity. In this way, we can hear the gentlest rustle of a leaf in a tiny breeze
at a sound pressure level (SPL) of 0dB, but are not instantly deafened by a nearby jack-hammer
at perhaps 1,000,000,000,000 (1 British billion, 1 US trillion, or 1 x 1012 ) times the sound power
(120dB SPL).

When two frequencies are close to each other, our hearing plays some interesting tricks on us. If
one tone is 6dB louder than the other, we will not hear the second tone. This is called acoustic
masking, and is used by the MP3 format to remove a great deal of the "redundant" audio
information. This reduces the size of the file dramatically, and with some of music the end result
may be almost indistinguishable from the original. Material with rich harmonic structure is less
successful, with cymbals and harpsichords suffering because there is simply too much
information and none of it is actually redundant.

1.1 - Musical Notation

In (western) music, we generally use the equally tempered scale. While not absolutely musically
accurate, it does allow musicians to make key changes (moving an entire piece of music up or
down the musical scale) without having to re-tune their instruments. This is a vast topic, and
requires a great deal more than you will find here if it is to be fully understood. Unless you are a
musician, a full understanding is not required.

Musical notation is based on the use of 12 semitones in each octave. An octave is the perfect
interval between the 1st and 8th tones of the diatonic scale. See Answers.com if you want more
specific information about the diatonic scale.

In western music, each octave is comprised of 12 semitones. An octave is double or half the
original frequency, so (for example) one octave from middle A (440Hz) is 880Hz or 220Hz. Both
"new" notes are called A. The word octave is derived from "Octo-" (Latin/Greek) meaning eight,
because the western octave is divided into 8 "full" tones in the diatonic scale.

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Figure 2 - Musical Scale & Frequencies

Figure 2 shows the range - the keyboard is shown as a reference only, and is not meant to be
that of a real piano. Of common musical instruments, open E on a (4 string) bass guitar or
double bass has a frequency of 41.2Hz, while a grand piano's bottom A is 27.5Hz. Many
instruments can get far lower - examples being pipe organs and electronic synthesisers.

High frequencies are more complex. Any note is made up from the fundamental (usually taken
as the lowest frequency component of the sound - the first harmonic) and a series of harmonics
above this (usually at octave intervals). While many instruments produce harmonics that are
exact multiples of the fundamental, others do not. A flute also contains wind noise, reed
instruments often have very complex harmonic relationships, and percussion instruments can
have harmonics that are not related, but extend to well beyond our hearing range (snare drums,
cymbals, etc). With many plucked or struck stringed instruments the second harmonic is
dominant (louder than the fundamental). This is especially noticeable with guitar, but is apparent
with many other instruments too.

The division of an octave into 12 equally spaced tones is done using the 12th root of 2
(approximately 1.0594631). If you multiply 440 by the full version of this number 12 times, you
get 880 - exactly one octave (depending on your calculator). The same method may be used to
divide an octave into any number of divisions - for example, 3 divisions are used for 1/3 octave
band graphic equalisers. The third root of 2 is approximately 1.26 in case you were wondering

A decade (one tenth or ten times the frequency) is approximately 3.2 octaves (3.1623 or the
square root of 10). Decades are sometimes used instead of octaves in engineering, although
current practice prefers to use octaves.

Frequency and amplitude are inextricably coupled in the real world, with both playing an equally
important role. It is only in test and measurement where these two functions are separated, and
that is so we can see how one affects the other to ensure that a reasonable standard is
achieved.

1.2 - Wavelength
The wavelength of any signal depends on the form of the signal (acoustic or electrical), the
transmission velocity in the medium (air, concrete, an electrical wire) and the frequency. For
audio, we are generally only concerned with the wavelength in air. While the wavelength of RF
(radio frequency) signals in cables is usually very important, the wavelengths at audio
frequencies are very large indeed. A 20kHz signal has a theoretical wavelength of 15,000 metres
(15 km) as an electrical signal, ignoring other effects such as velocity factor (look it up if you are
interested).

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Sound in air at 22C and at sea level has a velocity of 345m/s [2]. The speed of sound varies
markedly with temperature and is proportional to temperature, but the Hyperphysics calculator
will work it out for you if you need to know exactly.

The formula to convert frequency to wavelength (commonly written as - the Greek letter (lower
case) lambda) is ...

= c / f where c is velocity of sound and f is frequency

It is also useful to remember that sound travels at about 345mm / ms (both metres and 1 second
divided by 1,000). Our hearing mechanism is carefully refined to ensure that sounds we hear are
made as clear as possible, so we automatically reject repeat sounds (echoes) that arrive within
about 30ms of the original. This allows us to hear clearly even in a reverberant room (or a cave
a few millennia ago). 30ms means a distance of around 11.5 metres, meaning a cave room of
about 5 metres square. Such a room will sound somewhat odd, but speech is still clear. Larger
rooms (with longer delays) can cause a significant loss of intelligibility.

Being able to calculate wavelength is very important for anyone designing loudspeakers, as
there are many characteristics of a speaker box design and room placement that rely heavily on
knowledge of wavelength and time delay. These topics are covered in countless white papers,
articles and books, and are not relevant to the material in this article.

2.0 - Amplitude - dB

Most beginners in electronics find dB very confusing. This is understandable, but it is easy to
learn, and is every bit as important as Ohm's law when working with electronics or
loudspeakers. The main thing to remember is that 1dB remains 1dB, regardless of the context.
Likewise, 6dB remains 6dB. Let's look at the formulae first (no, they are not hard - calculators do
almost all the work). For those who prefer not to use a calculator, see the Lenard Audio [1]
website.

dB = 20 * log (V1 / V2 )
dB = 10 * log ( P1 / P2 )

Where V1 and V2 are any two voltages, and P1 and P2 are any two powers (in
Watts).

But why are there different formulae? This is simple - power into a given impedance or
resistance is determined by the square of the voltage. If 1 Volt into 1 Ohm gives 1 Watt, 2V into
1 gives not 2W, but 4W ( P = V / R ). The multiplication by 10 or 20 takes this into account, so
it doesn't matter if you work with power or voltage, you get the same answer in dB.

Using dB provides a convenient way to indicate very large numbers, and in a way that directly
relates to the way we hear. For example, it is standard practice to measure frequency response
of amplifiers, speakers and many other things at the -3dB points. Speakers are commonly
quoted as (for example) 40Hz - 20kHz 3dB. 3dB means half or double the power, or a voltage
ratio of 1.414:1

That last number is a good one to remember - the square root of 2 ( 2 ) is 1.414, and it is used
in many electronics calculations.

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Figure 3 - dB Range vs Voltage

Figure 3 shows the range generally accepted as the minimum dynamic range in audio. As you
can see it is vast, covering a span of 1 million to one. The total range that is of interest spans
120dB, being the dynamic range of typical good quality analogue and digital equipment. A
microphone preamp may be quoted as having an equivalent input noise of -127dBm ... feel free
to calculate the noise level in millivolts (it will actually be microvolts). Using dB to express such
small numbers is far more intuitive than specifying the noise level as 0.346uV, which although
impressively small, tells us nothing about its audibility.

Here are a couple of very useful dB facts that are worth remembering ...

3dB = half or double the power

10dB = half or twice as loud 10dB = one tenth or ten times the power

Perceived loudness is what you hear as the change, and means that if you have a 100W
amplifier and you want the sound to be twice as loud, you need to use a 1kW (1,000W) amplifier
to do so. Note that doubling the power results in a 3dB increase, and although audible it is not
dramatic. It was determined long ago that 1dB is the smallest change that the average listener
can hear. While open to some dispute at regular intervals, it still holds if the test is done with a
single tone under ideal conditions.

2.1 - dB Reference Levels

While it is sometimes believed that dB is either some absolute value or a "dimensionless
number", neither is correct. Many standards exist to refer to specific levels, both with sound and
electrically.

0 dB SPL = 20 microPascals (0.0002 dynes / cm) - the smallest sound we can (normally)
hear at 2kHz
0 dBm = 1mW in a 600 Ohm load (0.775mV) - based on the telephone system, which
specified the impedance to be 600 Ohms
0 dBu = 775mV - this looks the same as dBm, but no impedance is referenced, only the
voltage
0 dBV = 1V RMS, impedance not specified

2.2 - Weighting Curves

When sound level readings are taken, it is common to apply what is known as A-weighting (see
Project 17 for a design and frequency response of an A-weighting filter). The A-weighting curve
is designed to allow for the fact that out hearing is less sensitive at low and high frequencies, but
fails to account for the actual SPL. When sound is above 100dB SPL, our hearing response is
reasonably flat (see Figure 1), and the use of A-weighting is inappropriate. Under these
conditions, the C-weighting curve should be used, which has an essentially flat response over
the audio band.

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A-weighting is also often used for measuring amplifier noise, and because this is normally only
ever at very low volume, the use of the A-weighting filter is appropriate. Personally I prefer not
to use it at all, but most do.

3.0 - Frequency & Amplitude

A frequency response curve is an example of the use of both frequency and amplitude, with
frequency being shown on the X (horizontal) axis, and amplitude on the Y (vertical) axis.

Figure 4 - dB Range of Long-Term Music (Source: FM Radio)

Figure 4 shows an example of a frequency response curve, in this case taken from my Clio
analyser. The source material was an FM radio tuner, and the program was set up to show the
highest peaks over a 15 minute period. Note that the chart includes any equalisation applied by
the radio station (I used radio Triple J as the source - they do not play advertisements, thus
eliminating pollution caused by the often radical EQ and compression that is used in ads to
make them sound "loud". The 19kHz FM stereo pilot tone is just visible on the right side of the
graph, and you can see that the FM bandwidth is limited to 15kHz. (The pilot tone is used to
identify a stereo transmission, and is used by the stereo decoder to obtain separate left and right
channels.)

Figure 5 - Overall Energy Distribution of "Typical" Music

It is generally accepted that the overall energy distribution of music looks like that shown in
Figure 5. That there will be variations is obvious, and while interesting and potentially useful, you
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Figure 5. That there will be variations is obvious, and while interesting and potentially useful, you
cannot rely on any simple graph to determine how much power you need. Loudspeaker
efficiency and peak to average ratio of the signal must also be considered.

Peak to average ratio is an important topic itself. Because music has dynamics (loud and soft
passages), and because of the nature of a complex audio waveform, the RMS (root mean
squared) voltage is useful only to get an idea of the average power delivered to a speaker. The
RMS value of a sinewave is 0.707 of the peak voltage, as shown below.

Figure 6 - Peak vs RMS Value of a Sinewave

You may recall that I said earlier that one should remember the number 2 (1.414). The RMS
value of a sinewave is determined by dividing the peak value by 1.414, or you may multiply by
0.707 (the reciprocal of 1.414 ... i.e. 1 / 1.414 ). In Figure 6, the peak value of the sinewave is
1V, and the RMS value is 707.1mV. Most meters display the RMS voltage, but only those called
"True RMS" will get the value right for a complex waveform such as that shown in Figure 7. Not
that the waveform is especially complex - it is made up from 3 sinewaves, at 1kHz, 2kHz and
4kHz, all with a peak voltage of 1V.

Figure 7 - Peak vs RMS Value of a Non-Sinewave

The real RMS voltage of the waveform in Figure 7 is 1.225V. If one uses the calculated RMS
voltage (based on the peak voltage of 2.33V), the answer is 1.566V - an error of almost +22%
(+2.13dB). Most meters are average reading, RMS calibrated, meaning that the signal is
rectified and averaged, but the meter scale is calibrated to read RMS. Such a meter will give a
reading of 1.014V, a -12% error (-1.65dB). It is very easy to introduce serious errors into any
calculation that involves complex waveforms, and this is one of many reasons that a reasonably
pure sinewave is specified for most test procedures. While so-called "True RMS" multimeters
are more accurate, most do not handle high crest factors well. The crest factor is the ratio of the
peak and actual RMS values of a waveform, and to work well with high crest factors, some
serious maths is generally needed. Digital oscilloscopes with voltage readouts compute the
value, and will usually get it right.

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4.0 - Crossover Networks & Filters

Because crossover networks are an unavoidable requirement in quality loudspeaker systems,
they also require some explanation. Crossovers are used to separate the audio band into a
number of separate frequency bands. The frequencies are chosen to suit the loudspeaker
drivers being used, and (to some extent) the requirements of the designer.

Minimum Maximum
Driver Type
Frequency Frequency
Subwoofer < 20Hz 100Hz
Woofer 40Hz 300-3kHz
Mid Woofer 100Hz 3kHz
Midrange 300Hz 3kHz
Tweeter 1.5kHz > 20kHz
Super
10kHz 30kHz
Tweeter
Typical Loudspeaker Driver Ranges

The above table is not intended to be absolute. There are a great many factors that influence
the way a driver can (or should) be used, and these are not relevant to this article. The
crossover network is also subject to many variations. Apart from the choice of frequency, there
is also the choice of slope (the rate of attenuation with frequency), some networks are
deliberately designed to be asymmetrical, having different slopes for the high-pass and low-pass
sections.

Filters are divided into three different types ...

1. Low Pass - passes low frequencies, blocks high frequencies

2. High Pass - passes high frequencies, blocks low frequencies
3. Band Pass - passes frequencies within a specified bandwidth, blocks frequencies above or
below the passband

No filter simply stops all signals above or below the specified frequency. As the selected
frequency is approached, the signal level starts to reduce, and the filter frequency is usually
taken as that frequency where the signal level is 3dB below the passband. There are exceptions,
and these will usually be explained in the description of the network.

In order to obtain different rolloff slopes, filter "building blocks" can be connected in series to
obtain a greater rate of attenuation. The commonly used filter orders are as shown below. The
simplest filter is a first order, and uses one reactive component (a capacitor or an inductor). A
second order filter uses two reactive elements, and so on.

Filter Order Rolloff Slope Reactive Elements

First 6dB / octave 1
Second 12dB / octave 2
Third 18dB / octave 3
Fourth 24dB / octave 4
Commonly Used Filter Types

Active filters require power - they are called "active" because they use active components, such
as opamps, transistors or sometimes valves. Passive filters use only passive components -
capacitors and inductors. Passive filters always have losses (especially resistance in inductors),
so not all the amp power gets to the speakers. At high power levels the losses can become very
high, reducing the available power for the speakers.

Active filters require a separate power amplifier for each loudspeaker driver, while passive
networks use a single amp. There is a tradeoff - do we use large and expensive passive

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networks use a single amp. There is a tradeoff - do we use large and expensive passive
components and a single (relatively) large power amplifier, or an active crossover and a number
of smaller power amps?

It depends on what we are trying to achieve, the expected performance and the budget. It would
be silly to use an active crossover and separate amps for a cheap PC speaker, and equally silly
to use passive crossovers in a large sound reinforcement system running at perhaps 5,000W or
more. All filters (whether active or passive) will provide a rolloff slope based on the filter order.
With passive crossovers, it is usually necessary to compromise because high-order filters
become too expensive and consume excessive power.

Figure 8 - Typical Filter Slopes (Only 3 Shown for Clarity)

These filters are all set for 1.1kHz so they can be compared. This is not usually considered a
useful frequency for loudspeakers, but is convenient for purposes of illustration. Here you can
see the rate of rolloff for the 3 types shown. Higher order filters provide greater protection for the
speaker (especially tweeters), but cause greater phase shifts than low order filters. While not
usually audible, some designers will try to avoid phase shift as far as possible.

All filters cause phase shift - it is a characteristic of how they function in the analogue world.

Conclusion

All of the examples in this section show a combination of frequency and amplitude. It must be
stressed that a full and complete understanding of these topics is essential to your
understanding of audio as a whole. Without that understanding, you are left wondering what
certain terms really mean. You may also become less likely to believe some of the outrageous
drivel that is spouted by some manufacturers - they rely on a lack of understanding to baffle
people with pseudo-science.

References
Many of the images in this page came from Lenard Audio (with permission). They have been
modified and adapted to the style normally found in the ESP site for general compatibility.

1. Lenard Audio (Education Pages)

2. HyperPhysics

Articles Index
Main Index

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Copyright Notice. This article, including but not limited to all text and diagrams, is the intellectual property of Rod
Elliott, and is Copyright 2006. Reproduction or re-publication by any means whatsoever, whether electronic,
mechanical or electro- mechanical, is strictly prohibited under International Copyright laws. The author (Rod Elliott)
grants the reader the right to use this information for personal use only, and further allows that one (1) copy may
be made for reference. Commercial use is prohibited without express written authorisation from Rod Elliott. Some
parts of this article are copyright John Burnett (Lenard Audio).
Page created and copyright 01 December 2006

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