LETS TALK ABOUT AUDIO

Speed of Sound: 340m/s at 24C

At the forefront: Bell Telephone Laboratories

Audio Theory Basics

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UNDERSTANDING AUDIO

WHAT IS AUDIO
Simply put a number or range of frequencies detectable by human ear. The exact range
varies from source to source but everybody seem to agree that 20Hz to 20,000Hz is the
universal rough approximation of human hearing range. The true or exact range varies
form person to person, primarily depending on age, lifestyle and genetics. Those of us
exposed to loud environments are a subject to loosing the ability to hear or detect
frequencies within the widely accepted range much faster than the rest of us.
City life, traffic jams, air conditioning noise, building exhaust fans, fluorescent lights,
clubs, bars etc all have a tremendous impact on the overall health of our ears.

WHAT IS SOUND
When any object vibrates, air molecules surrounding it begin to vibrate sympathetically
in all directions creating a series of sound waves. These sound waves then create
vibrations in the ear drum that the brain perceives as sound.

WAVEFORM EXPLAINED
A waveform is a graphical representation of sound pressure level or voltage level as it
moves through a medium over time. The fundamental characteristics of a waveform can
be broken down into the following:
AMPLITUDE
Directly proportional to air pressure amplitude defines the actual volume or loudness of
a waveform

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WAVELENGTH
Defines the distance required to complete one waveform cycle.
FREQUENCY
The numbers of rarefactions and compressions, or cycles, that are completed every
second is referred to as the operating frequency and is measured in Hertz (Hz). Any
vibrating object that completes, say, 300 cycles/s has a frequency of 300 Hz while an
object that completes 3000 cycles/s has a frequency of 3 kHz.
VELOCITY
Is speed of sound which is about 340 m/s, provided the temperature outside is at 24C.
PHASE
When we deal with
multiple frequencies,
there

can

be

constructive and
d e s t r u c t i v e
i n t e r f e re n c e o r a
combination of both.

HARMONIC CONTENT
Each sound has a main frequency which is called the fundamental frequency. In
addition, sounds have a number of harmonic frequencies which are multiples of
fundamental frequency (more on this later).

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QUALITY OF SOUND
SAMPLE RATE
With any digital recording system, sound has
to be converted from an analogue signal (i.e.
the sound you hear) to a digital format that the
device can work with. Any digital recording
device accomplishes this by measuring the
waveform of the incoming signal at specific
intervals and converting these measurements
into a series of numbers based on the
amplitude of the waveform. Each of these
numbers is known as an individual sample
and the total number of samples that are taken every second is called the sample rate.
On this basis, the more the samples taken every second, the better the overall quality of
the recording. For instance, if a waveform is sampled 2000 times/s, it will produce more
accurate results than if it were sampled 1000 times/s.
While this may seem basic in principle, it becomes more complex when you consider
that the sampling rate must be higher than the frequency of the wave- form being
recorded in order to produce accurate results. If it isnt, then the analogue-to-digital
converter (ADC) could miss anything from half to entire cycles of the waveform, resulting
in a side effect known as aliasing.
This is the result of a real-world audio signal not being measured in the correct places.
For instance, a high-frequency waveform could confuse the ADC into believing its

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SOUND PROPAGATION & ACOUSTICS

actually of a lower frequency, which would effectively introduce a series of spurious lowfrequency spikes into the audio file.
To avoid this problem, you must make sure the sampling rate is greater than twice the
frequency of the waveform: a principle called the NyquistShannon theorem. This states
that to recreate any waveform accurately in digital form, at least two different points of a
waveforms cycle must be sampled. Consequently, as the highest range of the human
ear is a frequency of approximately 20kHz, the sample rate should be just over double
this range. This is the principle from which CD quality is derived. That is, the sample rate
of a typical domestic audio CD is 44.1kHz which is derived from the calculation:
Human hearing limit is 20 000 Hz
20000Hz x 2 = 40000Hz + 4100Hz (to make the rate more than twice the optimum frequency and
compensate for the anti-alias filter slope).
Although this frequency response has become the de facto standard used for all domestic audio CDs,
there are higher sampling frequencies available consisting of 48 000, 88 200, 96 000 and 192 000 Hz.

Though these sampling rates are far beyond the frequency response of CD, it is quite
usual to work at these higher rates while processing and editing. Although there has
been no solid evidence to support the theory, it is believed that higher sample rates
provide better spatial signal resolution and reduce phase problems at the highfrequency end of the spectrum as the anti-alias filters can be made much more
transparent due to gentle cut-off slopes. Thus, if the ADC supports higher rates,
theoretically, there is no harm in working at the highest available rate. Notably, many
engineers argue that because the signal must be reduced to 44.1 kHz when the mix is
put onto CD, the downward con- version may introduce errors, so working at higher
sampling rates is pointless. If you do decide to work at a higher sample rate, it may be
worthwhile using a rate of 88 200 kHz, as this simplifies the down-sampling process.

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BIT DEPTH
In addition to the sample rate, the bit rate
also determines the overall quality of the
results of recording audio into a digital
device. Within digital audio recording
system, the number of bits determines
the number of analogue voltages that are
used to measure the volume of a
waveform, in effect increasing the overall
dynamic range. In technical applications
the dynamic range is the ratio between
the residual noise (known as the noise floor) created by all audio equipment and the
maximum allowable volume before a specific amount of distortion is introduced.
According to the IEC 268 standard, the dynamic range of any professional audio
equipment is measured by the difference between the total noise floor and the
equivalent sound pressure level where a certain amount of total harmonic distortion
(measured in THD) appears.
In relation to music, the dynamic range is the difference between the quietest and
loudest parts. For instance, classical music utilizes a huge dynamic range by suddenly
moving from very low to very high volume, the cannons in the 1812 Overture being a
prime example. Most dance and pop music, on the other hand, has a deliberately
limited dynamic range so that it remains permanently up front and in your face.
Essentially, this means that if a low bit rate is used for a recording, only a small dynamic
range will be achieved. The inevitable result is that the ratio between the noise floor and
the loudest part of the audio will be small, so background noise will be more evident.

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When the bit rate is increased, each additional bit introduces another analogue voltage,
which adds another 6 dB to the dynamic range.

For example, if only one bit is used to record a sound, the recorder will produce a
square wave at the same frequency as the original signal and at fixed amplitude. This is
because only one voltage is used to measure the volume throughout the sampling
frequency. However, if an 8-bit system is used, the signals dynamic range will be
represented by 256 different analogue voltages and a more accurate representation of
the waveform will result. Its clear, then, that a 24-bit audio signal will have a higher
dynamic range than a 16-bit signal (the bit rate used by CDs)

Although 24-bit is the widely accepted standard resolution available to samplers and
soundcards ADCs, a proportionate amount of audio-capable software utilizes internal
32- or 64-bit processing. The reasoning behind using bit rates this high is that whenever
any form of processing is applied to a digitized audio signal, quantization noise is
introduced. This is a result of the hardware rounding up to the nearest measurement
level. The less rounding up the hardware has to do (i.e. the more bits used), the less
noise is introduced.
This is a cumulative effect, meaning that as more digital processing is applied, more
quantization noise is introduced, and quantization noise needs to be kept to a minimum.
In more practical terms, this means that while a CD may only accept a 16-bit recording,
if a 24-bit process is used throughout digital mixing, editing and processing, when the
final sound is dropped to 16-bit resolution for burning to CD the quantization noise will
be less apparent.

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DITHERING
The process of dropping out bits from a recording to reduce the bit rate is known as
dithering. Understanding how this actually works is not vital; what is important is that
the best available dithering algorithms are used. Poor quality algorithms will have a
detrimental effect on the music as a whole, resulting in clicks, hiss or noise. As a
reference, Apogee is well known and respected for producing excellent dithering
algorithms.
It isnt always necessary to work at such a high bit rate; some genres of electronic
music benefit from using a much lower rate. For instance, 12-bit samples are often used
in hip-hop to obtain the typical hip-hop sound. This is because the original artists used
old samplers that could only sample at 12-bit resolution; thus, to write music in this
genre its quite usual to limit the maximum bit rate in order to reproduce these timbres.
Similarly, with trip-hop and lo-fi, the sample rate is often lowered to 22 or 11kHz, as this
reproduces the gritty timbres that are characteristic of the genre.

HEALTHY WAVEFORM (RECORDING PROCESS)

Ultimately, no matter what sample or bit rate is used, its important that the converters
on the soundcard or digital recorder are of a good standard and that the amplitude of
the signal for recording is as loud as possible (but without clipping the recorder).
Although all digital editors allow the gain of a recorded signal to be increased after
recording, the signal should ideally be recorded as loud as possible so as to avoid
having to artificially increase a signals gain using software algorithms. This is because all
electrical circuitry, no matter how good or expensive, will have some residual noise
associated with it due to the random movement of electrons. Of course, the better the
overall design the less random movement there will be, but there will always be some
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noise present so the ratio between the signal and this noise should be as high as
possible. If not, when you artificially increase the gain after it has been recorded, it will
increase any residual noise by the same amount.
For instance, suppose a melodic riff is sampled from a record at a relatively low level,
then the gain is artificially increased to make the sound audible; the noise floor will
increase in direct proportion to the audio signal. This produces a loud signal with a loud
background noise. If, however, the signal is recorded as loud as possible, the ratio
between the noise floor and the signal is greatly increased. There is a fine line between
capturing a recording at optimum gain and actually recording the signal too loud so that
it clips the recorders inputs.
This isnt necessarily a problem with analogue recorders as they dont immediately
distort when the input level gets a little too high they have headroom in case the odd
transient hit pushes it too hard but digital recorders will clip immediately. Unlike
analogue distortion, which can often be quite pleasant, digital distortion cuts off the top
of the waveform, resulting in an ear-piercing clicking sound. This type of distortion
obviously should be avoided, so you need to set a recording level that is not loud
enough to cause distortion, yet not low enough to introduce too much noise.
All recording software or hardware, including samplers, will display a level meter
informing you how loud the input signal is, to help you determine the appropriate levels.
Before beginning to record, its vital to set this up correctly. This is accomplished by
adjusting the gain of the source so that the sig- nal overload peaks light on only the
most powerful parts and then backing off slightly so that they are just below the clipping
level. This approach is suitable only if there isnt too much volume fluctuation throughout
the entire recording, though. If a sample is taken from a record, CD or electronic

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instrument theres a good chance that the volume will be constant, but if a live source,
such as vocals or bass guitars, are recorded there can be a huge difference in the
dynamics. Its doubtful that any vocalist, no matter how well trained, will perform at a
constant level; in the music itself its quite common for the vocals to be softer in the
verse sections than they are in the chorus sections.
If the recording level is set so that the loudest sections dont clip the recorder, any
quieter section will have a smaller signal-to-noise ratio, while if the recording levels are
set so that the quieter sections are captured at a high level, any louder parts will send
the meters into the red. To prevent this, its necessary to use a device that will reduce
the dynamic range of the performance by automatically reducing the gain on the loud
parts, while leaving the quieter parts unaffected, thus allowing the recording levels to be
increased on the quietest parts.
This dynamic restriction is accomplished by strapping a compressor between the signal
source and the input of the sampler or recording device. When the compressor is set to
squash any signal from the source above a certain thresh- old, any peaks that could
cause clipping in the recorder are reduced and the recording can be made at a
substantially higher volume. (expanded on in the compressor chapters of this guide)

BITRATE (MP3)
The amount of information stored/processed/ reproduction per second, (bits per second).
96 kbit/s FM quality
128160 kbit/s Standard Bitrate; difference can sometimes be obvious (e.g. bass quality)
192 kbit/s DAB (Digital Audio Broadcast) quality
224320 kbit/s Near CD quality

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