WERABE UNIVERSITY

Institute of Technology
Department of Information Technology
Chapter 6: Basics of Digital Audio
Digitizing Sound
 Microphone produces analog signal
 Computer deals with digital signal

Sampling Audio
Analog Audio
Most natural phenomena around us are continuous; they are continuous transitions between two
different states. Sound is not exception to this rule i.e. sound also constantly varies. Continuously
varying signals are represented by analog signal.

Signal is a continuous function f in the time domain. For value y=f(t), the argument t of the function
f represents time. If we graph f, it is called wave. (see the following diagram)

Fig1: analog signal

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A wave has three characteristics:
 Amplitude
 Frequency, and
 Phase

Amplitude: is the intensity of signal. This is can be determined by looking at the height of signal. If
amplitude increases, the sound becomes louder. Amplitude measures the how high or low the voltage
of the signal is at a given point of time.

Frequency: is the number of times the wave cycle is repeated. This can be determined by counting
the number of cycles in given time interval.

Phase: related to the wave s appearance.

Fig2: recording sound and the need for digitization

When sound is recorded using microphone, the microphone changes the sound into analog
representation of the sound. In computer, we can’t deal with analog things. This makes it necessary
to change analog audio into digital audio.

Analog to Digital Conversion

Converting an analog audio to digital audio requires that the analog signal is sampled. Sampling is
the process of taking periodic measurements of the continuous signal. Samples are taken at regular
time interval, i.e. every T seconds. This is called sampling frequency/sampling rate. Digitized audio
is sampled audio. Many times each second, the analog signal is sampled. How often these samples
are taken is referred to as sampling rate. The amount of information stored about each sample is
referred to as sample size.

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Analog signal is represented by amplitude and frequency. Converting these waves to digital
information is referred to as digitizing. The challenge is to convert the analog waves to numbers
(digital information).
In digital form, the measure of amplitude (the 7-point scale - vertically) is represented with binary
numbers (bottom of graph). The more numbers on the scale the better the quality of the sample, but
more bits will be needed to represent that sample. The graph below only shows 3bits being used for
each sample, but in reality either 8 or 16-bits will be used to create all the levels of amplitude on a
scale. (Music CDs use 16-bits for each sample).

Fig3: quantization of samples

In digital form, the measure of frequency is referred to as how often the sample is taken. In the graph
below the sample has been taken 7 times (reading across). Frequency is talked about in terms of
Kilohertz (KHz).

Hertz (Hz) = number of cycles per second

KHz = 1000Hz

MHz = 1000 KHz

Music CDs use a frequency of 44.1 KHz. A frequency of 22 KHz for example, would mean that the
sample was taken less often.

Sampling means measuring the value of the signal at a given time period. The samples are then
quantized.

Quantization is rounding the value of each sample to the nearest amplitude number in the graph. For
example, if amplitude of a specific sample is 5.6, this should be rounded either up to 6 or down to 5.

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This is called quantization. Quantization is assigning a value (from a set) to a sample. The quantized
values are changed to binary pattern. The binary patterns are stored in computer.

Fig4፡ digitization process (sampling, quantization, and coding)

Fig5፡ Sampling and quantization

Example:

The sampling points in the above diagram are A, B, C, D, E, F, H, and I.

The value of sample at point A falls between 2 and 3, may be 2.6. This value should be represented
by the nearest number. We will round the sample value to 3. Then this three is converted into binary
and stored inside computer.

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Similarly, the values of other sampling points are:
B=1 C=3
D=1 F=1 H=3
E=3 G=2 I=1
The values of most sample points are quantized. After quantization, we convert sample values into
binary digits.

Sample Rate

A sample is a single measurement of amplitude. The sample rate is the number of these measurements
taken every second. In order to accurately represent all of the frequencies in a recording that fall
within the range of human perception, generally accepted as 20Hz 20KHz, we must choose a sample
rate high enough to represent all of these frequencies.
At first consideration, one might choose a sample rate of 20 KHz since this is identical to the highest
frequency.
This will not work, however, because every cycle of a waveform has both a positive and negative
amplitude and it is the rate of alternation between positive and negative amplitudes that determines
frequency. Therefore, we need at least two samples for every cycle resulting in a sample rate of at
least 40 KHz.
Sampling Theorem
Sampling frequency/rate is very important in order to accurately reproduce a digital version of an
analog waveform.

Nyquist s Theorem:
The Sampling frequency for a signal must be at least twice the highest frequency component in the
signal.
Sample rate = 2 x highest frequency

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Fig6: Sampling at signal frequency and at twice Nyquist frequency

When the sampling rate is lower than or equal to the Nyquist rate, the condition is defined as under
sampling. It is impossible to rebuild the original signal according to the sampling theorem when such
sampling rate is used.
Aliasing

What exactly happens to frequencies that lie above the Nyquist frequency? First, we ll look at a
frequency that was sampled accurately:

In this case, there are more than two samples for every cycle, and the measurement is a good
approximation of the original wave. we will get back the same signal we put in later on when
converting it into analog.

Remember: speakers can play only analog sound. You have to convert back digital audio to analog
when you play it.

If we under sample the signal, though, we will get a very different result:

In this diagram, the blue wave (the one with short cycles) is the original frequency. The red wave (the
one with lower frequency) is the aliased frequency produced from an insufficient number of samples.

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This frequency, which was in all likelihood a high partial in a complex timbre, has folded over and is
now below the Nyquist frequency. For example, a 11KHz frequency sampled at 18KHz would
produce an alias frequency of 7KHz. This will alter the timbre of the recording in an unacceptable
way.

Under sampling causes frequency components that are higher than half of the sampling frequency to
overlap with the lower frequency components. As a result, the higher frequency components roll into
the reconstructed signal and cause distortion of the signal. This type of signal distortion is called
aliasing.

Sample Resolution/Sample Size

Each sample can only be measured to a certain degree of accuracy. The accuracy is dependent on the
number of bits used to represent the amplitude, which is also known as the sample resolution.

How do we store each sample value (quantized value)?

fl 8 Bit Value (0-255)
fl 16 Bit Value (Integer) (0-65535)

The amount of memory required to store t seconds long sample is as follows:

 If we use 8-bit resolution, mono recording memory = f*t*8*1
 If we use 8-bit resolution, stereo recording
memory = f*t*8*2

 If we use 16-bit resolution, and mono recording memory = f*t*16*1

 If we use 16-bit resolution, and stereo recording
memory =f* t*16*2 where f is sampling frequency, and t is time duration in seconds

Examples:
CS Students sampled audio for 10 seconds. How much storage space is required if

a) 22.05 KHz sampling rate is used, and 8-bit resolution with mono recording?
b) 44.1 KHz sampling rate is used, and 8-bit resolution with mono recording?
c) 44.1 KHz sampling rate is used, 16-bit resolution with stereo recording?
d) 11.025 KHz sampling rate, 16-bit resolution with stereo recording?

Solution:
a) m=22050*8*10*1

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 m= 1764000bits=220500bytes=220.5KB

b) m=44100*8*10*1

m= 3528000 bits=441000butes=441KB

c) m=44100*16*10*2

m= 14112000 bits= 1764000 bytes= 1764KB

d) m=11025*16*10*2

m= 3528000 bits= 441000 bytes= 441KB

Implications of Sample Rate and Bit Size

 Affects Quality of Audio
 Affects Size of Data

File Type 44.1 KHz 22.05 KHz 11.025 KHz

16 Bit Stereo 10.1 Mb 5.05 Mb 2.52 Mb

16 Bit Mono 5.05 Mb 2.52 Mb 1.26 Mb

8 Bit Mono 2.52 Mb 1.26 Mb 630 Kb

Table Memory required for 1 minute of digital audio

Clipping

Both analog and digital media have an upper limit beyond which they can no longer accurately
represent amplitude. Analog clipping varies in quality depending on the medium.
The upper amplitudes are being altered, distorting the waveform and changing the timbre, but the
alterations are slightly different. Digital clipping, in contrast, is always the same. Once an amplitude
of 1111111111111111 (the maximum value in a 16-bit resolution) is reached, no higher amplitudes
can be represented. The result is not the smooth, rounded flattening of analog clipping, but a harsh
slicing of off the top of the waveform, and an unpleasant timbre result.

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An Ideal Recording

We should all strive for an ideal recording. First, don’t ignores the analog stage of the process. Use
a good microphone, careful microphone placement, high quality cables, and a reliable analog-to-
digital converter. Strive for a hot (high levels), clean signal.
Second, when you sample, try to get the maximum signal level as close to zero as possible without
clipping. That way you maximize the inherent signal-to-noise ratio of the medium. Third, avoid
conversions to analog and back if possible. You may need to convert the signal to run it through an
analog mixer or through the analog inputs of a digital effects processor. Each time you do this,
though, you add the noise in the analog signal to the subsequent digital reconversion.