Fundamentals Series

Analog vs. Digital

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 Fundamentals Series

Signals H.323

Analog vs. Digital SIP

Network
Defining Quality
Communication I

Network
Standards
Communication II

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Welcome to Analog vs. Digital, the second module in the Polycom Fundamentals series. This module
is approximately 12 minutes long.
 Introduction

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In order to understand how videoconferencing works it’s important to understand the underlying
technologies at work behind the scenes.

In this short module we will compare analog and digital as signaling methods, and find out some
basics about how digital signaling works.

To start, let’s recap from the Signals module that an analog signal is one which constantly varies in
amplitude and frequency, causing the voltage to vary constantly. In comparison to this, a digital signal
is one which only has discrete (ie separate) values.
 Digital Basics

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One of the most important technological shifts in the last century was that from analog
technology to digital technology. It has touched everything we do and it is integral to
understanding modern communications, including video.

First let’s look at the concept of a digital signal. We can use an analog clock as an example
of a basic analog signal. The hand always sweeps through the numbers in continuous
motion, never stopping on a specific number. This is how analog works.

Now, compare that to a digital clock. It conveys the same information, but very differently. A
digital clock clicks from one value directly to the next in a series of steps instead of flowing
smoothly between them. This is what we mean when we say that digital uses discrete steps,
or values – these are created using binary signals.
 Binary Signals

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Digital signals use binary for transmitting information. We use a sequence of on and off
voltages to represent the value of each bit, either 1 (on) or 0 (off). This is the basis for all
digital communications. A string of on and off voltages grouped together to recreate digital
values of signals.

As it turns out binary is a great numeral system to use when using machines. It is much
easier to create a device built on switches with only two positions, on and off, or 1 and 0. So
when the first computers were being invented they realized that a binary system was the
most efficient way to count using computing machines. So in effect a computer is just a
bunch of switches that can be set to on or off. Combine a bunch of them and and voila you
can do complicated calculations just by turning things on and off really quickly.
 Digital Signals

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We looked at an analog signal in the Signals module. Now let’s compare that to a digital
one. The digital signal shown here represents the analog waveform above it as a sequence
of discrete values. Each discrete value, when connected, gives us a representation of the
analog waveform. So rather than dealing with a continuously changing waveform we are
charting a series of discrete values and stringing them together.

Now we know the difference between analog and digital. Great. Now what? Why do we care
about digital?
 Why Digital?

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One advantage of digital signals when compared to analog signals is that digital signals can
be transmitted without the same noise problems as analog signals.

Because each step of a digital signal is seen as a specific value some fluctuation can be
introduced without changing the perceived value of that part of the signal, within a certain
tolerance. If our original 1v value drifts a little and is now 1.12v it is still recognized as the
original 1v by the digital system. It’s close enough.

This is quite different from our analog waveform, which reacts to every fluctuation in the
wave from noise. And, those changes are often difficult to distinguish from the original
analog information. Not to mention that when we then amplify that signal the noise is
amplified as well.

With digital we can filter out noise we don’t want and boost the voltage of the signal
confidently using a device called a repeater, which makes a new signal from the original but
boosts it so it can keep moving and still be usable at the other end.
 Why Digital?

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Another advantage of digital signals is that they can be more easily stored. If I simply record
the sequence of values I can easily recreate the original digital signal, unlike analog which
would have to actually record the whole waveform itself and then play it back.

This becomes critical not only because we can do things like record a video conference for
review later, but also because when we transmit digital data across our networks it can be
cached temporarily in devices along the way. Any endpoints and MCUs are able to take
advantage of this to buffer the data prior to creating a video image or audio signal.

Given the time-critical nature of video and audio data this is a really cool feature, as it
means that small network issues have the potential to be evened out and the video and
audio played back from the buffer correctly. Without this ability, any data which cannot be
processed in time would just be dropped as it would be unusable. This also allows the
endpoints and MCUs to make sure the video and audio are in sync when they are sent to
each endpoint.
 Why Digital?

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Another advantage of digital signaling is that it is very flexible. Since the digital signal is
made up of a series of numbers, it’s easy to come up with different ways to arrange the
numbers to be transmitted.

And, as we’re just manipulating numbers, we can also come up with different ways to
process the information, especially if we are using computer controlled devices to
manipulate the digital signals. Improvements or changes to the signals can be introduced by
changes in software even when using the same hardware. Making an improvement to an
analog system can be much more difficult since the hardware itself must often be altered
and improved to give better performance or a new functionality.

This comes into play when we’re doing things like receiving one type of digital audio or video
and reformatting it to be another type. This is known as transcoding and it’s invaluable when
you want to make different devices compatible. For example an MCU will allow different
endpoints to use different audio and video formats in the same call by transcoding the media
to send to each one individually.
 Pulse Code Modulation

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So we know why digital is a good thing. But how do we make something digital? We in the
normal universe are analog. Pretty much all things in nature work in an analog fashion as
constantly changing variations. Temperature is a great example. It doesn’t just jump from
one temperature at 7am to 4 degrees warmer at 7:45am and then step up another 9
degrees at noon. It fluctuates gradually and continuously. In order to take this natural analog
information and turn it into digital values we need a mechanism to digitize it (turn the
information into numbers). That method is called pulse-code modulation, usually shortened
to PCM.

PCM creates a digital value representation of an analog signal by recording a value at

regular intervals and stringing them together to record discrete values over time. Those
values become the basis for digitizing a signal, and PCM is the basis for all analog to digital
and digital to analog conversions in the modern world. This creates our ‘connect the dots’
picture of the original analog signal.
 Sampling

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PCM is based on the idea of sampling, which is simply a measurement of an analog signal
at regular intervals. It gives you a representative image of what the analog waveform looked
like at the moment the samples were taken, kind of like a ‘connect the dots’ puzzle… only
we’re not ending up with the Statue of Liberty or a cartoon character, we end up with a
picture of the signal that we’re sampling. We end up with a series of numerical values that
tell us what happened to the signal over time.

The more often you sample (the higher the frequency) the more accurate your
representation is. If we sample more frequently we will have a better idea what the actual
analog signal looked like, again just like ‘connect the dots’. As an exact measurement of
when a sample should happen is required to make the process accurate, timing is a critical
factor in sampling. The number of samples per second is known as the sample rate, and if
we’re sampling 1000 times per second it’s a 1000 Hz sample rate or a 1 kHz sample rate.

Along with sample rate, we also need to decide how many bits we use to code each sample.
The number of bits we use to code a signal gives us what is called the ‘bit depth’ – CD audio
is commonly sampled at a 16-bit bit depth, for example. As the number of bits used
increases, the quality increases, but so also does the amount of data required to transmit
the whole signal.
 Sampling Theorem

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A theorem is just a rule which is expressed by a formula. The sampling theorem (also known
as the Nyquist Shannon theorem after two of the people who discovered it), is an important
part of calculating how often one needs to take samples in order to create an accurate
representation of the original signal without introducing too much noise into the process. The
theorem states:

‘If a function x(t) contains no frequencies higher than B Hertz, it is completely determined by
giving its ordinates at a series of points spaced 1/(2B) seconds apart’.

An ordinate is the name for a value on the y axis.

Essentially, all this means is that in order to get the best signal, you need to sample at least
twice the frequency of the highest frequency in the analog signal. So, an analog audio signal
of 4kHz (4000 cycles per second) needs to be sampled 8000 times in each one second
cycle. From that you can be relatively sure that when you recreate the signal from the
sequence of values it will be an accurate re-creation.
 Hearing Perception

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Applying all this to an audio signal, remember that in the Signals module we said that the
range of human hearing is between 20 Hz and 20,000 Hz, with human speech centering
around 1,000 Hz (1kHz).

Logically, this is also the easiest range of sounds for us to hear, requiring the least effort.
Sounds above and below this frequency range are harder to hear. So as telephone system
engineers worked on how to make the best quality phone calls they determined that
focusing on handling signals in the range from about 300 Hz to 4,000 Hz gave the best
performance for cost balance in telephony equipment.

Using the sampling theorem to determine how many samples we need to accurately
represent that signal would give us what? Let’s figure it out. If our range of frequencies is
300 to 4,000 Hz, our top end audio frequency is then 4000 Hz.

So if our formula is 1/(2B) where B = the highest frequency, B = 4000, and 1/(2B) = 1/(2 x
4000). So the time between samples in seconds is 1/8000, or 8000 samples per second.
This is an 8 kHz sampling rate.
 Encoding and Decoding

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In order to turn our analog signal into a digital signal using PCM we follow this process:

The values of the analog signal are sampled at certain regular times, then
Recorded at each point in time as a sequence of numbers
Then the values of each sample are converted to binary (using 8 bits) and end up with a
series of 0s and 1s in a long line. This is called ‘encoding’ the signal.

To transmit the signal, electrical pulses (0v “off” and +5v “on” for each bit) are generated at
regular intervals (8 bits per second) onto a wire. Because the far end system knows 8 bit per
second sequences are being sent, it can receive the signal, reverse the process and the
original analog signal emerges. This is called decoding the signal.
 Thank You

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