Introduction
At the start of the satellite broadcast era, all radio and television programs
were transmitted
in an analogue format. Only the last couple of years one has started to
transmit in a digital
format. This was made possible by establishing digital transmission formats.
The advantage
of digital data transfer is the high quality of the picture and the low
losses involved. Also,
thanks to the used compression techniques, more programs can be transmitted
over the
available distribution channels.
Digital television systems
Using digital television, the amount of data when no compression is used,
is very high. For
digital television, the following sample frequencies are used according
to ITU-R
recommendation no. 601 :
13.5MHz for the luminance signal (Y) and 6.75MHz for both color difference
signals. Using a
8-bit quantising method, we end up with a bitstream of 216Mbit/s. The required
bandwidth for
a signal like this is so big, that even a satellite transponder can not
cope with it. The
technique used in digitising and compressing digital broadcasts has been
developed by the
Motion Picture Expert Group (MPEG). Digital Video Broadcasting (DVB) in
Europe uses the
MPEG-2 format. Using this format and a modem, a variety of extra services
becomes
available, like extensive interactive services.
Compression converts the analogue video signal in a digital signal with
a bitstream varying
between 2 and 15Mbit/s (MPEG-2 video). The audio signal is compressed between
32 and
448Kbit/s. Multiplexing (MPEG-2 systems) combines video and audio, but
can also add
multiple AV signals together in a single Transport Stream (TS). - Modulation
takes care of a
transparent bitstream of say 38.1Mbit/s in a single 8MHz channel. For cable
systems, QAM
is used, for satellite transponders QPSK is used. By using compression,
it is now possible to
transport more than a single channel over either cable or satellite transponder.
Within DVB, the scrambling of the signals is standarized. Various Conditional
Access
(CA)-systems are offered. The use of these enables services like pay-TV
and pay per view.
The DVB service information (SI) offers the possibility to add special
information to the
datastream to describe the contents of the program transmitted. It enables
the set top box to
configure itself and aid the viewer in finding TV or radio programs.
Digital signal processing
Audio and video signals are essentially analogue signals, that is, signals
of which shape,
amplitude and frequency continuously changes. Until recently, processing
of these signals
could only be accomplished in an analogue way. The characteristics of analogue
signals,
when processed in electronic circuits, will be influenced. At every processing
stage, the
quality of the signal degrades. Just think about copying videotapes. A
copy of a copy
posesses less quality than the original tape.
By applying digital techniques, these disadvantages have come to an end.
Using this
technique, we can now keep the quality of the processed signals at a constant
quality and
level. A digitised signal can be displayed, transported and processed completely
free of
distortion.
A digital signal no longer exists of a continuously varying signal but
of several individual
signals. Every momentarely signal is represented by a digital code. We
call it digital signal
processing when all binary representations of the original signal are processed
following the
rules of digital technique.
Source coding of high quality picture- and sound signals plays an important
role. One of the
reasons is the high transferbandwidth and enormous storage capacity needed
for linear
coded signals.
Digitising
Digitising means the conversion of an analogue signal into a digital signal
consisting of
zeroes and ones. Converting analogue signals to digital (A/D conversion
in short) is done in
three steps :
Sampling
Quantising
Conversion to digital numbers
To convert a digital signal back into its analogue format, all three stages
of the process are
carried out in reverse order. One than uses a D/A convertor.
Sampling means that the source signal is sliced into equal sections over
a time period of 1
second. The audio on a Compact Disc e.g. is divided in 44100 sections per
second.
Therefore, the sampling frequency used is 44.1KHz.
For a video signal, the signal is divided into 13500000 sections per second.
This explains the
sampling frequency of 13.5MHz. For this process, a couple of conditions
apply. The most
important being that the sampling frequency has to be at least twice the
highest frequency
present in the source signal.
Quantising means that every section has its own scale with which the amplitude
of the signal
is measured. This scale is notated in bits. For video, an 8-bit quantizing
method is used.
Using 8 bits, one can have a total of 256 variances of a signal. This is
more than sufficient for
the human eye, since we can only recognise about 200 variances in luminance
anyway.
For audio, normally 16 bits are used. Using 16 bits, a total of 65536 different
sound variances
are possible. In other words, the resolution is 65536, which is more than
sufficient for the
human ear.
Conversion to digital numbers means, that a measured audio value of say
32768 is not
represented as a number but as a binary value of ones and zeroes (in this
example as
0111111111111111).
Following this, the digital signal is coded and given an error coding to
enable us to correct
errors at an later stage. The amount of data is given in the number of
bits per second. A
digital signal contains a fair amount more information than an analogue
signal. To be able to
store all this information, datacompression is used.
DIGITIZING AND COMPRESSING AUDIO
Introduction
To digitise audio signals, a couple of different sampling frequencies (depending
on the
application) are used : 32KHz, 44.1KHz or 48KHz. The quantising scheme
is normally 16
bits. On an audio CD, all information is registered and therefore a lot
of bits are used. This
form of coding is called linear coding.
By using compressing techniques, the amount of data can be strongly reduced.
These forms
of datareduction are used in e.g.
Digital Compact Cassette (DCC)
Mini Disc
Digital Audio Broadcasting (DAB)
Astra Digital Radio (ADR)
Digital Video Broadcast (DVB)
Digital Video Disc (DVD)
Source coding
The MPEG system committee determines the norm for the combination of a
large number of
coded audio- and videosignals in a single datastream. This norm guarantees
the
synchronisation of audio and video and enables the transfer of combined
information by
using various digital media. After having tested various applications,
the MPEG experts have
established the audio coding algorithm. Depending on the application, a
total of three layers
with increasing complexity and reduction capacity can be used. For a recording
that can not
be distinguished from the original, this comes to :
Layer 1, 2x 192Kbit/s
Layer 2, 2x 126Kbit/s
Layer 3, 2x 64Kbit/s
Important for an economical use of the number of available bits is the
source coding of the
signal. Source coding means the amount of bits that are created after the
A/D conversion. As
an example : By sampling an audio signal with a sampling frequency of 44.1KHz,
using 16
bits per sample, an audio stream of 44100 x 16 x 2 = 1410000 bits/s is
generated.
By using intelligent algorithms which take the properties of the human
ear into account, the
amount of data can be strongly reduced. Bits can be saved by redundancy,
or by simply
throwing away irrelevant parts of the signal. With irrelevant, we mean
those parts that the
human ear does not use. In other words, frequencies that are outside our
hearing capabilities
do not have to be registered.
Apart from this, there is another important dynamical effect. This is the
phenomena that a
very load tone masks a weaker tone so that it can not be heard anymore.
This is a very
complex psycho-acoustic effect. By leaving off this information, the total
soundimpression is
not effected. To calculate which parts of an audio signal can be heard
or not, the signal is
divided into subbands. For Musicam for instance, the digital audio signal
is divided into 32
subbands with an equal width. In the coder, 12 subsequent samples of the
subband are
combined to a block to calculate the mask effect.
Every subband is allocated a couple of bits, depending on the need. This
way, no more bits
than stricktly necessary are used. Also, this way, the accuracy is as high
as possible. By
using this method, it is now possible to reduce the 1.4Mbit/s bitstream
on a CD to just
200Kbit/s.
The most important data-reduction and coding methods for recordings are
:
MUSICAM (Masking Pattern Universal Subband Integrated Coding and Multiplexing
PASC
(Precision Adaptive Subband Coding).
Musicam is used in DVB, DAB and ADR, whilst PASC is used on a DCC.
DIGITIZING VIDEO IMAGES
Introduction
Digitising a video image is far more complex than for an audio signal.
Because of the far
higher frequencies used in television, the datarate is much higher. For
a video image, this is
about 100 times than what is needed for audio.
Television pictures consist of lines. In Europe, we use 25 frames per second,
each frame
consisting of 625 lines. At 25 frames per second, the human eye experiences
the frame
changes as a flicker. For this reason, interlacing is used. That is, every
625-line image is
divided into two equal 312.5 line frames. The first frame carries the even
lines, the second
carries the odd lines. The two frames combined for the complete image again.
To get the proper definition, the television signal should have a certain
bandwidth. A complete
picture should consist of 530 x 400 - 212000 pixels. The required bandwidth
than is 5MHz.
This applies to a black and white picture.
For a color picture, another calculation applies. A PAL color pictures
is made up out of 768 x
576 pixels for a complete frame.
Color television
Color television uses the primairy colors RED, GREEN and BLUE. By adding
those primairy
colors, other colors, including white, can be constructed. Those three
colors are not
transmitted individually by a television transmitter, but as a luminance
signal (Y) and the color
difference signals R-Y and B-Y. Both color difference signals R-Y and B-Y
are also called U-
and V signals, once adapted in amplitude.
The required bandwidth for the color signals is far less than for the Y-signal.
For TV signals,
the ratio between luminance (Y) and color signals is given as 4:1:1. The
required bandwidth
of the color signal is four times less than that of the luminance signal.
In professional studios,
one normally uses a ratio of 4:2:2, but due to the limited bandwidth of
a television transmitter,
it can not be broadcasted in this format.
When we want to digitise such a signal, we can only do it on a component
level, which
means that the video signal has to be split up.
For a sampling frequency of 13.5MHz and an eight bit quantising scheme,
we get the
following video bitstream :
Y-signal : 13.5MHz x 8 bit = 108Mbit
R-Y signal : 3.375MHz x 8 bit = 27Mbit
B-Y signal : 3.375MHz x 8 bit = 27Mbit
which totals to 162Mbit/s. To transmit this amount of data, a very high
bandwidth is required.
Was the bitstream for CD audio 1.4Mb/s, for a video signal this is about
100x higher.
MPEG VIDEO COMPRESSION
Introduction
When we have to start using digital source coding in television systems,
some international
agreements have to be made. This not only applies to video, but also audio,
the multiplexing
of video and audio as well as other signals like teletext.
Between 1988 and 1994 an international standard has been agreed on by MPEG,
a subgroup
of ISO and IEC.
The goal of MPEG was :
1.To produce a world wide standard for video and audio coding, with options
for various
applications
2.To define transmissions specifics that can be used for all media, including
transmission and recording
3.Defining a compliance procedure by which systems can be evaluated
4.Defining a datastructure that can be used to develop encoders and decoders
The first standard agreed on in 1992 was MPEG-1, used for computers and
CD-ROM. The
datastream here is 1.15Mbit/s. Picture quality is comparable to VHS recorders.
In november 1994, MPEG-2 was established. This not only enables a datastream
of
100Mbit/s but also it created the possibility to have multiple programs
in one single
datastream.
MPEG-2 is the basis for Digital Video Disc and Digital Video Broadcast.
For the European
market, the DVB project has established almost the complete system for
the new generation
of digital TV on both cable and satellite. This standard not only allows
data to be transported
more efficiently, bit also various new serviced can be exploited. DVB has
standarized the
scrambling of the signals and can add special information about things
like program
contents, transmission path etc. It not only allows the set top box to
configure itself but also
aids the viewer in finding programs.
MPEG-1
Name : ISO/IEC 11172
Bit rate : Usually 1.5Mbit/s
Video : Resolution CIF (354 pixels * 256 lines * 25Hz)
Audio : 64Kbit/s to 384 Kbit/s (Musicam)
Systems : Multiplexing 1 * video + stereo audio + data
Applications : CDI and computer games
MPEG-2
Name : 13818
Bit rate : Usually 2 - 15Mbit/s
Video : Resolution ITU-R 601 (720 pixels * 576 lines * 25Hz) and HDTV
Audio : 64Kbit/s to 384Kbit/s, stereo + surround (5 channels)
Systems : Multiplexing video, audio, data, conditional access, multiple
video signals,
each with their own timebase
Applications : Digital TV
To reduce the number of bits in a digital TV system, reduction and compression
techniques
have been developed to make this possible. It is called compression once
the picture image
in the decoder can be perfectly reconstructed. Reduction will allways show
a difference
between an original and a decoded image.
Compression techniques
By using the properties of the human senses like eyes and ears, it is possible
to apply a
hardly visible datareduction hence is acceptable for certain applications.
The basis is formed by leaving off information that can not be registered
by the eye
(irrelevance-reduction) because of which not always the same information
has to be
transmitted when the picture contents has not changed. This is called redundancy
reduction.
In MPEG-videocompression, multiple methods are used to reduce the number
of bits like :
Compression based on Discrete Cosinus Transformation (DCT)
Segmentation, splitting the image into blocks
Movement Compensation
Temporal prediction and interpolation
Adjacent pictures in a television signal are pretty much the same. Every
picture is built out of
2 frames which carry the same information. This is what we call redundancy
information.
There are several forms of redundancy :
Spatial redundancy
Temporal redundancy
Static redundancy
To understand redundancy, a couple of agreements have been made within
MPEG. A MPEG
decoder has various picture memories in which different frames for decoding
are stored and
out of which the original picture can be reconstructed.
This way, the bandwidth necessary for a single analogue television channel
can now contain
5 - 7 digital television channels using the MPEG-2 data compression. This
technique allows a
4-hour movie to be recorded on a double sided Digital Video Disc (DVD).
ADR / DMX DIGITAL RADIO
Digital Music Express (DMX) is a digitally coded radio signal complying
to the Astra Digital
Radio (ADR) specifications. The applied technique is constructed in a way
the standard
180KHz spacing of satellite audio transponders could be used. This enables
the
simultaneous transmission of both analogue and digital audio to assist
in a fluent transition
from analogue to digital. Per transponder, a maximum of 12 radio programmes
can be put
'behind' the television program, or a total of 48 radioprogrammes can be
transmitted on a
single transponder.
Important is, that a current analogue system needs two carriers to carry
both left- and right
channel for stereo transmissions in contrast to ADR and DMX where only
a single carries is
needed.
Firstly, the left- and right audio signal are digitised with a sampling
frequency of 48KHz / 16
bits resulting in an audio stream of 1.536Mbit/s. This has to be reduced
by a factor 8 to be
put in the narrow banded transmitter signal. This is accomplished using
the MUSICAM
encoder technique.
After the MUSICAM encoder, extra data like RDS and programme information
is added. The
encoder output delivers a datastream of 192Kbit/s.
