JPR Satcom-Unit 4 Notes
JPR Satcom-Unit 4 Notes
JPR Satcom-Unit 4 Notes
UNIT IV
Communications satellites are used to carry telephone, video, and data signals, and can use both
analog and digital modulation techniques.
Modulation:
Multiplexing:
Task of multiplexing is to assign space, time, frequency, and code to each communication
channel with a minimum of interference and a maximum of medium utilization Communication
channel refers to an association of sender(s) and receiver(s) that want to exchange data one of
several constellations of a carrier’s parameters defined by the used modulation scheme.
The modulation and multiplexing techniques that were used at this time were analog, adapted
from the technology developed for the change to digital voice signals made it easier for long-
distance.
Primarily for video provided that a satellite link's overall carrier-to-noise but in to older receiving
equipment at System and Satellite Specification Ku-band satellite parameters.
In analog television (TV) transmission by satellite, the baseband video signal and one or two
audio subcarriers constitute a composite video signal.
Digital modulation is obviously the modulation of choice for transmitting digital data are
digitized analog signals may conveniently share a channel with digital data, allowing a link to
carry a varying mix of voice and data traffic.
Digital signals from different channels are interleaved for transmission through time division
multiplexing TDM carry any type of traffic â€‖ the bent pipe transponder that can carry voice,
video, or data as the marketplace demands.
Hybrid multiple access schemes can use time division multiplexing of baseband channels which
are then modulate.
Also Transmission—transmits without regard to signal content vs. being concerned with signal
content. Difference in how attenuation is handled, but not focus on this.
Must convert digital data to analog signal such device is a modem to translate between bit-serial
and modulated carrier signals?
• To send digital data using analog technology, the sender generates a carrier signal at some
continuous tone (e.g. 1-2 kHz in phone circuits) that looks like a sine wave. The following
techniques are used to encode digital data into analog signals.
• Amplitude-shift modulation (keying): vary the amplitude (e.g. voltage) of the signal. Used to
transmit digital data over optical fiber.
• Frequency-shift modulation: two (or more tones) are used, which are near the carrier
frequency. Used in a full-duplex modem (signals in both directions).
• Phase-shift modulation: systematically shift the carrier wave at uniformly spaced intervals.
For instance, the wave could be shifted by 45, 135, 225, 315 degree at each timing mark. In this
case, each timing interval carries 2 bits of information.
Why not shift by 0, 90, 180, 270? Shifting zero degrees means no shift, and an extended set of
no shifts leads to clock synchronization difficulties.
Frequency division multiplexing (FDM): Divide the frequency spectrum into smaller sub
channels, giving each user exclusive use of a sub channel (e.g., radio and TV). One problem with
FDM is that a user is given all of the frequency to use, and if the user has no data to send,
bandwidth is wasted — it cannot be used by another user.
Time division multiplexing (TDM): Use time slicing to give each user the full bandwidth, but
for only a fraction of a second at a time (analogous to time sharing in operating systems). Again,
if the user doesn’t have data to sent during his time slice, the bandwidth is not used (e.g.,
wasted).
Statistical multiplexing: Allocate bandwidth to arriving packets on demand. This leads to the
most efficient use of channel bandwidth because it only carries useful data.That is, channel
bandwidth is allocated to packets that are waiting for transmission, and a user generating no
packets doesn’t use any of the channel resources.
Digital Video Broadcasting (DVB) has become the synonym for digital television and for data
broadcasting world-wide.
DVB services have recently been introduced in Europe, in North- and South America, in Asia,
Africa and Australia.
This article aims at describing what DVB is all about and at introducing some of the technical
background of a technology that makes possible the broadcasting.
4.2 DUPLEXING
For voice or data communications, must assure two way communication (duplexing, it is
possible to talk and listen simultaneously). Duplexing may be done using frequency or time
domain techniques.
Because transmission from mobile to BS and from BS to mobile alternates in time, this scheme
is also known as ―ping pong‖.
As a consequence of the use of the same frequency band, the communication quality in both
directions is the same. This is different from FDD.
With single access, a single modulated carrier occupies the whole of the available bandwidth of a
transponder. Single-access operation is used on heavy-traffic routes and requires large earth
station antennas.
The earth station employs a 30-m-diameter antenna and a parametric amplifier, which together
provide a minimum [ G/ T] of 37.5 dB/K.
The transmission from the BS in the downlink can be heard by each and every mobile user in the
cell, and is referred as broadcasting. Transmission from the mobile users in the uplink to the BS
is many-to-one, and is referred to as multiple access.
Multiple access schemes to allow many users to share simultaneously a finite amount of radio
spectrum resources. Should not result in severe degradation in the performance of the system as
compared to a single user scenario. Approaches can be broadly grouped into two categories:
narrowband and wideband.
Multiple Accessing Techniques : with possible conflict and conflict- free
SSMA PAMA
Spread Spectrum Pulse Address
Multiple Access Multiple Access
The most widely used of the multiple access techniques is FDMA. In FDMA, the available
satellite bandwidth is divided into portions of non-overlapping frequency slots which are
assigned exclusively to individual earth stations. A basic diagram of an FDMA satellite system is
shown in Fig.
In FDMA, each user is allocated a unique frequency band or channel. During the period of the
call, no other user can share the same frequency band.
All channels in a cell are available to all the mobiles. Channel assignment is carried out on a
first-come first- served basis.
The number of channels, given a frequency spectrum BT, depends on the modulation
technique (hence Bw or Bc ) and the guard bands between the channels guard.
These guard bands allow for imperfect filters and oscillators and can be used to minimize
adjacent channel interference.
In FDMA systems, multiple signals from the same or different earth stations with different
carrier frequencies are simultaneously passed through a satellite transponder. Because of the
nonlinear mode of the transponder, FDMA signals interact with each other causing
intermodulation products (intermodulation noise) which are signals at all combinations of
sum and difference frequencies as shown in the example given in Fig.
Forms of DMA:
Frequency slots may be preassigned to analog and digital signals, and to illustrate the method,
analog signals in the FDM/FM/FDMA format will be considered first. As the acronyms indicate,
the signals are frequency-division multiplexed, frequency modulated (FM), with FDMA to the
satellite.
For Ex, Consider each earth station will be assumed to transmit a 60-channel supergroup. Each
60-channel supergroup is then frequency modulated onto a carrier which is then upconverted to a
frequency in the satellite uplink band.
one in Ottawa, one in New York, and one in London. All three earth stations access a single
satellite transponder channel simultaneously, and each communicates with both of the others.
Thus it is assumed that the satellite receive and transmit antenna beams are global, encompassing
all three earth stations. Each earth station transmits one uplink carrier modulated with a 60-
channel supergroup and receives two similar downlink carriers.
As in the preassigned access mode, carriers may be frequency modulated with analog
information signals, these being designated FM/SCPC, or they may be phase modulated with
digital information signals, these being designated as PSK/SCPC.
Demand assignment may be carried out in a number of ways.
1. In the polling method, a master earth station continuously polls all the earth stations in
sequence, and if a call request is encountered, frequency slots are assigned from the pool
of available frequencies. The polling delay with such a system tends to become excessive
as the number of participating earth stations increases.
2. Instead of using a polling sequence, earth stations may request calls through the master
earth station as the need arises. This is referred to as centrally controlled random
access. The requests go over a digital orderwire, which is a narrowband digital radio link
or a circuit through a satellite transponder reserved for this purpose. Frequencies are
assigned, if available, by the master station, and when the call is completed, the
frequencies are returned to the pool. If no frequencies are available, the blocked call
requests may be placed in a queue, or a second call attempt may be initiated by the
requesting station.
SPADE System:
The CSC bandwidth is 160 kHz, and its center frequency is 18.045 MHz below the pilot
frequency, as shown in Fig..
Signaling information is routed through the CSC. Each earth station has DASS unit
(Demand Assignment Signaling and Switching unit). It is used to perform the functions
needed by the CSC.
To avoid interference with the CSC, voice channels 1 and 2 are left vacant, and to
maintain duplex matching, the corresponding channels 1 and 2 are also left vacant.
Recalling from previous secetion that channel 400 also must be left vacant, this requires
that channel 800 be left vacant for duplex matching.
Thus six channels are removed from the total of 800, leaving a total of 794 one-way or
397 full-duplex voice circuits, the frequencies in any pair being separated by 18.045
MHz, as shown in Fig. . (An alternative arrangement is shown in Freeman, 1981.)
All the earth stations are permanently connected through the CSC.
This is shown diagrammatically in Fig. for six earth stations A, B, C, D, E, and F.
Each earth station has the facility for generating any one of the 794 carrier frequencies
using frequency synthesizers.
Furthermore, each earth station has a memory containing a list of frequencies available
and the list is continuously updated through CSC.
TDMA systems divide the channel time into frames. Each frame is further partitioned into
time slots. In each slot only one user is allowed to either transmit or receive.
Unlike FDMA, only digital data and digital modulation must be used.
Each user occupies a cyclically repeating time slot, so a channel may be thought of as a
particular time slot of every frame, where N time slots comprise a frame.
With TDMA, only one carrier uses the transponder at any one time, and therefore,
intermodulation products, which result from the nonlinear amplification of multiple carriers,
are absent.
This leads to one of the most significant advantages of TDMA, which is that the TWT can be
operated at maximum power output or saturation level.
Because the signal information is transmitted in bursts, TDMA is only suited to digital
signals.
Digital data can be assembled into burst format for transmission and reassembled from the
received bursts through the use of digital buffer memories.
Figure illustrates the basic TDMA concept, in which the stations transmit bursts in sequence.
Burst synchronization is required, and in the system illustrated in Fig. 14.10, one station is
assigned solely for the purpose of transmitting reference bursts to which the others can be
synchronized.
The time interval from the start of one reference burst to the next is termed a frame. A frame
contains the reference burst R and the bursts from the other earth stations, these being shown
as A, B, and C in Fig. 14.10.
Fig 4.17 Time-division multiple access (TDMA) using a reference station for burst
synchronization.
Figure illustrates the basic principles of burst transmission for a single channel.
Overall, the transmission appears continuous because the input and output bit rates are
continuous and equal.
However, within the transmission channel, input bits are temporarily stored and transmitted
in bursts. Since the time interval between bursts.
Time interval between Burst is the frame time TF, The required Buffer capacity is
M= RbTF
The M bits are transmitted in the burst time TB, and the transmission rate (Burst Bit Rate) is
RTDMA= M/Tb
= M (TF/Tb)
It is also referred to as Burst Rate.
TDMA – Transmission
In TDMA, the transmit timing of the bursts is accurately synchronized so that the
transponder receives one burst at a time.
Each earth station receives an entire burst stream and extracts the bursts intended for it.
A frame consists of a number of bursts originating from a community of earth stations in a
network.
A TDMA frame structure is shown in Fig.
Where,
G – Guard Time
CBR - Carrier and Bit Timing Recovery
BCW - Burst Code Word
SIC - Station Identification Code
Q - Postamble
Fig 4.19 TDMA Frame Format
It consists of two reference bursts RB1and RB2, traffic bursts and the guard time between
bursts.
As can be seen, each TDMA frame has two reference bursts RB1 and RB2.
The primary reference burst (PRB), which can be either RB1 or RB2, is transmitted by one of
the earth stations in the network designated as the primary reference earth station.
For reliability, a second reference burst (SRB) is transmitted by a secondary reference earth
station.
To ensure an undisrupted service for the TDMA network, automatic switchover between
these two reference stations is provided.
The reference bursts carry no traffic information and are used to provide synchronization for
all earth stations in the network.
The traffic bursts carry information from the traffic earth station.
Each earth station accessing a transponder may transmit one or two traffic bursts per TDMA
frame and may position them anywhere in the frame according to a burst time plan that
coordinates traffic between earth stations in the network.
The Guard time between bursts ensures that the bursts never overlap at the input to the
transponder.
The TDMA bursts structure of the reference and traffic burst are given in Fig
Various sequences in the reference burst and traffic burst are as follows:
The channel also carries monitoring and control information to the traffic stations. The SC of
the traffic burst carries the traffic station’s status to the reference station (value of transmit
delay used and reference station from which the delay is obtained). It also contains other
information such as the high bit error rate and UW loss alarms to other traffic stations. The
INTELSAT V TDMA has an 8-symbol SC for each of the bursts.
a) Reference Burst
b) Traffic Burst
6. Traffic data
This portion contains the information from a source traffic station to a destination traffic
station. The informants can be voice, data, video or facsimile signals. The traffic data pattern
is divided into blocks of data (referred to as subburst).
The INTELSAT TDMA with a frame length of T f = 2 msec for PCM voice data has a subburst
size of 64 symbols long.
TDMA- Features:
Burst transmission since channels are used on a time sharing basis. Transmitter can be turned
off during idle periods.
High ISI – Higher transmission symbol rate, hence resulting in high ISI. Adaptive equalizer
required.
A guard time between the two time slots must be allowed i n order to avoid interference,
especially in the uplink direction. All mobiles should synchronize with BS to minimize
interference.
Efficient power utilization: FDMA systems require a 3- to 6-dB power back off in order to
compensate for inter-modulation effects.
Efficient handoff: TDMA systems can take advantage of the fact that the transmitter is
switched off during idle time slots to improve the handoff procedure. An enhanced link
control, such as that provided by mobile assisted handoff (MAHO) can be carried out by a
subscriber by listening to neighboring base station during the idle slot of the TDMA frame.
Efficiency of TDMA
Efficiency of TDMA is a measure of the percentage of bits per frame which contain
transmitted data. The transmitted data include source and channel coding bits.
2020 - 2021 18 Jeppiaar Institute of Technology
EC8094: Satellite Communication Department of ECE
bOH includes all overhead bits such as preamble, guard bits, etc.
CDMA Features:
Spreading signal (code) consists of chips
Has Chip period and hence, chip rate
Spreading signal use a pseudo-noise (PN) sequence (a pseudo-random sequence)
PN sequence is called a code word
Each user has its own cord word
Code words are orthogonal. (low autocorrelation)
Chip rate is order of magnitude larger than the symbol rate.
The receiver correlator distinguishes the senders signal by examining the wideband signal
with the same time-synchronized spreading code
The sent signal is recovered by dispreading process at the receiver.
CDMA Advantages:
Low power spectral density.
Signal is spread over a larger frequency band
Other systems suffer less from the transmitter
Privacy
The code word is known only between the sender and receiver. Hence other users cannot
decode the messages that are in transit
CDMA Data:
DSSS Transmitter:
DSSS Receiver
FDMA/CDMA
Available wideband spectrum is frequency divided into number narrowband radio channels.
CDMA is employed inside each channel.
DS/FHMA
The signals are spread using spreading codes (direct sequence signals are obtained), but these
signal are not transmitted over a constant carrier frequency; they are transmitted over a frequency
hopping carrier frequency.
Each cell is using a different spreading code (C DMA employed between cellss) that is conveyed
to the mobiles in its range. Inside each cell (inside a CDMA channel), TDMA is employed to
multiplex multiple users.
At each time slot, the user is hopped to a new frequency according to a pseudo-random
hopping sequence.
Employed in severe co-interference and multi-path environments. Bluetooth and GSM
are using this technique
A large number of independently steered high-gain beams can be formed without any
resulting degradation in SNR ratio.
Beams can be assigned to individual users, thereby assuring that all links operate with
maximum gain.
Adaptive beam forming can be easily implemented to improve the system capacity by
suppressing co channel interference.
Advantage of CDMA
Disadvantages of CDMA:
Satellite transponders are channelized too narrowly for roadband CDMA, which is the
most attractive form of CDMA.
Power control cannot be as tight as it is in a terrestrial system because of long round-trip
delay.
In radio resource management for wireless and cellular network, channel allocation schemes are
required to allocate bandwidth and communication channels to base stations, access points and
terminal equipment.
Fixed: FCA, fixed channel allocation: Manually assigned by the network operator
Dynamic:
In Fixed Channel Allocation or Fixed Channel Assignment (FCA) each cell is given a
predetermined set of frequency channels.
FCA requires manual frequency planning, which is an arduous task in TDMA and FDMA based
systems, since such systems are highly sensitive to co-channel interference from nearby cells that
are reusing the same channel. This result in traffic congestion and some calls being lost when
traffic gets heavy in some cells, and idle capacity in other cells.
Dynamic Frequency Selection (DFS) may be applied in wireless networks with several
adjacent non-centrally controlled access points.
A more efficient way of channel allocation would be Dynamic Channel Allocation or Dynamic
Channel Assignment (DCA) in which voice channel are not allocated to cell permanently,
instead for every call request base station request channel from MSC.
Spread Spectrum:
Thus the frequency channel allocation problem is relaxed in cellular networks based on a
combination of Spread spectrum and FDMA, for example IS95 and 3G systems.
In packet based data communication services, the communication is bursty and the traffic load
rapidly changing. For high system spectrum efficiency, DCA should be performed on a packet-
by-packet basis.
Examples of algorithms for packet-by-packet DCA are Dynamic Packet Assignment (DPA),
Dynamic Single Frequency Networks (DSFN) and Packet and resource plan scheduling
(PARPS).
Techniques known since the 1940s and used in military communication systems since the 1950s
"spread" a radio signal over a wide frequency range several magnitudes higher than minimum
requirement.
Resistance to fading. The high bandwidth occupied by spread-spectrum signals offer some
frequency diversity, i.e. it is unlikely that the signal will encounter severe multipath fading over
its whole bandwidth, and in other cases the signal can be detected using e.g. a Rake receiver.
4.5 COMPRESSION
Lossless Compression :
where data is compressed and can be reconstituted (uncompressed) without loss of detail or
information. These are referred to as bit-preserving or reversible compression systems also.
Lossy Compression :
where the aim is to obtain the best possible fidelity for a given bit-rate or minimizing the bit-rate
to achieve a given fidelity measure. Video and audio compression techniques are most suited to
this form of compression.
Lossy compression use source encoding techniques that may involve transform encoding,
differential encoding or vector quantization.
MPEG is a group within the International Standards Organization and the International
Electrochemical Commission (ISO/IEC) that undertook the job of defining standards for the
transmission and storage of moving pictures and sound.
The standards are concerned only with the bit stream syntax and the decoding process, not with
how encoding and decoding might be implemented. Syntax covers matters such as bit rate,
picture resolution, time frames for audio, and the packet details for transmission.
The design of hardware for the encoding and decoding processes is left to the equipment
manufacturer. The MPEG standards currently available are MPEG-1, MPEG-2, MPEG-4,
MPEG-7 and MPEG-21.
Standard Application
MPEG-1 The original standard for encoding and decoding streaming video and audio
files.
MPEG-2 The standard for digital television, this compresses files for transmission of
high-quality video.
MPEG-4 The standard for compressing high-definition video into smaller-scale files that
stream to computers, cell phones and PDAs (personal digital assistants).
MPEG-7 It is a multimedia content description standard This description will be
associate with the content itself, to allow fast and efficient searching for
material that is of interest to the user.
MPEG-21 It is also referred to as the Multimedia Framework. The standard that interprets
what digital content to provide to which individual user so that media plays
flawlessly under any language, machine or user conditions
MPEG – 1
In Digital Broadcast Service (DBS) systems, MPEG-1 is used for audio compression, and
MPEG-2 is used for video compression. Both of these MPEG standards cover audio and video,
but MPEG-1 video is not designed for DBS transmissions.
MPEG-1 audio supports mono and two channel stereo only, which is considered adequate for
DBS systems currently in use.
MPEG-2 audio supports multichannel audio in addition to mono and stereo. It is fully compatible
with MPEG-1 audio.
The need for audio compression can be seen by considering the bit rate required for high-quality
audio. The bit rate is equal to the number of samples per second (the sampling frequency fs)
multiplied by the number of bits per sample n:
For a stereo CD recording, the sampling frequency is 44.1 kHz, and the number of bits per
sample is 16:
The factor 2 appears on the right-hand side because of the two channels in stereo. This bit rate,
approximately 1.4 Mb/s, represents too high a fraction of the total bit rate allowance per channel,
and hence the need for audio compression.
Audio compression in MPEG exploits certain perceptual phenomena in the human auditory
system. In particular, it is known that a loud sound at one particular frequency will mask a less
intense sound at a nearby frequency.
For example, consider a test conducted using two tones, one at 1000 Hz, which will be called the
masking tone and the other at 1100 Hz, the test tone. Starting with both tones at the same level,
say, 60 dB above the threshold of hearing, if now the level of the 1000Hz tone is held constant
while reducing the level of the 1100-Hz tone, a point will be reached where the 1100-Hz tone
becomes inaudible. The 1100-Hz tone is said to be masked by the 1000-Hz tone.
Assume for purposes of illustration that the test tone becomes inaudible when it is 18 dB below
the level of the masking tone. This 18 dB is the masking threshold. It follows that any noise
below the masking threshold also will be masked. For the moment, assuming that only these two
tones are present, then it can be said that the noise floor is 18 dB below the masking tone. If the
test-tone level is set at, say, 6 dB below the masking tone, then of course it is 12 dB above the
noise floor. This means that the signal-to noise ratio for the test tone need be no better than 12
dB. Now in a pulsecode modulated (PCM) system the main source of noise is that arising from
the quantization process.
This shows that increasing n by 1 bit increases the signal-to-quantization noise ratio by 6 dB.
Another way of looking at this is to say that a 1-bit decrease in n increases the quantization noise
by 6 dB.
In the example above where 12 dB is an adequate signal-to-noise ratio, the above Equation
shows that only 2 bits are needed to encode the 1100-Hz tone (i.e., the levels would be quantized
in steps represented by 00, 01, 10, 11). By way of contrast, the CD samples taken at a sampling
frequency of 44.1 kHz are quantized using 16 bits to give a signal-to-quantization noise ratio of
96 dB. Returning to the example of two tones, in reality, the audio signal will not consist of two
single tones but will be a complex signal covering a wide spectrum of frequencies.
In MPEG-1, two processes take place in parallel, as illustrated in Fig. The filter bank divides the
spectrum of the incoming signal into sub bands. In parallel with this the spectrum is analyzed to
permit identification of the masking levels. The masking information is passed to the quantizer,
which then quantizes the sub bands according to the noise floor.
The masking discussed so far is referred to as frequency masking for the reasons given earlier. It
is also an observed phenomenon that the masking effect lasts for a short period after the masking
signal is removed. This is termed temporal masking, and it allows further compression in that it
extends the time for which the reduction in quantization applies.
The compressed bit rate for MPEG-1 audio used in DBS systems is 192 kb/s.
MPEG -2
In DBS systems, MPEG-2 is used for video compression. As a first or preprocessing step, the
analog outputs from the red (R), green (G), and blue (B) color cameras are converted to a
luminance component (Y) and two chrominance components (Cr) and (Cb). This is similar to the
analog NTSC arrangement
In matrix notation, the equation relating the three primary colors to the Y, Cr, and Cb
components is
[ ] [ ][ ]
It is an observed fact that the human eye is less sensitive to resolution in the color components
(Cr and Cb) than the luminance (Y) component. This allows a lower sampling rate to be used for
the color components. This is referred to as chroma subsampling, and it represents one step in the
compression process.
Sampling is usually indicated by the ratios Y: U: V where Y represents the luminance (or luma)
sampling rate, U the Cb sampling rate, and V the Cr sampling rate. The values for YUV are
normalized to a value of 4 for Y, and ratios commonly encountered with digital TV are 4:4:4,
4:2:2 and 4:2:0.
4:4:4 means that the sampling rates of Y, Cb, and Cr are equal. Each pixel would get three
digital words, one for each of the component signals. If the words are 8-bits then each pixel
would be encoded in 3 bytes.
4:2:2 means that the Cb and Cr signals are sampled at half the rate of the Y signal component.
Every two pixels would have two bytes for the Y signal, one byte for the Cb signal and one byte
for the Cr signal, resulting in 4 bytes for the 2-pixel block.
4:2:0 means that Cb and Cr are sampled at half the Y sampling rate, but they are sampled only
on alternate scan lines. Thus vertical as well as horizontal resolution is reduced by half. A 2 * 2
pixel block would have 6 bytes, 4 bytes for Y, 1 byte for Cb and 1 byte for Cr. MPEG-2 uses
4:2:0 sampling.
Following the digitizer, difference signals are formed, and the discrete cosine transform (DCT)
block converts these to a ―spatial frequency‖ domain. The familiar Fourier transform transforms
a time signal g(t) to a frequency domain representation G(f), allowing the signal to be filtered in
the frequency domain. Here, the variables are time t and frequency f. In the DCT situation, the
input signals are functions of the x (horizontal) and y (vertical) space coordinates, g(x, y). The
DCT transforms these into a domain of new variables u and v, G(u, v). The variables are called
spatial frequencies in analogy with the time-frequency transform. It should be noted that g(x, y)
and G(u, v) are discrete functions.
In the quantizer following the DCT transform block, the discrete values of G(u, v) are quantized
to predetermined levels. This reduces the number of levels to be transmitted and therefore
provides compression. The components of G(u, v) at the higher spatial frequencies represent
finer spatial resolution. The human eye is less sensitive to resolution at these high spatial
frequencies; therefore, they can be quantized in much coarser steps. This results in further
compression.
Compression is also achieved through motion estimation. Frames in MPEG-2 are designated I, P,
and B frames, and motion prediction is achieved by comparing certain frames with other frames.
The I frame is an independent frame, meaning that it can be reconstructed without reference to
any other frames. A P (for previous) frame is compared with the previous I frame, and only those
parts which differ as a result of movement need to be encoded. The comparison is carried out in
sections called macroblocks for the frames. A macroblock consists of 16 * 16 pixels. A B (for
bidirectional) frame is compared with the previous I or P frame and with the next P frame. This
obviously means that frames must be stored in order for the forward comparison to take place.
Only the changes resulting from motion are encoded, which provides further compression.
An estimate of the compression required can be made by assuming a value of 200 Mb/s for the
uncompressed bit rate for SDTV, and taking 5 Mb/s as typical of that for a TV channel, the
compression needed is on the order 200/5 = 40:1. The 5 Mb/s would include audio and data, but
these should not take more than about 200 kb/s.
The whole encoding process relies on digital decision-making circuitry and is computationally
intensive and expensive. The decoding process is much simpler because the rules for decoding
are part of the syntax of the bit stream. Decoding is carried out in the integrated receiver decoder
(IRD) unit.
MPEG – 4:
MPEG-4 was developed jointly by the Video Coding Experts/Group (VCEG) of the International
Telecommunication Union (ITU), Telecommunication Standardization Sector (ITU-T) which
uses the designation H.264, and the MPEG of the ISO/IEC.
This version of MPEG is known by at least six different names (H.264, H.26L, ISO/IEC 14496-
10, JVT, MPEG-4 AVC, and MPEG-4 Part 10) and the abbreviation AVC is commonly used to
denote advanced video coding. Following the usage in Sullivan et al., it will be denoted here by
H.264/AVC. Areas of application include video telephony, video storage and retrieval (DVD and
hard disk), digital video broadcast, and others.
In general terms, MPEG-4 provides many features not present with other compression schemes,
such as interactivity for viewers, where objects within a scene can be manipulated, but from the
point of view of satellite television, the major advantage is the reduction in bit rate offered.
About a 2:1 reduction in bit rate, on average is achievable with H.264/AVC compared with
MPEG-2. Fidelity Range Extensions (FRExt) was added to H.264/AVC that can provide a
reduction of as much as 3:1 in certain situations.
FRExt supports 4:2:2 and 4:4:4 sampling. As with MPEG-2 the analog outputs from the red (R),
green (G), and blue (B) color cameras are converted to a luminance component (Y) and two
chrominance components (Cr) and (Cb) but with a different M matrix, this being:
[ ] [ ][ ]
It follows that any format conversion would require a matrix recalculation. H.264/AVC takes
advantage of the increases in processing power available from computer chips, but at the cost of
more expensive equipment, both for the TV broadcaster and the consumer.
As with MPEG-2, frames are compared for changes through comparing macro blocks of 16 * 16
pixels, but H.264/AVC also allows for comparisons of sub macro blocks of pixel groups 16 * 8,
8 * 16, 8 * 8, 8 * 4, 4 * 8, and 4 * 4. At present it is not backward compatible with MPEG-2,
which may present a problem with some high definition TV.
At the broadcast center, the high-quality digital stream of video goes through an MPEG encoder,
which converts the programming to MPEG-4 video of the correct size and format for the satellite
receiver in your house.
Encoding works in conjunction with compression to analyze each video frame and eliminate
redundant or irrelevant data and extrapolate information from other frames. This process reduces
the overall size of the file. Each frame can be encoded in one of three ways:
As an intraframe, which contains the complete image data for that frame. This method provides
the least compression.
As a predicted frame, which contains just enough information to tell the satellite receiver how to
display the frame based on the most recently displayed intraframe or predicted frame
This process occasionally produces artifacts -- glitches in the video image. One artifact is
macroblocking, in which the fluid picture temporarily dissolves into blocks.
Macroblocking is often mistakenly called pixilating, a technically incorrect term which has been
accepted as slang for this annoying artifact.
There really are pixels on your TV screen, but they're too small for your human eye to perceive
them individually -- they're tiny squares of video data that make up the image you see. (For more
information about pixels and perception,
The rate of compression depends on the nature of the programming. If the encoder is converting
a newscast, it can use a lot more predicted frames because most of the scene stays the same from
one frame to the next.
In more fast-paced programming, things change very quickly from one frame to the next, so the
encoder has to create more intraframes. As a result, a newscast generally compresses to a smaller
size than something like a car race.
After the video is compressed, the provider encrypts it to keep people from accessing it for free.
Encryption scrambles the digital data in such a way that it can only be decrypted (converted
back into usable data) if the receiver has the correct decryption algorithm and security keys.
Once the signal is compressed and encrypted, the broadcast center beams it directly to one of its
satellites. The satellite picks up the signal with an onboard dish, amplifies the signal and uses
another dish to beam the signal back to Earth, where viewers can pick it up.
Video and Audio files are very large beasts. Unless we develop and maintain very high
bandwidth networks (Gigabytes per second or more) we have to compress to data.
Relying on higher bandwidths is not a good option -- M25 Syndrome: Traffic needs ever
increases and will adapt to swamp current limit whatever this is.
As we will compression becomes part of the representation or coding scheme which have
become popular audio, image and video formats. Some popular coding techniques are
1. Entrophy Encoding
Repetitive Sequence Supression
Zero Length Suppresion
Run Length Encoding
Statistical Encoding
Pattern Substitution
Shannon Fano & Huffman Coding
2. Source Coding
Transform Coding
FFT Fast Fourier Transform
ENCRYPTION:
It is the most effective way to achieve data security. To read an encrypted file, you must have
access to a secret key or password that enables you to decrypt it. Unencrypted data is called
plain text ; encrypted data is referred to as cipher text.
In public-key encryption schemes, the encryption key is published for anyone to use and encrypt
messages. However, only the receiving party has access to the decryption key that enables
messages to be read.
DECRYPTION:
It is the process of taking encoded or encrypted text or other data and converting it back into text
that you or the computer are able to read and understand.
This term could be used to describe a method of un-encrypting the data manually or with un-
encrypting the data using the proper codes or keys.
Data may be encrypted to make it difficult for someone to steal the information. Some
companies also encrypt data for general protection of company data and trade secrets. If this data
needs to be viewable, it may require decryption