GSM Presentation7

Speech and Channel Coding

Channel Coding
The following figure shows the steps involved to transform speech audio to radio
waves and vice versa.

GSM Speech Processing Steps

Speech compressed using a predictive coding scheme Divided into blocks, each of
which is protected partly by cyclic code and partly by a convolutional code Inte
rleaving to protect against burst errors Encryption for providing privacy Assemb
led into time slots Modulated for analog transmission using GMSK

Speech coding
The GSM speech codec that transforms the analog signal (voice) into a digital re
presentation, must meet the following criteria: Maintain speech quality. Reduce
redundancy in voice utterances. This reduction is essential due to transmission
capacity limitation on the data channel. Adopt low complexity speech codec to re
duce production costs.

GSM TRANSMISSION PROCESS

STAGE 1: ANALOG TO DIGITAL (A/D) CONVERSION STAGE 2: SEGMENTATION STAGE 3: SPEEC
H CODING STAGE 4: CHANNEL CODING STAGE 5: INTERLEAVING STAGE 6: CIPHERING/ENCRYP
TION STAGE 7: BURST FORMATTING STAGE 8: MODULATION & TRANSMISSION

Speech Coding

In order to send the voice information across a radio network, first thing to be
done is to turn the voice into a digital signal. GSM uses a method called RPE-L
TP (Regular Pulse Excited - Long Term Prediction) with Linear Predictive Coding
to turn our analog voice into a compressed digital equivalent. One of the primar
y functions of an MS is to convert the analog speech information into digital fo
rm for transmission using a digital signal. The analog to digital (A/D) conversi
on process outputs a collection of bits: binary ones and zeros which represent t
he speech input.

In modern phone systems, digital coding is used. The electrical variations induc
ed into the microphone are sampled and each sample is then converted into a digi
tal code. The voice waveform sampled at a rate of 8 kHz and sample is converted
into an 8 bit binary number, representing 256 distinct values . Since we sample
8000 times per second and each sample is 8 binary bits, we have a bitrate of 8kH
z X 8 bits = 64kbps.

This bitrate is unrealistic to transmit across a radio network. GSM speech codin
g works to compress the speech waveform into a sample that results in a lower bi
trate using RPE-LTP. The speech signal is divided into blocks of 20ms . Once we
have a digital signal we have to add some sort of redundancy so that we can reco
ver from errors when we transmit our digital voice over the radio channel.

These blocks are then passed to the speech codec of 13 kbps, to obtain speech fr
ames of 260 bits each.

GSM Channel Coding

Once we have a digital signal we have to add some sort of redundancy so that we
can recover from errors when we transmit our digital voice over the radio channe
l. Channel coding add s redun da ncy bi ts to the original information to detect
and correct, errors occurred during transmission. GSM uses convol uti on coding
and int erlea ving to achieve this protection. The exact algorithms used differ
for speech and for different data rates

Channel Coding

In digital transmission, the quality of the transmitted signal is often expresse

d in terms of how many of the received bits are incorrect. This is called Bit Er
ror Rate (BER). BER defines the percentage of the total number of received bits
which are incorrectly detected.

This percentage should be as low as possible. It is not possible to reduce the p

ercentage to zero because the transmission path is constantly changing.

Channel coding is used to detect and correct errors in a received bit stream. It
adds bits to a message. These bits enable a channel decoder to determine whethe
r the message has faulty bits, and to potentially correct the faulty bits.

Channel coding for GSM speech

Recall that the RPE-LTP Encoder produces a block of 260 bits every 20 ms. It was
found (though testing) that some of the 260 bits were more important when compa
red to others. Below is the composition of these 260 bits. Cl as s Ia - 50 bits
(most sensitive to bit errors) Cl as s Ib - 132 bits (moderately sensitive to bi
t errors) Cl as s II - 78 bits (least sensitive to error)

As a result of some bits being more important than others, GSM adds redundancy b
its to each of the three Classes differently. The Class IA bits are encoded in a
cyclic encoder. The Class Ib bits (together with the encoded Class IA bits) are
encoded using convolutional encoding. Finally, the Class II bits are merely add
ed to the result of the convolutional encoder.

Class Ia bits have a 3 bit Cyclic Redundancy Code added for error detection. The
se 53 bits, together with the 132 Class Ib bits and a 4 bit tail sequence (a tot
al of 189 bits), are input into a rate convolutional encoder. Each input bit is
encoded as two output bits. The convolutional encoder thus outputs 378 bits, whi
ch are added to the 78 remaining Class II bits, which are unprotected. Thus ever
y 20 m sec speech sample is encoded as 456 bits, giving a bit rate of 22.8 kbps

Interleaving

To further protect against the burst errors common to the radio interface, each
sample is interleaved. This method rea rra nge s a group of bits in a particular
way. After encoding resultant sample block consists of 456 bits. These blocks a
re then divided into eight blocks each containing 57 bits. The first four blocks
will be placed in the even bit positions of the first four bursts. The last fou
r blocks will be placed in the odd bit positions of the next four bursts.

Because of interleaving lost bits are part of several different packets and each
packet loses only a few bits out of a large number of bits. So Interleaving dec
reases the possibility of losing whol e bur st s during the transmission, by di
spers in g the errors. Since the errors become less concentrated, it is then eas
ier to correct them.

Encryp ti on
It is used to protect signaling and data. This process is done using A3, A5 and
A8 algorithms

Modula tion The modulation chosen for the GSM system is the Ga us si an M ini mu
m Shi ft K eyin g (GMSK ) .

Discontinuous Transmission (DTX)

Discontinuous Transmission (DTX) is a method of saving battery power for the MS.
An MS with the DTX function detects the input "voice" and turns the transmitter
ON only while "voice is present. When there is no voice input, the transmitter i
s turned OFF.

Discontinuous transmission (DTX)

So DTX is used to suspend the radio transmission during the si lence periods. Th
is exploits the observation that only 4050% during a conversation does the speak
er actually talk. DTX helps also to reduce interference between different cells
and to increase system capacity. An added benefit of DTX is that power is conser
ved at the mobile unit.

Voic e A ctivi ty D etection (VAD)

The DTX function is performed by means of VAD It is this which has to determine
whether the sound represents speech or noise, even if the background noise is ve
ry important. If the voice signal is considered as noise, the transmitter is tur
ned off producing then, an unpleasant effect called clipping.

Co mfort n oise

A side-effect of the DTX function is that when the signal is considered as noise
, the transmitter is turned off and therefore, a total silence is heard at the r
eceiver. This can be very annoying to the receiving user since it appears as a d
ead connection. In order to overcome this problem, the receiver creates a minimu
m of background noise called comfort noise. Comfort noise eliminates the impress
ion that the connection is dead.

Pow er con trol

To minimize co-channel interference and to conserve power, both the mobiles and
the Base Transceiver Stations operate at the lowest power level that will mainta
in an acceptable signal quality. The BTSs perform timing measurements; they also
perform measurements on the power level of the different mobile stations. These
power levels are adjusted so that the power is nearly the same for each burst.
The BTS controls its power level. The MS measures the strength and the quality o
f the signal between itself and the BTS. If the mobile station does not receive
correctly the signal, the BTS changes its power level and retransmits.

Dis con tin uou s r ecep tion

Another method used to conserve power at the MS is Discontinuous Reception (DRX)

. The paging channel, used by the BTS to signal an incoming call, is structured
into subchannels. Each MS is assigned one of these subchannels and needs to list
en only to its own sub-channel. In the time between successive paging subchannel
s, the mobile can go into sleep mode, when almost no power is used.

Timing Advance

In the GSM cellular mobile phone standard, timi ng ad vance value corresponds to
the length of time a signal from the mobile phone takes to reach the base stati
on. GSM uses TDMA technology in the radio interface to share a single frequency
between several users, assigning sequential timeslots to the individual users sh
aring a frequency. Each user transmits periodically one-eighth of the time withi
n one of the eight timeslots. Since the users are various distances from the bas
e station and radio waves travel at the finite speed of light, the precise time
at which the phone is allowed to transmit a burst of traffic within a timeslot m
ust be adjusted accordingly. Timing Advance (TA) is the variable controlling thi
s adjustment.

This synchronization between BTS and MS is achieved by using the concept of Timi
ng Advance (TA). From the measurements, the BTS can calculate the Timing-Advance
and send it back to the MS in the first downlink transmission. From the TA valu
e received from the BTS, the MS know when to send the frame, so that it can arri
ve at the BTS in synchronism. The values of the TA is continuously calculated an
d transmitted to the MS during the call.

TRANSMISSION RATE

The amount of information transmitted over a radio channel over a period of time
is known as the transmission rate. Transmission rate is expressed in bits per s
econd or bit/s. In GSM the net bit rate over the air interface is 270kbit/s.