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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
 Interleaving to protect against burst errors
 Encryption for providing privacy
 Assembled into time slots
 Modulated for analog transmission using
GMSK
Speech coding
The GSM speech codec that transforms the
analog signal (voice) into a digital representation,
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 reduce
production costs.
GSM TRANSMISSION PROCESS
 STAGE 1: ANALOG TO DIGITAL (A/D)
CONVERSION
 STAGE 2: SEGMENTATION
 STAGE 3: SPEECH CODING
 STAGE 4: CHANNEL CODING
 STAGE 5: INTERLEAVING
 STAGE 6: CIPHERING/ENCRYPTION
 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-LTP (Regular Pulse
Excited - Long Term Prediction) with Linear Predictive
Coding to turn our analog voice into a compressed
digital equivalent.
 One of the primary functions of an MS is to convert
the analog speech information into digital form for
transmission using a digital signal.
 The analog to digital (A/D) conversion process
outputs a collection of bits: binary ones and zeros
which represent the speech input.
 In modern phone systems, digital coding is used.
 The electrical variations induced into the microphone
are sampled and each sample is then converted into
a digital 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 8kHz X 8
bits = 64kbps.
 This bitrate is unrealistic to transmit across a radio
network.
 GSM speech coding works to compress the speech
waveform into a sample that results in a lower bitrate
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 recover 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 frames 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 channel.
 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 expressed in terms
of how many of the received bits are
incorrect.
 This is called Bit Error 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 percentage 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 whether 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 compared 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 bit
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 bits 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 added to
the result of the convolutional encoder.
 Class Ia bits have a 3 bit Cyclic Redundancy Code
added for error detection.
 These 53 bits, together with the 132 Class Ib bits
and a 4 bit tail sequence (a total of 189 bits), are
input into a ½ rate convolutional encoder.
 Each input bit is encoded as two output bits.
 The convolutional encoder thus outputs 378 bits,
which are added to the 78 remaining Class II bits,
which are unprotected.
 Thus every 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 are 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 four 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 decreases 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 easier 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 mum 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
is turned OFF.
Discontinuous transmission (DTX)
 So DTX is used to suspend the radio
transmission during the si lence periods.
 This exploits the observation that only 40-
50% during a conversation does the speaker
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
conserved 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 very important.
 If the voice signal is considered as noise, the
transmitter is turned 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 receiver.
 This can be very annoying to the receiving
user since it appears as a dead connection.
 In order to overcome this problem, the
receiver creates a minimum of background
noise called comfort noise.
 Comfort noise eliminates the impression 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
maintain 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 of 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 sub-
channels and needs to listen only to its own
sub-channel.
 In the time between successive paging sub-
channels, 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 station.
 GSM uses TDMA technology in the radio interface to share a
single frequency between several users, assigning sequential
timeslots to the individual users sharing a frequency.
 Each user transmits periodically one-eighth of the time within
one of the eight timeslots.
 Since the users are various distances from the base 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 must be adjusted accordingly.
 Timing Advance (TA) is the variable controlling this adjustment.
 This synchronization between BTS and MS is
achieved by using the concept of Timing 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 value received from the BTS, the MS
know when to send the frame, so that it can arrive at
the BTS in synchronism.
 The values of the TA is continuously calculated and
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
second or bit/s.
 In GSM the net bit rate over the air interface
is 270kbit/s.

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