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 adds redundancy bits to
the original information to detect and
correct, errors occurred during transmission.
GSM uses convolution coding and
interleaving 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.
Class Ia - 50 bits (most sensitive to bit errors)
Class Ib - 132 bits (moderately sensitive to bit
errors)
Class 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 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 rearranges 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 whole bursts during the
transmission, by dispersing the errors.
Since the errors become less concentrated,
it is then easier to correct them.
 Encryption
It is used to protect signaling and data. This
process is done using A3, A5 and A8
algorithms

Modulation
The modulation chosen for the GSM
system is the Gaussian Minimum
Shift Keying (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 silence 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.
 Voice Activity Detection (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 of producing then, an
unpleasant effect called clipping.
Comfort noise
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.
Power control
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.
Discontinuous
reception
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, timing
advance 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.