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Institutionen fr systemteknikDepartment of Electrical Engineering

Master thesis

Design and implementation of a test tool for the GSM traffic channel

Theo jerteg

ISY-LITH-EX-3169-20022002-06-04

TEKNISKA HGSKOLANLINKPINGS UNIVERSITET

Department of Electrical engineering Linkpings tekniska hgskolaLinkping University Linkpings UniversitetS-581 83 Linkping 581 83 Linkping


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Design and implementation of a test tool for the GSM traffic channel

Examensarbete utfrt i datatransmissionvid Linkpings Tekniska Hgskola av

Theo jerteg

Reg nr:LiTH-ISY-EX-3169-2002

Handledare: Michael Lundkvist, Danjel McGougan, Enea EpactExaminator: Ulf Henriksson, ISY

Linkping 2002-06-04


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Avdelning, Institution

Division, Department

Institutionen fr Systemteknik581 83 LINKPING

Datum

Date2002-06-04

SprkLanguage RapporttypReport category ISBN

Svenska/SwedishX Engelska/English

LicentiatavhandlingX Examensarbete

ISRN LITH-ISY-EX-3169-2002

C-uppsats

D-uppsats

Serietitel och serienummer

Title of series, numberingISSN

vrig rapport_____

URL fr elektronisk version

http://www.ep.liu.se/exjobb/isy/2002/3169/

Titel

Title

Design och implementation av ett testverktyg fr GSM talkanal.

Design and implementation of a test tool for the GSM traffic channel.

Frfattare

AuthorTheo jerteg

SammanfattningAbstractTodays systems for telecommunication are getting more and more complex. Automatic testing isrequired to guarantee quality of the systems produced. An actual example is the introduction ofGPRS traffic in the GSM network nodes. This thesis investigates the need and demands for such anautomatic testing of the traffic channels in the GSM system. A solution intended to be a part of theEricsson TSS is proposed. One problem to be solved is that todays tools for testing do not supporttesting of speech channels with the speech transcoder unit installed. As part of the investigation, aspeech codec is implemented for execution on current hardware used in the test platform. The

selected speech codec is the enhanced full rate codec, generating a bitstream of 12.2 kbit/s, andgives a good trade-off between compression and speech quality. The report covers the design of the

test tool and the implementation of speech codec. Particularly performance problems in the imple-mentation of the encoder will be addressed.

Nyckelord

KeywordGSM, speech codec, traffic generation, test tool, test automation, DSP, telecommunication, TSS,encoder, decoder


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Design and implementation of a test tool for the GSM traffic channel vii

Contents

CHAPTER 1 Preface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1

CHAPTER 2 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3

2.1 Background . . . . . . . . . . . . . . . . . . . . . . . 3

2.2 Assignment . . . . . . . . . . . . . . . . . . . . . . . 4

2.3 Limitations . . . . . . . . . . . . . . . . . . . . . . . . 5

2.4 Purpose of this report . . . . . . . . . . . . . . . 5

2.5 Reading instructions . . . . . . . . . . . . . . . . 5

CHAPTER 3 GSM basics. . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

3.1 Telecommunication overview. . . . . . . . . 7

3.2 Networking elements. . . . . . . . . . . . . . . . 7

3.3 Interfaces . . . . . . . . . . . . . . . . . . . . . . . . . 9

3.4 Inside the equipment. . . . . . . . . . . . . . . 10

CHAPTER 4 Speech coding . . . . . . . . . . . . . . . . . . . . . . . . 13

4.1 General . . . . . . . . . . . . . . . . . . . . . . . . . . 13

4.2 Human speech model . . . . . . . . . . . . . . 13

4.3 Different codecs. . . . . . . . . . . . . . . . . . . 15

4.4 Enhanced full rate codec . . . . . . . . . . . 16

4.5 Speech encoding. . . . . . . . . . . . . . . . . . 19

4.6 Speech decoding . . . . . . . . . . . . . . . . . . 22


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viii Design and implementation of a test tool for the GSM traffic channel

4.7 Discontinuous Transmission Mode . . . 23

4.8 Voice Activity Detection . . . . . . . . . . . . 24

4.9 Substitution and muting . . . . . . . . . . . . 26

4.10Comfort Noise . . . . . . . . . . . . . . . . . . . . 264.11Homing . . . . . . . . . . . . . . . . . . . . . . . . . . 27

4.12Transmission parameters . . . . . . . . . . . 27

CHAPTER 5 Analysis. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

5.1 TSS overview . . . . . . . . . . . . . . . . . . . . . 295.1.1 Overview . . . . . . . . . . . . . . . . . . . . 29

5.1.2 TSS architecture . . . . . . . . . . . . . . 29

5.2 Problems. . . . . . . . . . . . . . . . . . . . . . . . . 30

5.3 Use cases . . . . . . . . . . . . . . . . . . . . . . . . 315.3.1 Use case 1. . . . . . . . . . . . . . . . . . . 32

5.3.2 Use case 2. . . . . . . . . . . . . . . . . . . 325.3.3 Use case 3. . . . . . . . . . . . . . . . . . . 33

5.3.4 Use case 4. . . . . . . . . . . . . . . . . . . 33

5.3.5 Use case 5. . . . . . . . . . . . . . . . . . . 34

5.3.6 Use case 6. . . . . . . . . . . . . . . . . . . 34

5.3.7 Use case 7. . . . . . . . . . . . . . . . . . . 355.3.8 Use case 8. . . . . . . . . . . . . . . . . . . 35

5.4 Requirements. . . . . . . . . . . . . . . . . . . . . 36

CHAPTER 6 Design . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

6.1 Proposed solution . . . . . . . . . . . . . . . . . 39

6.2 Functional overview . . . . . . . . . . . . . . . 406.2.1 PCM Frame adaption. . . . . . . . . . . 40

6.2.2 TRAU/TRAB Frame adaption. . . . . 416.2.3 Speech codecs . . . . . . . . . . . . . . . 41

6.2.4 PCM conversion. . . . . . . . . . . . . . . 426.2.5 Traffic generator . . . . . . . . . . . . . . . 42

6.2.6 Statistics . . . . . . . . . . . . . . . . . . . . 43

6.2.7 User interface . . . . . . . . . . . . . . . . 44

6.3 Communication interfaces . . . . . . . . . . 466.3.1 A interface . . . . . . . . . . . . . . . . . . . 46

6.3.2 Abis interface . . . . . . . . . . . . . . . . . 46


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Design and implementation of a test tool for the GSM traffic channel ix

6.3.3 TSS Interface . . . . . . . . . . . . . . . . .47

6.3.4 Signal interface. . . . . . . . . . . . . . . .47

6.4 Software implementation. . . . . . . . . . . . 48

6.4.1 External traffic generator . . . . . . . . 496.4.2 Driver modules . . . . . . . . . . . . . . . . 49

6.4.3 Proxy module . . . . . . . . . . . . . . . . .49

6.4.4 Traffic generator links . . . . . . . . . . .50

6.4.5 Traffic generators . . . . . . . . . . . . . . 51

6.4.6 Traffic generator manager . . . . . . . 51

6.4.7 PCI Interface. . . . . . . . . . . . . . . . . .52

6.4.8 ESSI Interface . . . . . . . . . . . . . . . . 52

6.4.9 Codecs and frame adapters . . . . . .52

CHAPTER 7 Implementation and Test . . . . . . . . . . . . . . . . . 53

7.1 Implementation of test tool . . . . . . . . . . 53

7.2 Implementation of speech codec . . . . .547.2.1 Multichannel mode . . . . . . . . . . . . .54

7.2.2 DSP implementation. . . . . . . . . . . .54

7.3 Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55

7.4 Problems . . . . . . . . . . . . . . . . . . . . . . . . . 557.4.1 Performance of generated code . . . 557.4.2 Size of generated code. . . . . . . . . .57

7.4.3 Word length . . . . . . . . . . . . . . . . . . 58

7.4.4 Performance of individual routines . 58

7.5 Testing . . . . . . . . . . . . . . . . . . . . . . . . . . .607.5.1 Multichannel mode . . . . . . . . . . . . .60

7.5.2 DSP implementation. . . . . . . . . . . .60

7.6 Improvements . . . . . . . . . . . . . . . . . . . . . 61

CHAPTER 8 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . .63

8.1 Need of system . . . . . . . . . . . . . . . . . . . .63

8.2 Implementation so far . . . . . . . . . . . . . . 63

8.3 Alternative solutions . . . . . . . . . . . . . . . 64


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x Design and implementation of a test tool for the GSM traffic channel

APPENDIX A Generated code examples . . . . . . . . . . . . . . . 65

APPENDIX B Original C-code . . . . . . . . . . . . . . . . . . . . . . . . 69

References . . . . . . . . . . . . . . . . . . . . . . . . . . . 71

Glossary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75

Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79


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Design and implementation of a test tool for the GSM traffic channel 1

CHAPTER 1 Preface

Thanks to all people at Enea Epact and Ericsson who have

been answering my questions and helping with whatever

problem Ive had.

Also a lot of thanks to my wife who has been very support-

ive when I really wanted to throw the computer out of the

window.

Linkping 2002-06-04

Theo jerteg


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2 Design and implementation of a test tool for the GSM traffic channel


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CHAPTER 2 Introduction

2.1 Background

This master thesis work was done as part of the programme

for Computer Science and Engineering at Linkping Uni-versity.

The employer was Enea Epact, but the real customer was

Ericsson Radio Network Center.

With the introduction of GPRS in the GSM network, the

need for verification of the traffic channels increase. For

example, a useful scenario for this new tool would be when

there are a lot of GPRS traffic going on through the BSC,

then a voice call is being set up and GPRS traffic channels

has to be pre-empted. It would also be interesting to use the

system for generating background load on the traffic chan-

nels, while performing other automatic tests.

The thesis consists of two parts. The first part consists of an

analysis to find out the needs and to see how it is possible to

implement the system. The second part involves implement-

ing part of the system.


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Assignment


2.2 Assignment

The customer has a product called Test and Simulation

Solutions (TSS), used for testing nodes in the Global Sys-tem for Mobile Communications (GSM) and General Packet

Radio Systems (GPRS). The TSS consists of special hard-

ware as well as software and is used to simulate traffic in

mobile cell phone networks.

Within TSS there is a problem when it comes to circuit

switched networks like the GSM. The test tool is only able

to set up the connections using the signalling channels, the

physical channels used for traffic are not verified with thistool. When it comes to packet switched networks, like the

GPRS, this problem has been handled.

The first assignment was to see how the product TSS could

be enhanced with a new tool allowing for verification of the

GSM traffic channels. This includes how to insert traffic

(i.e. simulated speech) into the physical channel, how to

generate it, and how to control the tool in a way that is con-

sistent with the rest of the product. To do this, several sub-

areas have to be studied. The GSM network itself has to be

studied in order to find out scenarios where this new tool

can become useful. What standards are there that the new

tool must comply to in order to connect to the GSM net-

work? And what about the user interface part, what are the

common standard the tool must comply to in order to be eas-

ily integrated with the TSS?

The second assignment for this master thesis was to imple-

ment part of the system proposed. In this case the choosen

part is the enhanced full rate speech codec, which is the

most commonly used codec of cellular phone systems. This

part will involve examining the speech compression and

decompression algorithms to be able to implement them on

the target hardware.


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Limitations


2.3 Limitations

Only the speech encoding and decoding functions will be

implemented since this is an area where there is not enoughknowledge at the employer or the customer. This is basically

one function which can be developed detached from the rest

of the test system and easily integrated later.

The full test system, with traffic generator, user interface

etc., will not be implemented within this master thesis due to

limitations in time. Also this area is more well known and

the both the employer and the customer has a lot of experi-

ence in this area.

2.4 Purpose of this report

The purpose of this report is to summarize the work per-

formed and to document the results achieved. The situation

before is described as well as the gains from implementing

and using the proposed test system.

2.5 Reading instructions

The report is intended to be readable by most senior students

at a Master of Science programme and teachers at those pro-

grammes. However it will also include details about the sys-

tem useful for professionals working in the area of

telecommunication using or further developing the system.

Professionals implementing the system should consider

reading [1] and [2].

Chapter 2 gives a brief introduction to the report and defines

the assignment.

Chapter 3 deals with the theory behind the GSM network

and is useful for those not working within the telecommuni-

cation area.


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Reading instructions


Chapter 4 gives an introduction to the theory behind speech

coding and the parts of a full featured speech coder.

Chapter 5 gives a deeper analysis of the need and usage of a

test tool.

Chapter 6 describes the design to be used for the proposed

system.

Chapter 7 discusses issues concerning test and implementa-

tion. Problems, improvements and errors are discussed.

Chapter 8 summarizes the thesis and discusses drawn con-clusions.


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CHAPTER 3 GSM basics

3.1 Telecommunication overview

The following sections will introduce terminology from the

area of mobile telecommunication. The intention is to give

an overview and to define the concepts of mobile telecom-

munication and its notion. Details can be found in [3] and

[4].

3.2 Networking elements

The GSM network consists of a hierarchy of elements as can

be seen in figure 3.1. In this particular network there are two

Mobile Switching Centres (MSC), one Gateway MSC

(GMSC) and three different access networks. There are also

some databases to store the information needed in the net-

work. This public land mobile network (PLMN) is also con-nected to the public switched telephone network (PSTN)

and one other PLMN.

The mobile station (MS) can be any type of end user equip-

ment like a cellular phone, laptop or fax machine with radio

modem.


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Networking elements


FIGURE 3.1. GSM network hierarchy.

The Base Transceiver Station (BTS) contains equipment for

the physical radio transmitting and receiving, signal meas-

urements, encryption and communication with the base sta-

tion controller (BSC).

Base station controllers serve as concentrator nodes for sev-

eral BTS. It is responsible for allocating radio channels used

by the BTS and to connect speech and signalling channels tothe MSC. It also handles transfers, handovers, of mobile sta-

tions between different BTS. This occurs for example, when

the mobile station is being moved within a limited area, for

example in a city.

The mobile switching centre corresponds to the local station

in a public switched telephone network, though it does not

have any hard wired subscribers. It is responsible for mov-

G M S CP S TN

Other

PLM N

A U C EIR

H L R

V LR V L R

B S C

M S C M SC

B T S

B T S

B T S M S

A

Abis

U m


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Interfaces


ing mobile stations between different BTS and between dif-

ferent MSC. Transfer of mobile stations between BTS

occurs for example when the end user is travelling longer

distances.

The GMSC is similar to the MSC. It serves as an interface

between the network and other networks, for example the

PSTN or other PLMN. It is responsible for charging

between different network operators. There may be several

GMSC in a network, and they may be integrated with the

MSC.

There are also databases connected to the switching networkelements. The home location register (HLR) contains infor-

mation about the end user and under which MSC it currently

is registered. At each MSC there is a visitor location register

(VLR) which contains information about all mobile stations

that are currently in that MSC service area.

Authentication centre (AUC) contains information used for

encryption and authorization.

Equipment identity register (EIR) stores information about

the MS identity. It is used to control that a mobile station is

not blocked.

3.3 Interfaces

The different interfaces between the network elements are

named in figure 3.1. Starting from the MS side, first comesthe Um interface. This is the radio interface between a

mobile station and the BTS [5]. Each logical speech channel

requires 16 kbit/s data rate. With error correction codes,

burst management and encryption the used data rate is 33.8

kbit/s per channel.

The interface between the BTS and the BSC is named Abis.

It uses standard T1 or E1 cables for interconnection. Four 16


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Inside the equipment


kbit/s speech channels multiplexed into one 64 kbit/s chan-

nel corresponds to one timeslot in the E1 or T1 link between

the BTS and BSC. The job to transform the speech informa-

tion to the correct format is done by the transcoding and rate

adaption unit (TRAU). This is where the speech encoding

and decoding functionality are situated. The Abis interface

is defined in [6].

The A interface between BSC and MSC consists of 64 kbit/s

PCM links on a E1 or T1 interconnection. The speech is

here coded with a-law or -law PCM depending on if a

European or American/Pacific system is used. More infor-

mation about the A interface is found in [7].

3.4 Inside the equipment

The basic structure inside the switching network elements is

shown in figure 3.2. The main components in the BSC, and

MSC are a number of line interfaces (LIF), i.e. the physical

connections to the other elements, a switching unit and a

control system. There is also some kind of terminal equip-ment (TERM) for operation and maintenance.

Inside the BSC there is also a pool of cpu resources for

speech transcoding and rate adaption in the TRAU module,

available for use when connecting traffic channels between

the MSC and BSC.

The BTS consists of a number of transceivers (TRX), which

are multiplexed into the radio interface. The transceivers areaccessed from the BSC via a transceiver radio interface

switching traffic channels from the BSC to the correct trans-

ceiver and vice versa.


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FIGURE 3.2. Inside the switching elements.

T R X

T R I

C O N T R O L

T R X

L IF

L IF

T E R M

T R A U

C O N T R O L

M

UX

L IF

L IF

T E R M

C O N T R O L

B SC

GM SC/M SC

BT S

Other networks


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CHAPTER 4 Speech coding

4.1 General

A general communication system can be split into a couple

of modules, as in figure 4.1. The transmitter is modelled by

a source encoder, channel encoder and modulator. The

receiver can be modelled by a demodulator, channel decoder

and source decoder as seen in . Between the source and des-

tination side, there is the transmission media, called a chan-

nel. Since most channels not are perfect, the channels will

add some noise to the signal.

The following sections will concentrate on the features of

the source encoder and decoder.

4.2 Human speech model

Since we are interested in compression of human speech , it

is interesting to study how speech is produced.

Sounds are produced in two different ways. There are

voiced sounds, all vowels for example, that are produced by

the vibrating vocal cords at the glottis. There are also

unvoiced sounds, like p, t, and f, which are the result of


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Human speech model


FIGURE 4.1. General communication system.

air flow through a construction in the vocal tract. Note that

consonants can also be voiced sounds like b, d and j.Also some sounds are the result of a combination of these.

The excitation signal for these sounds is the vibration of the

vocal cords. The frequency of the vibration does not vary

greatly in time due to the physiology of the glottis, and can

therefore be modelled using a slowly varying model.

The vocal tract is a mechanical system, which means that

the vocal gestures are relatively slow. It can be modelled by

a parametric filter, where the parameters will vary with time.

The bandwidth for these filter parameters are lower than for

the speech signal itself. This is called the short time station-

arity of the vocal tract filter.

The human ear is also to be considered in the models. It is

able to perceive signals between 20 Hz - 20 kHz. However,it is most responsive to frequencies between 200 and 5600

Hz which usually carries the most important information

when it comes to human speech. Most telephone systems

has a bandwidth limitation to the range of 300 to 3400 Hz,

which will be possible without losing to much of the quality.

Another important aspect is masking, which means that one

sound can be obscured by the presence of another. Masking

Informationsource

Sourceencoder

Channelencoder

Modulator

Channel

DemodulatorChanneldecoder

Sourcedecoder

Informationdestination

Noise


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Different codecs


can be done either as temporal masking or as frequency or

spectral masking. Temporal masking means that signals

with energy in frequencies close to each other can mask

each other, if the time delay between the signals are short.

Frequency masking means that a lower frequency can be

masked by a simultaneous higher frequency. This can be

used to distribute quantization noise to the masked fre-

quency.

4.3 Different codecs

There are different types of speech codecs. The two maintypes of coders are waveform coders and vocoders (voice

coders). There are also hybrid coders which combine the

properties of those coders. Some properties of these codecs

are shown in figure 4.2.

FIGURE 4.2. Comparison of compression ratio and

speech quality of different codec categories.

Waveform coding is the process of describing the signalsamplitude curve with a number of discrete values. This is

basically done in three steps - sampling, quantization and

coding. The quality of the signal will in general be very

good but requires more bandwidth. Waveform coders tend

to be more robust against unexpected input, like music, than

vocoders.

Lo w

Excellent

Good

Average

Hybrid codersWaveform coders

Vocoders

Speech qual ity

bitrate(kbit/s)2 4 8 16 32 64


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Enhanced full rate codec


Vocoders use a totally different approach. They try to figure

out which parameters were used to create the signal, given

that it passed through a known filter model. The filter model

in mind is of course a model of the human speech organs.

Vocoders lose a bit of quality of the transferred signal, but

will on the other hand use less bandwidth. The process of

transmitting speech parameters instead of the speech can be

compared to transmitting notes instead of the music itself.

Because of the two contradictionary demands of low band-

width usage and high signal quality in cellular phone sys-

tems, hybrid coders are used. The coders use a combination

of parameterizing and waveform coding to achieve this. Thebitrate can be reduced to levels under the 64 kbit/s PCM

coding with good signal quality.

There are several possible speech coders for usage in cellu-

lar phone systems (See also [8]):

Full rate 13 kbit/s

Enhanced full rate 12.2 kbit/s

Half rate 5.60 kbit/s Adaptive multi rate 4.75-12.2 kbit/s

Next section will give a further study of the enhanced full

rate (EFR) coder.

4.4 Enhanced full rate codec

The enhanced full rate codec is an Algebraic Code ExcitedLinear Prediction (ACELP) system. A simple description of

the synthesis model can be found in figure 4.3.


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FIGURE 4.3. CELP synthesis model

The excitation signal is constructed by adding the excitation

vectors from an adaptive and a fixed codebook. The result isthen filtered through the short-term synthesis filter to recon-

struct the speech. To get the code book vectors, analysis-by-

synthesis is used, in a procedure where the error between

original speech and reconstructed speech is minimized.

The error minimization is done by filtering the error signal

through a weighted perception filter, which masks the error

by weighting it less in regions near the vocal tract reso-

nances and more in regions away from them.

The codec encodes/decodes 20 ms frames of speech consist-

ing of 260 samples at 8000 Hz. The samples are encoded

with 13 bit uniform PCM. Encoded speech are delivered in

50 frames/s with 244 bits in each frame.

Other features than compression are often used in conjunc-

tion with the speech codecs [9]. Techniques for comfortnoise generation, voice activity detection (VAD) and discon-

tinuous transmission (DTX) are integrated into the same

unit, figure 4.4. The different parameters sent between the

blocks of the codec are explained below.

1. Uncompressed speech samples.

2. The voice activity detector checks if the input signal con-

tains speech.

3. Encoded speech parameters.

Adapt ive codebook

Fixedcodebook

L P s yn th es is P os t f il te rin g+


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4. Background noise is evaluated and a silence descriptor

(SID) frame is calculated. This frame is passed between

the comfort noise module and the DTX handler.

5. Speech flag, indicates whether information bits arespeech or silence descriptor information.

6. Speech information bits transferred over radio system.

7. A flag is set if a corrupt frame has been received.

8. The DTX handler gets to know whether the delivered

frame is a SID frame or not.

9. The substitution module is notified if a frame is corrupt

or lost.

Also there are parts not shown in the figure 4.4, responsible

for AD/DA conversion on the MS side and conversion

between a-law/-law PCM used in transmission media on

the base station side and linear PCM used as input and out-

put to the speech codec.

FIGURE 4.4. Parts of the EFR speech codec.

For a more detailed description of the mathematics behind

the enhanced full rate codec the interested could have a look

in [10]. Also check[11] for voice activity detection, [12] for

discontinuous transmission and [13] for comfort noise gen-

eration.

V oice

activity

detector

Speechencoder

Com fo r tnoice TXfunct ions

D TXControl

an doperation

Transm it side R ece ive side

D T XControl

andoperat ion

Speechframe

substitution

Speechdecoder

Com fo r tnoise RXfunct ions

1

2

3

4

3

4

15

6

6

7

8

9

3

3

1

1


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Speech encoding


4.5 Speech encoding

The encoding process is described visually in figure 4.5.

The first step is to filter some unwanted low frequency com-ponents from the input signal.

The Linear Prediction (LP) coefficients are calculated twice

per frame. The reason is to remove redundancy from the

speech signal. The sampled speech signal is not stationary

and therefore the predictor coefficients must be adapted to

the changing statistics of the signal. This is done by consid-

ering the signal stationary for a short time interval, called a

frame of the speech signal. When using the enhanced fullrate codec the frame length is 20 ms, but linear prediction

calculations are done twice per frame, i.e. on a 10 ms basis.

By minimizing the mean squared error of the error signal, an

expression for the linear prediction coefficients in the form

ofp linear equations withp unknowns can be found. This

system is then solved by using the Levinson-Durbin algo-

rithm.

The LP coefficients are then to be quantizised before trans-

mission. To further reduce the transmitted number of bits,

the parameters are interpolated between the subframes.

Quantization and interpolation of linear prediction coeffi-

cients is hard to do, because even small changes in the

parameters may cause big changes in the power spectrum

and possibly in an unstable synthesis filter. Therefore the

parameters are transformed into Line Spectral Pairs (LSP)

or Line Spectral Frequencies (LSF).

LSF are calculated only for the second and fourth 5 ms sub

frame of a full 20 ms frame of speech, and interpolation is

used to get the LSF parameters for the first and third sub-

frame.


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Speech encoding


To simplify the adaptive code book search, a sub-optimal

lag is calculated directly from the weighted speech input.

This will confine the closed-loop pitch search to a small

number of values around the estimated lag value.

The following steps are then performed for every sub frame:

First the target signal for the adaptive codebook search iscalculated. This is done by subtracting the zero input

response of the weighted synthesis filter from the

weighted speech signal. The zero input response is the

output of the filter due to past input, i.e. with a current

input of zero.

The impulse response of the weighted synthesis filter iscalculated.

The target signal and the impulse response of theweighted synthesis filter are then used to search around

the open-loop pitch lag for the optimal pitch lag and gain.

The adaptive codebook contribution to the target signal iscalculated and removed from the target signal giving the

target signal for the innovative codebook search.

Finally filter memories are updated for use in the nextsubframe.


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Speech encoding


FIGURE 4.5. Encoder.

frame

subframe

Pre-processing

LPC

analysis

(twiceperframe)

Open-looppi

tchsearch

(twiceper

frame)

Adaptivecodebo

ok

search

Innovativecodebook

search

Filtermemory

upda

te

Pre-processing

windowi

ng

and

autocorrela

tion

R[]

Levinson-

Durbin

R[]->

A(

z)

A(z)->L

SP

LSP

quantization

LSP

indices

interpolation

forthe4

subframes

LSP->

(z)

interpolation

forthe4

subframes

LSP->

A(z)

s(n) A

(z)

compute

weighted

speec

(4subfram

es)

find

open-loopp

itch

A(z)

(z)

h(n)

To

x(n)

x(n)

A(z)

(z)

h(n)

computetarget

foradaptive

codebook

findbestdelay

andgain

quantize

LTP

_gain

compute

adaptive

codebook

contribution

computetarget

forinovation

code

index

LTP

gain

index

update

filter

memoriesfor

nextsub

frame

fixe

d

codebook

gain

quantiz

ation

fixedcod

ebook

gainin

dex

pitch

index

x2

(n)

compute

impulse

response

findbest

innovation

comp

ute

exitat

ion


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Speech decoding


4.6 Speech decoding

The decoder part, figure 4.6, is simpler than the encoding

part.

For every frame the transmitted LSP indices are used to

decode the LSP coefficients. These coefficients are interpo-

lated to get coefficients for all four subframes. Finally the

LSP coefficients are transformed back to the LP filter coeffi-

cient domain to be used in the synthesizing filter recon-

structing the speech.

The next five steps are performed for every subframe:

The adaptive codebook vector is decoded.

The adaptive codebook gain is decoded by using thereceived index to find the gain from the quantization

table.

The innovative codebook vector is decoded by using thereceived index to extract positions and amplitudes of the

excitation pulses and find the algebraic code vector.

The fixed codebook gain is decoded from the receivedindex.

Finally the speech is reconstructed by filtering of theexcitation vectors through the synthesis filter.

The reconstructed speech signal has to be filtered through an

adaptive postfilter and up-scaled before it is totally decoded.


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Discontinuous Transmission Mode


FIGURE 4.6. Decoder.

4.7 Discontinuous Transmission Mode

To minimize interference in the air interface and to save

power in the mobile station the radio transmitter in the

mobile station can be turned off during pauses in speech.

The technique is called discontinuous transmission mode

[12]. During a normal phone conversation, the participants

alternate so that, on the average, each direction of transmis-

sion is occupied about 50% of the time.

For the DTX handler the following functions are needed: Voice activity detector on transmit side

Evaluation of background noise on transmit side.

Comfort noise generation on receive side.

The control of the DTX handler is indirectly done from the

voice activity detector decision, see figure 4.4. The result

from the voice activity detector is used to decide whether

fram e sub fram e

LS Pindices

decode LSP

interpolationof LSP for the

4 subframes

LS P -> (z )

pitchindex

gainsindices

codeindex

decodeadapt ive

codebook

decodeinnovative

codebook

decodegains

const ructexcitation

post filter

post -processing

synthesisfilter

s n( ) s' n( )


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Voice Activity Detection


the frame is to be passed to the transmitting radio subsystem

or not. This has the effect that when the speaker stops talk-

ing, the transmission is cut off. During the speech pause,

transmission is resumed at regular intervals to send an

updated silence descriptor frame, in order to make the lis-

tener perceive the presence of background noise.

On the receiver side, the DTX handler receives frames from

the radio subsystem:

Good speech frames are passed directly to the speechdecoder.

Valid SID frames are used for comfort noise generation,see 4.10.

Bad or lost frames are replaced by a substitution andmuting procedure described in 4.9.

4.8 Voice Activity Detection

Voice activity detection [11] is used together with the dis-

continuous transmission mode. The purpose is to provide aflag indicating whether a frame contains speech or not. The

algorithm does this by comparing the energy of the input

signal to a threshold value.

Mobile environments are often subject to a lot of back-

ground noise, making the noise/speech ratio low. The

speech signal is filtered to improve the noise/speech ratio,

and the energy is calculated. For each frame the threshold

value is adapted, taking into concern:

Presence of information tones. DTMF tones should notbe filtered out, while strong resonances from vehicles for

example should be classified as noise.

Periodic component in input signal. If the input signalcontains stationary noise the threshold should be raised,

however vowels also produce a stationary frequency


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Voice Activity Detection


spectrum. This is detected by comparing adjacent long

term predictor values.

Stationary input signal. Spectral characteristics of the

input signal is obtained from several frames to detect sta-tionary frequencies that can be removed in the adaptive

filtering of the input signal.

The energy of the input signal is then compared to the

threshold to get a boolean VAD decision. The final decision

includes a hangover period to make sure enough frames are

sent to be able to calculate useable SID frames.

The outline of the algorithm is described in figure 4.7. Ascan be seen the algorithm uses values from the encoder part

as its input to minimize the computation needs.

FIGURE 4.7. Schematic overview of voice activitydetection algorithm.

VAD

decisionVAD

hangover

Periodicity

detection

Tonedetection

Predictorvalues

computation

Autocorrelation

averaging

Spectralcomparison

Threshold

adaption

Adapt. filtering

and energycomputation

Autocorrelation

Lag values

Reflection

VAD flag

coefficients

vector


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Substitution and muting


4.9 Substitution and muting

Substitution and muting is performed as a result of lost or

invalid frames received by the radio subsystem [14]. Mutingis performed to avoid generating annoying sounds as a result

from the frame substitution.

Normal decoding of lost frames will generally result in very

unpleasant noise effects. To improve sound quality, lost

frames are replaced by a repetition or extrapolation of previ-

ous good speech frames. For each frame being substituted,

the output level is decreased resulting in silence if too many

frames are lost.

For lost SID frames, the first lost frame is substituted by the

last valid SID frame. Subsequent lost SID frames are also

replaced but the signal level is decreased, until silence is

achieved.

4.10 Comfort Noise

The speech normally transmitted will always contain some

acoustic background noise. When the DTX cuts the radio

transmission, a total silence would be inserted. This will be

very annoying to the user, and speech may even be hardly

intelligible.

To avoid this, a synthetic noise similar to the background

noise on the transmit side, is generated on the receive side.

Comfort noise parameters are estimated on the transmit side

and sent in special frames called silence descriptor frames

[13]. The SID frame is transmitted at the end of speech

bursts and serves as an end of speech marker for the receive

side. In order to update the comfort noise characteristics at

the receive side, SID frames are transmitted at low, but regu-

lar intervals also during speech pauses. This also serves the

purpose of improving the measurement of the radio link

quality by the radio subsystem.


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Homing


4.11 Homing

Homing is the process of resetting the codec to a pre-defined

state. To allow reset of remote codecs, special homingframes have been defined for both the coder and the

encoder. The homing frames are needed to ensure that the

local and the remote speech codecs are synchronized in

order to transcode speech information correct.

When the codec receives a homing frame at its input, it is

processed as normal, generating an output frame which con-

tents is usually unknown. After successful completion of

frame processing, the reset functions are invoked for allmodules - codec, VAD, DTX and comfort nose generator,

setting the internal variables to pre-defined values. When

the next frame arrives at the input, the codec will start from

its home state.

Values for internal variables are defined in [10].

4.12 Transmission parameters

The parameters transmitted from the encoder are the quan-

tizised linear spectral pair parameters, adaptive codebook

gain and pitch, fixed codebook gain and pitch.

From the encoder, a total of 244 bits/20 ms are delivered for

each frame, corresponding to a bitrate of 12.2 kbit/s. These

are packed by the transcoding and rate adaption unit into

frames of 320 bits resulting in a bitrate of 16 kbit/s. The

additional bits are for CRC, synchronization, frame tagging

and speech flag indicating if the frame contains speech or

SID data. The layout and contents of the TRAU frames are

fully described in [16].

When no speech is detected and SID frames are transmitted,

only parts of the frame need to be used for data transfer. The

remaining bits are then set to a fixed bit pattern called the


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Transmission parameters


SID code word. The parameters replaced by the SID code-

word are the adaptive codebook gain and pitch and the fixed

codebook gain and pitch.


39/94


CHAPTER 5 Analysis

5.1 TSS overview

5.1.1 Overview

With increased complexity and number of functions in cel-

lular phone systems there is a need for more advanced sys-

tems for test and verification. To meet these demands, the

TSS has been developed by Ericsson. It consists of hardwareand software simulating most parts in a PLMN to be able to

put each part in the PLMN under as realistic tests as possi-

ble.

TSS can be connected to different interfaces in the tele-

phone network depending on what aspect is going to be

tested. For example, it can be used to test the BSC, by con-

necting to both sides of the BSC, simulating several BTS on

one side of the BSC and one MSC on the other side of the

BTS.

5.1.2 TSS architecture

The TSS architecture is based on two different platforms - a

unix workstation and standard VME hardware, see

figure 5.1.


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Problems


The workstations provides TSS with a user friendly devel-

opment and an execution environment with a common GUI

and a test programming environment with a high level test

program language. There are also compilers, editors and log

tools available.

Test programs running on the workstation will then perform

the tests, by controlling the applications and protocols run-

ning on VME hardware.

FIGURE 5.1. TSS architecture overview

5.2 Problems

The TSS is only able to simulate the signalling part in the

network. No checks are done that the physical connections

for the calls simulated really are set up. Therefore the traffic

channels remain untested. This is a problem for circuit

switched networks like the GSM.

There exists similar systems today, however they suffer

from several problems:

Different systems for the GSM and Personal Digital cel-lular (PDC) mobile systems.

They are more of 'hacks' than officially supported prod-ucts.

Ethernet E1/T1

VME

VME

Workstation

BSC


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Use cases


Lack of documentation.

Troublesome installation and setup.

Non standardized user interface.

Not designed for testing with the transcoding and rateadaption unit installed. The unit is simply removed and a

loop back is set up.

5.3 Use cases

As an important step, to understand the requirements for this

tool, some questions have to be answered. How will it be

connected to the system under test and what must it be ableto do? To get the answer to this question, we have to look at

the different use cases for the tool. A use case, or scenario, is

a description of how the tool is going to be used, from the

users point of view.

The use cases are based on the different ways of how a

phone call can be initiated. It can be a call from a mobile

station to a subscriber in the public switched telephone net-

work, or a call from a mobile station to another mobile sta-

tion. Also taken into consideration is the possibilities to use

TSS to simulate different parts of the GSM network. This

gives the opportunity to perform tests with either real hard-

ware or with the TSS simulating selected parts.

Eight different use cases have been identified in table 5.1. It

is also listed in the table if TSS is used to simulate parts of

the network, and which interface the test tool is connectedto.

The term point of interconnection (POI) is here used to point

out the interface where the GSM network connects to other

networks through a GMSC.


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Use cases


TABLE 5.1. Identified use cases.

5.3.1 Use case 1

In this case, figure 5.2, we have a simulated BTS and a real

MSC. Basic testing of the BSC with the TRA unit can be

done by connecting the test tool only to the Abis interface

and looping back the call, to the Abis interface in the MSC.

This would simulate a MS to MS call. The basic use case

would allow the test to generate idle pattern on the traffic

channels to simulate idle channels without traffic.

5.3.2 Use case 2

This case is identical to use case 1 except that the test tool

now also generate traffic frames and inserts them on the

Abis interface. The call is set up using normal signallingfrom TSS. When receiving information about the physical

channel allocated, the TSS test program tells the test tool to

use this channel for transmitting generated traffic.

No. Use case MSC BTS Interface

1 idle pattern real simulated Abis2 MS-MS real simulated Abis

3 MS-PSTN real simulated Abis+POI

4 MS - MS simulated real A

5 MS - PSTN simulated real A

6 idle pattern simulated simulated Abis

7 MS - MS simulated simulated Abis + A

8 MS - PSTN simulated simulated Abis + A


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Use cases


FIGURE 5.2. Use case 1 and 2 with simulated BTS andreal MSC. Case 1 with no traffic and case 2 with trafficframes.

5.3.3 Use case 3

In this case, figure 5.3, the BTS is simulated and there is a

real MSC. The test tool is connected to the Abis interface

and the POI interface. This simulates an MS to PSTN call.

The call is set up using normal signalling from TSS. When

receiving information about the physical channel allocated,

the TSS test program tells the test tool to use this channel for

transmitting generated traffic. The test tool connected to the

MSCs point of interconnection is set up to terminate the

traffic on the allocated channel.

FIGURE 5.3. Use case 3. MS to PSTN call with simulatedBTS and real MSC.

5.3.4 Use case 4

In this case, figure 5.4, we have a real BTS and a simulated

MSC. The test tool on the A interface is set up to generate

traffic on specific channels. The test tool connected to the A

B SCRea l

M S CSimulated

B TS

A bis AU m P O I

Test tool

B S CReal

M SCSimulated

B TS

A bis AU m PO I

T est tool T est too l


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Use cases


interface is setup to loop back the traffic frames. This is nec-

essary since this is not done in the simulated MSC. This

setup will simulate a MS to MS call.

FIGURE 5.4. Use case 4. MS to MS call with real BTS and

simulated MSC.

5.3.5 Use case 5

In this case, figure 5.5, we have a real BTS and a simulated

MSC. The test tool on the Abis interface is set up to gener-

ate traffic on specific channels. The test tool connected to

the A interface is setup to terminate the traffic frames. This

setup will simulate an MS to PSTN call.

FIGURE 5.5. Use case 5. MS to PSTN call with real BTSand simulated MSC.

5.3.6 Use case 6

In this case, figure 5.6, both the BTS and MSC are being

simulated. The basic function is to generate idle frames in

the test tool to simulate idle traffic channels.

B S CRea l

B T S

A bis AU m P O I

T est tool T est tool

Simulated

M SC

B SCSimulated

M SCRea l

B T S

A bis AU m P O I

T est too l Test tool


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Use cases


FIGURE 5.6. Use case 6. Simulated BTS and simulatedMSC with no traffic.

5.3.7 Use case 7

In this case, figure 5.7, both the BTS and MSC are simu-

lated. The test tool is connected to the Abis interface to gen-

erate traffic frames. It is also connected to the A interface to

loop back the traffic. This use case would simulate an MS to

MS call.

FIGURE 5.7. Use case 7. MS to MS call with simulatedBTS and MSC.

5.3.8 Use case 8

In this case, figure 5.8, both the BTS and MSC are being

simulated. The test tool on the Abis interface is set up to

generate traffic. The test tool on the A interface is setup to

terminate the traffic. This use case would simulate an MS to

PSTN call.

B SCSimulated

B TS

A bis AU m PO I

Test tool

Simulated

M SC

B SCSimulated

B TS

A bis AU m PO I

Test tool

Simulated

M SC

Test tool


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Requirements


FIGURE 5.8. Use case 8. MS to PSTN call with simulatedBTS and MSC.

5.4 Requirements

People from management, development, support and test

have been interviewed and the following basic requirements

for the system were found, table 5.2. The right column

shows where in the design chapter, the requirement has been

considered. This is also further described in the Ericsson

internal document [1].

B S CSimulated

B TS

A bis AU m PO I

Test tool

Simulated

M S C

Test tool


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Requirements


TABLE 5.2. Basic requirements of test system.

RequirementDesign

area

The traffic generator shall create correct frames

to be sent through the BSC.

6.2.2

Generated traffic shall be inserted primary on

the Abis interface and also secondary on the A

interface.

6.2.1

6.2.2

6.2.4

6.3.1

6.3.2

Contents of generated frames shall be selecta-

ble.

6.2.5

6.3.4Control of the traffic generator shall be done

from TSS.

6.2.7

6.3.3

Different speech coding algorithms shall be

supported.

6.2.3

Each instance of the traffic generator function

shall be identified by PCM-link, timeslot and

subchannel.

6.2.5

6.2.7

TRAU/TRAB frame parameters shall be

possible to set from TSS test programs.

6.2.2

6.2.56.2.7.5

Insertion of user defined frames shall be

possible.

6.2.5

6.2.7.5

6.3.4

Insertion of an external generated signal shall

be possible.

6.2.5

6.3.4

Configuration shall be possible to do dynami-

cally.

6.2.7.7


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Requirements



49/94


CHAPTER 6 Design

6.1 Proposed solution

The solution proposed here, is that the system is imple-

mented as an add-on to the existing TSS system. The unit

can be seen as a new feature adding value to the current

product.

The hardware platform to be used for the system is a

Motorola MVME2400 carrier board with the QPM/56 DSP

PMC Board [15] from Blue Wave Systems with Motorola

56301 DSP.

The unit will be placed as a component in TSS connecting to

one, or both of the A and Abis interfaces of the BSC, see

figure 6.1. It consists of the hardware it is running on and a

piece of software.

The design only covers the software running on VME hard-

ware, not test programmes running on the workstation.

The details of the design and implementation are also elabo-

rated in the Ericsson internal document [1].


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Functional overview


FIGURE 6.1. Solution running in TSS hardware.

6.2 Functional overview

This section describes the function blocks in the system. The

interfaces for connection to other systems are described in

section 6.3. The different functional blocks are shown in

figure 6.2.

FIGURE 6.2. Functional blocks in the test tool.

6.2.1 PCM Frame adaption

The PCM frame adaption module handles adaption of

speech information to and from the frame format used on

the A interface. This includes both a-law PCM coding for

SignalTerminalC on tro l T RA U

B S C

Line

Line

Abis

A

T SS

Test tool

PM C D SP

VME Carr ie r Boa rd

Board

PCM Frame

Adapt ion

T R A U / T R A B

Frame Adapt ion

PC Mconversion

Speechcodecs

TrafficGenerator

U I

AbisA

T S S

Test tool

Signal interface

Statistics


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Functional overview


European systems and -law PCM coding for American and

Pacific systems.

6.2.2 TRAU/TRAB Frame adaption

Different TRA protocols can be supported by implementing

these as separate modules. The modules must be able to

assemble correct frames for the TRAU or TRAB transmit-

ting them on the Abis interface.

For GSM, 40 byte frames are sent on 16 kbps subchannels

which are multiplexed into a 64 kbps channel. Information

about TRAU frames are found in [17] and [18]. For TDMA/

PDC 160 byte frames, consisting of three 53 bytes individu-

als, are transmitted over 64 kbps channels [19].

6.2.3 Speech codecs

Different speech codecs must be supported. Today mainly

the full rate and enhanced full rate codecs are used. How-

ever, in the future adaptive multirate might be of interest as

well [8]. The design is made so that new codecs are easily

added. The election of codec is made during setup of the

traffic generator for a specific channel.

Depending on how well simulations need to be done it is

possible to delimit the implementation of the speech codecs

to only encode/decode speech, omitting functions for voice

activity detection and discontinuous transmission. Howeverexisting codecs might also be used and adapted to the sys-

tem.

It might also be possible to have a null codec module. That

is a module that can be chosen just like any other codec but

it does not do anything with the signal. This can be useful if

one wants to insert an already encoded signal or to minimize

the demands for cpu processing time.


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Functional overview


6.2.4 PCM conversion

The speech codecs for GSM are normally defined for taking

13 bit linear PCM-coded speech as input. The traffic genera-tor therefore outputs a signal using this coding. A conver-

sion module will be needed to send traffic to the A interface.

On the A interface the speech is encoded according to

CCITT G.711 8 bit a-law PCM in Europe and 8 bit -law for

American and Pacific systems.

6.2.5 Traffic generator

There are several possible sources that might be used as

input signal used for testing the traffic channel. Some inter-

esting sources are:

Natural speech through headset.

Sampled signal read from file.

Internally generated signals.

User defined frames.

To be able to insert speech from a headset, speech will have

to be recorded and sampled on an external workstation and

streamed over the network. This will require extra software

for recording and streaming the speech on an workstation to

the test tool and is not part of this design, except for an inter-

face for receiving the sampled speech information.

Insertion of user-defined frames might be useful if one

wants to send DTMF tones through the system under test.

Se also section 6.3.4 for more information about inserting

external signals. Some aspects on signals useful for testing

purposes are also described in [20].

Common to all signal sources are that the generated signal

passes through the traffic generator. It is the module respon-

sible for producing traffic at the set-up rate on the correct


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Functional overview


channels. It will also select the correct signal source for each

channel.

During a normal conversation the line is only occupied by

speech approximately 50% of the time. This can be simu-

lated by implementing a model using distributions for calcu-

lating the length of, and interval between each speech

session, i.e. the number of frames filled with speech data

and the number of idle frames.

The traffic generator also acts as the signal destination.

Therefore it must be able to receive and terminate incoming

traffic that has been generated either by itself or by anotherinstance of the traffic generator. Terminating the traffic can

be done either by simply dropping the information, record-

ing it to file through the signal interface or comparing it to

the source signal and sending any error messages back to

TSS and to error logs.

6.2.6 Statistics

Statistics might be useful to determine the capacity of the

system and to see where and when bottlenecks occur. It

could collect information like:

number of incoming/outgoing packets.

number of idle/data packets.

number of erroneous packets.

number of specific frames.

Statistics would be reported to TSS on demand. This feature

has not been investigated because it has low priority from

the customer and it is not immediately needed.


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Functional overview


6.2.7 User interface

The user interface is the part responsible for the communi-

cation between TSS and the test tool. It shall be imple-mented using the TSS network interface[22] allowing

communication to and from TSS test programs.

Some message primitives and their parameters that would

be used are described in the following sections.

6.2.7.1 Setup traffic generator

This message is sent to the test tool from TSS test programto create and configure a traffic generator on a given traffic

channel. Parameters are:

PCM Link, timeslot, subchannel, speedTogether the PCM link number, timeslot, subchannel and

speed identifies the physical traffic channel to be used.

CodecUsed to indicate which speech codec to be used.

Source interfaceUsed to specify which protocol that should be used when

reading frames from the source channel. For example

TRAU frames on Abis interface or 8 bit PCM on A inter-

face.

Destination interfaceUsed to specify which protocol that should be used when

sending frames to the destination channel. For example

TRAU frames on Abis interface or 8 bit PCM on A inter-

face.

Signal sourceUsed to specify where to get the source signal from. It

could be generated internally or inserted through the sig-

nal interface.

ActionUsed to specify what to do with the received signal. It

can be forwarded to an external source through the signal


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Functional overview


interface, validated for correctness or simply just

dropped.

6.2.7.2 Remove traffic generator

This message is sent to the test tool from TSS test program

to remove a previously created traffic generator on a given

traffic channel. Parameters are:



6.2.7.3 Start traffic generator

This message is sent from TSS to the test tool to activate

traffic generation on the associated channel. Parameters are:



6.2.7.4 Stop traffic generator

This message is sent from TSS to the test tool to activate

traffic generation on the associated channel. Parameters are:



6.2.7.5 Frame request

This message is sent from TSS test program to the test tool

to insert a user defined frame on a traffic channel. Parame-

ters are:




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Communication interfaces


Frame dataThe user defined data to be inserted. The length will vary

depending on if it is a TRAU/TRAB on the Abis inter-

face or speech frame on the A interface.

6.2.7.6 Status indication

This message is sent from the test tool to the TSS test pro-

gram in order to signal an error condition. Parameters in the

message are:



Error codeSome unique identifier for this kind of error message.

6.2.7.7 Configuration

The final product should have configuration possibilities

similar to those in the GPRS GSL [21]. This however has to

be further investigated since it might involve other parts ofthe TSS and is therefore not be covered within this report.

6.3 Communication interfaces

6.3.1 A interface

This part of the test tool connects directly to the A interface

on the BSC. The test tool shall use allocated physical chan-

nels and fill them with traffic. It shall also listen to allocated

channels and terminate incoming traffic.

6.3.2 Abis interface

This part of the test tool connects directly to the Abis inter-

face on the BSC. The test tool will use allocated physical


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Communication interfaces


channels and fill them with traffic. It shall also listen to allo-

cated channels and terminate incoming traffic.

6.3.3 TSS Interface

This part of the test tool connects to the TSS test program. It

uses the TSS network interface for communication with

TSS. The messages sent are described in 6.2.7.

6.3.4 Signal interface

This interface shall be used if a signal is loaded from a fileor if a signal is sampled and played through a headset. The

input signal shall be encoded according to 13 bit linear PCM

coding and is parsed as a stream of bytes.

The interface is implemented as a generic link, using the

TSS network interface [22]. It will connect to a source,

whose location is given either through the configuration or

when the link is created.


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Software implementation


6.4 Software implementation

FIGURE 6.3. Software modules running on hardware.

The user interface module is implemented to be run on the

VME carrier board, and the other modules are implemented

to be run on a DSP, figure 6.3. The TSS and external traffic

generator can be run on their own boards or workstations.

This division makes it possible to use the TSS framework

[23] for implementing the user interface, hiding DSP details

from the user and increasing portability. The functions run-ning on the DSPs will be implemented in ANSI-C for porta-

bility.

Each DSP on the PMC module will run one instance of the

traffic generator manager and codec modules. For each

channel handled by the DSP there will be one traffic genera-

tor. The user interface module will be responsible for dis-

Host CPU

DS P D r ive r

SC 4000 D river E SS I D river

PCI Driver

T G M

T G TG T G

PCI Interface

ESSI Interface

ETG

Codecs andFrame adapt ion

QPM DSP Ifc

SCBus Ifc

Proxy

TG Link

U I

DSPs on PM C board

S C B u s

TS S

Signal Interface

TSS Interface

VM E Carrier board


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tributing requests for DSP resources among the DSPs to

reduce the load on each DSP.

Scalability is introduced in the meaning that it will be possi-

ble to switch to another DSP board with more DSPs, thanks

to the proxy in the user interface. Another feature is that it is

possible to use only a limited number of DSPs on the PMC

module, in order for the application to share the resources on

the DSP board with other applications.

6.4.1 External traffic generator

The location of the external traffic generator (ETG), is inde-

pendent of this implementation since it uses the signal inter-

face for communication with the test tool. The basic idea of

this module is that it shall deliver a bitstream representing

the signal to be transmitted by the traffic generators.

The external traffic generator can be implemented as a sim-

ple server reading a file and sending the contents to the test

tool. It might also be possible to control this module fromTSS. The exact details and features of this module are not

investigated further.

6.4.2 Driver modules

The DSP-, PCI-, SC4000- and ESSI Driver modules are

existing drivers from the DSP board manufacturer that are

being used for low level board control. These will be usedwithout modification.

6.4.3 Proxy module

Messages from TSS are routed through the proxy module, to

the correct DSP. On the DSP the traffic generator manager is

responsible for dispatching the message to the correct traffic

generator. Messages are either management messages like


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starting and stopping traffic generation or insertion of user

defined data frames, for example frames containing DTMF

signalling.

At start-up this module sets up the communication between

the DSPs and the carrier board. It will load code into the

DSPs and make them start execution. The code to be loaded

into the DSPs shall be possible to choose through the system

configuration.

Parts of the Proxy, the QPM DSP and SC4000 interfaces

might be reused from existing implementation of the GPRS

GSL layer, see [24] and [25].

6.4.4 Traffic generator links

This is a client link that will connect to an external traffic

generator. It will handle incoming data from the external

traffic generator and pack it into messages understandable

by the traffic generator manager (TGM), to separate them

from management messages.

It will also receive incoming data from the DSP side,

through the associated traffic generator. Data in received

messages will be extracted and forwarded to the external

traffic generator.

Expected data format from the external traffic generator is

13 bit linear PCM encoded speech, but any signal could be

inserted.

Note that it is not necessary for every traffic generator to be

associated with one link, but a link is always associated with

a traffic generator. This is because the link only exists for

traffic generators using an external traffic generator.


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6.4.5 Traffic generators

The purpose of this module is to generate a signal that can

be transmitted. This module must be invoked on a regularbasis - every 20 ms - to fill a timeslot on the PCM link. If

there is no data to transmit, idle frames shall be generated.

This means that the module has to implement at least part of

the TRAU/TRAB finite state automata protocols as

described in [26] and [27].

This module will also be responsible to decide what to do

with incoming data. When the channel is set up and a traffic

generator is associated with it, parameters must be providedto configure the traffic. It shall be possible just to drop

incoming data, to compare them with the original generated

signal or to forward them on the A interface or to the exter-

nal traffic generator.

Validation of data can only be done when a known sequence

of data bits is generated. Therefore the generator mechanism

will take a bit pattern as input and use it for both generation

and validation. In this way the generating and validatingtraffic generator can be remote (i.e not the same instance

and not even running on the same board).

The traffic generators will use the codec and frame adaption

library to pack and unpack data in correct frames.

6.4.6 Traffic generator manager

This module will contain the main loop in the DSP software.

It will receive management messages from TSS and dis-

patch them to the correct traffic generator. Typical there will

be one traffic generator manager running on each DSP and

several traffic generators, one traffic generator for each logi-

cal channel to generate traffic on.


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6.4.7 PCI Interface

This module will implement an interface for communication

over the carrier boards internal PCI bus. It will be reusedfrom the GPRS GSL protocol layer, see [24] and [25].

6.4.8 ESSI Interface

This module will implement an interface for switching of

data to 16 kbps channels. It will be reused from the GPRS

GSL protocol layer, see [24] and [25].

6.4.9 Codecs and frame adapters

This will basically be a library of functions used by the traf-

fic generators. It will consist of functions as discussed in

6.2.1, 6.2.2, 6.2.3 and 6.2.4 to implement the interfaces in

6.3.

The implementation of the enhanced full rate speech codec

follows the overview in figure 4.5 and figure 4.6. The detail

design is described in [2] and is intended to be an introduc-

tion to the source code.


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CHAPTER 7 Implementation and

Test

7.1 Implementation of test tool

The GPRS GSL protocol layer as developed by Enea Epact

for Ericsson can be used as a base for implementation of the

test tool.

There are several similarities between the functions of

GPRS GSL and the test tool.

They access the same type of hardware functions like theDSP and SC-bus.

The user interface has a well known structure and format.

Several modules for internal communication betweendifferent parts of the hardware can be reused with little or

no modification.

Documentation and personnel skilled with GPRS GSL

exist.

The test system proposed in previous chapters is not imple-

mented within the limits of this thesis due to lack of time.

Further discussion which parts can be reused, need to be

modified or implemented from scratch are described in [1].


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Implementation of speech codec


7.2 Implementation of speech codec

As a part of this master thesis a part of the proposed system

was implemented. The part chosen for implementation wasthe most commonly used speech codec - the GSM enhanced

full rate codec.

The implementation of the speech codec was divided into

two parts

Making the codec using several channels.

Making the coder run in the target environment.

7.2.1 Multichannel mode

From the basic implementation of the coder the possibility

to use several channels in parallel was implemented. This

was basically to provide several instances of the codecs

internal state variables, one instance per channel.

7.2.2 DSP implementation

From the start everything was implemented in ANSI-C to

achieve as much portability as possible. When entering the

DSP environment, performance became the crucial prob-

lem.

The most frequently used functions are basic operations like

addition, subtraction and multiplication. This also implies

composed functions like MAC and MSU. All these operationshad to be implemented in assembler as a first step to achieve

performance.

From Motorola there is a document [28] available, describ-

ing this procedure. This document is for use with another

compiler than the one used, but the basic ideas are still the

same.


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Tools


7.3 Tools

The tools used during development was Emacs and the

Suite56 for Motorola DSP 566xx/DSP563xx, available from[29] and [30].

The suite contains tools for compiling, linking and assem-

bling both C and assembler code. There are also modules for

simulation and hardware debugging. The simulator uses the

host machine to emulate all the internal functionality of the

DSP. This is very useful for debugging. However, the simu-

lator is very slow, about 1000 times slower than running on

hardware!

One of the most valuable tool in the suite were definitely the

profiler. From the reports generated by the profiler, it was

possible to find the bottlenecks in the code.

There is another compiler available from Tasking which

claims up to 40% smaller and more effective code [31]. The

compiler was not available for use during development. A

demo version of the product was tested for comparing per-

formance to the Motorola compiler, see 7.4. The main rea-

son for not using the Tasking compiler is that it is too

expensive compared to todays needs.

7.4 Problems

7.4.1 Performance of generated code

One of the goals were to write as much code as possible in C

in order to make the code more portable, however a lot of

assembler code had to be written for performance reasons.

Some of the performance problems can be derived from the

compiler used. The code generated is far from optimal.


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Problems


The compiler does not generate any hardware loops. For

every loop an external memory reference is used as a loop

counter, updated and checked against another memory refer-

ence. The DSP supports special hardware loops through the

DO assembler statement. The DO statement will use internalregisters for the loop counter, and does only require an over-

head of one instruction.

Several NOP instructions can be found in the generatedcode. However, several of these NOPs can be avoided byrescheduling of other instructions.

The parallel instructions in the DSP are not used. Using par-allel moves can speed up the implementation significant,

since data fetching can be done simultaneously as arithmetic

operations are performed in the ALU. The flag -alo usedat compilation with the Motorola compiler will enable the

assembly language optimizer to enable use of parallel

moves [32], but very few parallel moves are actually pro-

duced.

In appendix A there is assembler code generated from boththe Motorola and the Tasking compiler for one selected

function. The original C function is included in appendix B.

Comparing these files one can se that the Tasking compiler

makes use of the DO and DOR instructions on several occa-sions. The Motorola compiler instead uses memory loca-

tions to store the loop counters, loads them into the ALU,

increase the value and store them back in memory.

Several of the most time consuming functions could not be

compiled with the Tasking Compiler because of limitations

in the demo version used. The table 7.1 above shows that a

significant increase of speed/size would be reached by

changing compiler. As a comparison the number of instruc-

tions in the hand optimized functions are also included. The

differences between the compilers are propably even more

noticeable when compiling larger files.


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Problems


TABLE 7.1. Number of instructions generated by

compiler1.

7.4.2 Size of generated code

As can be seen in table 7.1 the Motorola compiler does notoptimize code very well. This also turned out to be a prob-

lem when the code had to be fitted into the memory of the

DSP. A tedious setup in the makefile for the project, made

the binary fit into the memory of the DSP. This will proba-

FunctionTasking

advancedTasking

compatible MotorolaHand

optimized

RatioMotorola/Tasking

advanced

cor_h_x 124 120 155 106 1.25

autocorr 157 141 219 101 1.39

pred_lt6 103 100 125 57 1.21

Pitch_ol 171 180 227 -- 1.33

Pitch_f6 105 110 134 -- 1.28

Lag_max 94 102 175 83 1.86

convolve 48 62 91 52 1.89

Syn_filt 83 93 115 62 1.36

1. Tasking compiler in Motorola compatibale mode used the following flags:

C-compiler: -Cacrs -M24xL -OAcefghijnoprswUvxyz -gn

Assembler: -Ogjmnprs -Rdrs -Jl -S -M24xL

Tasking compiler in advanced mode used the following flags:

C-compiler: -M24xL -OAcefghijnoprswUvxyz -gn

Assembler: -Ogjmnprs -Rdrs -Jl -S -M24xL

Functions compiled with the Motorola Suite56 compiler used the followingflags:

C-compiler: -S -mx-memory -o -S-alo -finline-functions -fforce-addr

Assembler flags for the Motorola compiler are included in flags to the C-compiler.

The Motorola compiler did not have any preset modes for compiling forspeed or size, therefore only one mode utilizing both speed and size has beenused, and compared to the Tasking Advanced mode.


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Problems


bly become a problem in the future if other functions are to

be fitted into the DSP as well.

7.4.3 Word length

The algorithm for the speech codec uses 16/32 bit calcula-

tions. All 32 bit operations are performed with simulated 32

bit instructions using only 16 bit instruction. The target DSP

has a 24 bit architecture, with possibility to perform opera-

tion in sixteen bit arithmetic mode.

This gives rise to several problems:

32 bit numbers and 24 bit memory word lengthOne problem occurring when using the 16-bit arithmetic

mode is when trying to read/write 32 bit of data from/to

memory. A 32 bit word requires two 24 bit words in

memory. When writin

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