Umeå University October 2005 Department of Computing Science Olov Holmlund

Master’s Thesis

Voice Quality Measurement over Bluetooth

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Abstract

TEMS Automatic is an autonomous system that measures the quality of mobile network from a subscriber perspective. One of the measurement probes in the TEMS Automatic is the Mobile Test Unit. Within the Mobile Test Unit audio quality measurements are performed with a digital signal processor, which compares two sentences. The audio quality algorithm calculates a score that describes how the subscribers experience the quality of the mobile network.

This Master’s Thesis has investigated how the audio quality algorithm in the Mobile Test Unit is affected by the Synchronous Connection-Oriented link in the Bluetooth standard version 1.1.

To manage this task, a prototype was built with a Bluetooth link as the transmission media in the Mobile Test Unit. Tests have been performed and the results have shown that the present audio quality algorithm in the Mobile Test Unit is severely affected when the Bluetooth link is used.

Keywords: Bluetooth, CVSD, MTU, PESQ

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Table of Contents

1 INTRODUCTION ...................................................................................................... 1 1.1 THESIS OUTLINE .................................................................................................. 2 1.2 PURPOSE AND MOTIVATION ................................................................................ 2 1.3 BACKGROUND ..................................................................................................... 2 1.4 METHOD .............................................................................................................. 3

2 TEMS AUTOMATIC................................................................................................. 5 2.1 DATABASE........................................................................................................... 6 2.2 OPERATOR CONSOLE........................................................................................... 6 2.3 COMSERVER........................................................................................................ 6 2.4 MOBILE TEST UNIT ............................................................................................. 6 2.5 CALL GENERATOR............................................................................................... 7 2.6 TEMS LOGFILE HANDLER .................................................................................. 7 2.7 TEMS PRESENTATION AND REPORT................................................................... 7

3 BLUETOOTH............................................................................................................. 9 3.1 GENERAL DESCRIPTION ....................................................................................... 9 3.2 BLUETOOTH RADIO LAYER................................................................................. 9 3.3 BLUETOOTH BASEBAND LAYER........................................................................ 10

3.3.1 Physical Links ........................................................................................... 11 3.3.2 Packet format and types ............................................................................ 11

3.4 ADDITIONAL LAYERS IN THE BLUETOOTH PROTOCOL STACK ......................... 12 3.5 BLUETOOTH AUDIO........................................................................................... 13

3.5.1 CVSD......................................................................................................... 13 3.5.2 PCM .......................................................................................................... 15

3.6 FREE2MOVE F2M03AC2 MODULE .................................................................. 16 3.6.1 PCM interface ........................................................................................... 16

4 PERCEPTUAL EVALUATION OF SPEECH QUALITY.................................. 19 4.1 MAIN OUTPUT FROM PESQ MOS-LQO............................................................ 19 4.2 PSYCHOACOUSTIC DOMAIN ............................................................................... 20 4.3 COGNITIVE DOMAIN .......................................................................................... 21

5 IMPLEMENTATION.............................................................................................. 23 5.1 DESIGN SOFTWARE REQUIRED FOR THE F2M03 MODULE................................. 23 5.2 AUDIO IN A TEMS SONYERICSSON V800......................................................... 24

5.2.1 Audio Path................................................................................................. 24 5.2.2 Acoustic Parameters ................................................................................. 25

5.3 BUILDING PROTOTYPE....................................................................................... 26 5.4 AQM IN A MTU INCLUDING PESQ .................................................................. 26 5.5 AQM USING THE PROTOTYPE IN THE MTU ...................................................... 27

5.5.1 PESQ-tester............................................................................................... 28 5.6 INTERFERENCE .................................................................................................. 29 5.7 USING USB BLUETOOTH DEVICES .................................................................... 30 5.8 TRANSMISSION STUDIES .................................................................................... 31

6 CONCLUSION ......................................................................................................... 33 6.1 FURTHER WORK................................................................................................. 33

7 ACKNOWLEDGEMENT ....................................................................................... 35 8 TERMINOLOGY..................................................................................................... 37 9 REFERENCES ......................................................................................................... 39

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List of figures FIGURE 1 SCHEMATIC PICTURE OVER A SYSTEM LAYOUT FOR TEMS AUTOMATIC. 5 FIGURE 2: THE LOWEST DEFINED LAYERS IN THE BLUETOOTH PROTOCOL STACK. 9 FIGURE 3: MASTER AND SLAVE FREQUENCY HOPPING. 10 FIGURE 4: DIFFERENT PICONET COMPOSITION IN A AND B; A SCATTERNET EXAMPLE IN C. 11 FIGURE 5: STANDARD BLUETOOTH PACKET ENTITIES. 11 FIGURE 6: VIEW OF THE CVSD ENCODER WITH SYLLABIC COMPOUNDING [22]. 13 FIGURE 7: VIEW OF THE CVSD DECODER WITH SYLLABIC COMPOUNDING[22]. 14 FIGURE 8: VIEW OF THE ACCUMULATOR ACTION[22]. 14 FIGURE 9: 16-BIT PCM SAMPLING WITH 8 KHZ SAMPLINGS FREQUENCY. 15 FIGURE 10: FREE2MOVE F2M03AC2 MODULE. 16 FIGURE 11: LONG AND SHORT SYNCHRONIZATION-PULSES, 8- AND 16-BIT WORDS. 17 FIGURE 12: VIEW OF THE PESQ ALGORITHM. 19 FIGURE 13: PSYCHOACOUSTIC PART OF THE PESQ ALGORITHM. 20 FIGURE 14: COGNITIVE DOMAIN OF THE PESQ ALGORITHM. 21 FIGURE 15: THE PC RESETS AND RESTARTS THE F2M03AC2. 23 FIGURE 16: PACKET FORMAT, COMMAND LENGTH AND COMAND PARAMETERS AREA. 24 FIGURE 17: A COMPARISON BETWEEN AN ORIGINAL SENTENCE AND A SENTENCE SENT THROUGH THE MOBILE NETWORK. 25 FIGURE 18: SIMPLIFIED AUDIO PATH IN THE SE V800. 25 FIGURE 19: EXCHANGING THE SE V800 WITH THE F2M03 MODULE TO CREATE A NEW CONNECTION WITH THE DSP. 26 FIGURE 20: THE RELATIONSHIP BETWEEN THE CG AND THE MTU IN AN AUDIO QUALITY MEASUREMENT. 27 FIGURE 21 INTERACTION BETWEEN THE PROTOTYPE MTU AND THE CG. 28 FIGURE 22 PESQ-TESTERS SYMMETRIC AND ASYMMETRIC VALUES. 29 FIGURE 23 SYMMETRIC AND ASYMMETRIC VALUES AFTER FILTER AND NOISE PROCESSING. 29 FIGURE 24 ASYMMETRIC AND SYMMETRIC VALUES WITH A USB BLUETOOTH TRANSMISSION. 30 FIGURE 25 SYMMETRIC AND ASYMMETRIC VALUES AFTER FILTER AND NOISE PROCESSING. 30 FIGURE 26 TRANSMITTED CORRUPTED WAVEFORM. 31 FIGURE 27 CVSD QUANTIZATION NOISE. 32 FIGURE 28 RELATIONSHIP BETWEEN THE AMPLITUDE AND MOS_LQO SCORE. 32

List of Tables TABLE 1: VOICE CODING PLAN SUPPORTED ON THE AIR INTERFACE. 13 TABLE 2: CVSD PARAMETER VALUES [22]. 15 TABLE 3: RELATIONSHIP, MOS VALUES AND INTENTION OF THE VALUES. 20

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1 Introduction

Many telecommunications companies around the world today are working in a saturated market, with decreasing revenues. Therefore, operators are trying to reduce costs, maximize usage of resources and maintain high quality in order to attract new customers while keeping existing customers satisfied.

The key to a satisfied customer is Quality of Service (QoS). In fact, good QoS can make the difference between a satisfied subscriber and a former subscriber. Because of that it is important for the operator to maintain control of the network quality. This is best achieved by a continuous gathering of live information about the network’s QoS, by observing the benefits of optimization efforts on a regular basis, and by measuring the quality as it is perceived by the customers.

To fulfill these requirements, operators collect information from the fixed-side of the network. While this data is necessary, it is generally used only for statistical analyses, and does not contain detailed information about the QoS nor the end user experience.

Another valuable source of information is the company’s own Customer Care Department (CCD). In the CCD, data regarding the problems that subscribers experience in the network is collected. However, very few subscribers will call the CCD when problems occur, but they may still consider switching to another operator. Therefore, the amount of dissatisfied subscribers obtained by the CCD may be a fallible source of information regarding the operator’s network.

An additional way for the operators to acquire information about the network quality is by performing a manual drive test. Drive tests provide essential information about the network, but are time consuming and costly. Therefore, they are limited by the time and resources available. Despite the necessity of records from the highest network load, which is mainly during rush hour and weekends, drive tests are seldom performed at these times due to the expenses.

It is here that autonomous test systems exceed the limitations of any other methods. The autonomous system provides the operator with realistic and reliable measurement data 24 hours a day using minimal human resources. The system uses a series of measurement probes emplaced in strategic places, providing end-to-end voice and data measurements.

Within each measurement probe in an autonomous automatic system, audio data is transported between different processes. One way to do this is by using the specified synchronous connection-oriented (SCO) link in the Bluetooth standard as the transmission media. The measurement probes commonly perform more or less advanced audio processing. Hence different audio quality algorithms have been developed to mimic the human audio perceptual capability, to get as realistic audio quality scores of the measurements as possible.

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1.1 Thesis outline

Because of the usage of internal confidential documents during the work with this Master’s Thesis, some of the chapters in this paper will focus on a discussion around problems proximate to the actual problems that have arisen during the work on this project.

The second chapter is an overview of the TEMS Automatic (TA), containing a short presentation of the processes in the system; what they do and the interfaces between them. Chapter three is a brief overview of the lowest layer in the Bluetooth stack. It will also provide a deeper insight into how Pulse Code Modulation (PCM) is managed before the air interface, how the audio channel is present, and finally a description of the Free2move module F2M03AC2. Chapter four gives a review of the different parts of the Perceptual Evaluation of Speech Quality (PESQ) algorithm, and how speech quality tests including Mean Option Score (MOS) are performed. Further, it will provide details on how it correlates to MOS-Listener Quality Option (MOS-LQO). The fifth chapter describes how the prototype is built and contains some information about the program that was produced to control the F2M03AC2 module. It also explains how an original Audio Quality Measurements (AQM) is achieved in the TA system and how the AQM are performed with the prototype. The final part of chapter five describes a test accomplished with two USB Bluetooth devices.

1.2 Purpose and motivation

Within a Mobile Test Unit (MTU), there is a mobile phone card and a Digital Signal Processor (DSP) card. These two cards communicate via wires that exchange Pulse Code Modulation (PCM) information, i.e. audio. The PCM is collected from a PCM bus of the mobile phone [23].

The audio information can also be redirected to the Bluetooth circuit in a mobile phone that is able to communicate with wireless devices like headsets. This study shall investigate whether it is possible to use the SE V800 Bluetooth circuit and an external Bluetooth circuit as the transmission media for the PCM information from the mobile phone to the DSP processor. The existing wired interfaces shall be exchanged to a Bluetooth solution, a Bluetooth prototype shall be constructed, and AQM values that deviate between the two transfer media shall be investigated.

1.3 Background

Most cellular phones today have the ability to transfer different types of information to and from the cellular phone via a Bluetooth interface. This interface has two different types of transmissions links [17, 18, 14]. One is a reliable link, where acknowledgements and retransmission schemes are used to ensure the reliability of the data transmission. The other is an unreliable circuited service link, which allocates slots in a periodic manner for the data transmission. The data transmitted is never acknowledged and never retransmitted. These two links have different purposes; the reliable one is used for file transfers while the unreliable link is intended to handle streaming data.

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As mentioned above, in the Bluetooth standard v1.1 [17, 18, 20] streaming data is never retransmitted because delayed data will interrupt the parts participating in the connection. In addition, excessive changes in the audio picture will cause degeneration of the audio quality, i.e. corrupt the audio information that is arriving at the receiver. Hence, audio quality algorithms have been developed to measure the degeneration in the audio between the sender and the receiver.

1.4 Method

The work started with the production of a time plan containing all the possible tasks and their respective time requirements. It also included an approximate plan for when the usage of different equipment would be needed.

A deeper study of papers delving into the PESQ [12] algorithm was made. Most of the information describing PESQ was found in the ITU-T P.862 Perceptual Evaluation of Speech Quality as well as the document AQM in TEMS Automatic –PESQ. The TEMS Automatic UMTS MTU700 Configuration Guide and the TEMS Automatic MTU700 Installation Guide were read to obtain an understanding of how the MTU is configured for AQM. The SE V800 PCM interface was also studied.

A prototype was built. The wired PCM solution between the phone card and the DSP card was replaced by a new link. This link was equipped with a Bluetooth device, which was used as the transmission media instead of the wired link.

A graphical user interface for the Free2move module was implemented. The Free2move evaluation kit F2M03AC2 [6, 9,24] (F2M03) was used as the Bluetooth module in the prototype. The DSP software was then exchanged due to incompatibility with the new interface from the Bluetooth device.

MTU test measurements were performed with the prototype hardware. The audio recordings were analysed with tools such as Microsoft Excel and PESQ-tester. These programs were able to provide statistical information about the audio quality and a sufficient overview of the sets of speech samples.

Most of the project documentation was performed in the last stage of the project, due to the many unknowns that required answers during the early stages.

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2 TEMS Automatic

TEMS Automatic (TA) is an autonomous automated system that gives an overview of networks from subscribers’ perspectives. It also provides reports and tools for troubleshooting and analysis. Because the TA system is a completely automatic system, it can execute measurements 24 hours a day, 7 days a week. This gives the operator the possibility to access measurements during rush hour and holidays when staff resources are limited.

The TA system is divided into several separate parts that interact with each other via specified interfaces. Each interface in the system is kept as simple as possible to make the system easy to understand, to maintain and to troubleshoot. Figure 1 below shows the interfaces between the processes in the system.

Figure 1 Schematic picture over a system layout for TEMS Automatic.

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2.1 Database

The database is the central hub in the TA system. Its interface is used by many different applications and since most of the system components are not real-time applications, the database is used as a communication node. This provides an option to build the system in a modular way, and an opportunity to run the system even though not all components have been completed.

The design of the database is simple; this is to enable easy access from different types of applications such as web-servers, report generators and third party products.

2.2 Operator Console

The Operator Console (OC) process is responsible for administrating information that is being handled in the system and gives the user an entry point to interact via. This entry point can be used to change settings or to send new information, Work Orders (WO), to the MTU:s. The OC only interacts with the database via an external interface.

2.3 ComServer

This part of the system serves as an advanced gateway for interaction between the database and the MTU. All information exchanged between the MTU and the ComServer is transferred via an ordinary File Transfer Protocol (FTP) server.

When the MTU sends information files to the ComServer, the ComServer will translate all incoming information to a format that fits the database. It will update the database with the new information; the location from which the information can be read and manipulated via the OC. Some collected files will be TEMS log files, which are to be placed in the directory for the TEMS log file reader for further processing. The ComServer is also responsible for unpacking any compressed files.

When the OC updates the database with information that belongs to the MTU, the ComServer will create new files including this content and send them to the MTU.

2.4 Mobile Test Unit

The MTU is a measurement probe in the TA System. The MTU is often installed in a vehicle such as a taxi, bus or delivery vehicle that moves around a network area. The MTU can also be installed into a fixed position such as a large arena. The MTU carries out measurements following a WO received from the ComServer. These measurements include, among other things, AQM with PESQ. After the WO has been completed within a fixed time frame, the MTU will send the measurement data to the ComServer via the FTP server.

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2.5 Call Generator

In the TA system the Call Generator (CG) is responsible for making and receiving calls from the MTU. The CG implements AQM including PESQ, and saves fixed line events that are recognized by the tones signaled from the Public Switch Telephone Network (PSTN). The measurement data is stored in the database by the CG, and when merged with information that the MTU stores in the log files, the user can pinpoint the geographical position of the fixed side event.

2.6 TEMS Logfile Handler

The task of converting the received air interface measurements is handled by a tool known as the TEMS Logfile Handler (TLH). It is also responsible for organizing the measurement data into the database. Voice quality measurements (VQM) from the uplink are merged with those from the downlink, which results in a geographic positioning of the VQM originating from the uplink.

2.7 TEMS Presentation and Report

TEMS Presentation models the measurement data collected in the MTU on a map, in spreadsheets or in line charts. TEMS Presentation can either display measurement routes or statistics drawn from the raw data. These statistics are used to configure and generate the statistical database. The purpose of the statistical database is to be able to distinguish trends in the measurements over a certain period of time or to identify problem areas by analyzing vast amounts of data. [8]

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3 Bluetooth

The first two sub-chapters under this heading describe the lower parts of the Bluetooth stack and how the Bluetooth entities can form different kind of deployments. Packet types and different physical channels are also presented. The part of the chapter where audio is discussed contains a detailed declaration of the Bluetooth audio air interface. This has been included because it has a very deep impact on the result of the Master’s Thesis. The four layers above the radio layer and the baseband layer included in figure 2 are also shortly described. In the end of the chapter, there is a declaration of the Free2Move handsfree module used in the prototype.

3.1 General description

Bluetooth is a short-range radio standard (10-100 meter), intended to replace cable(s) between portable and fixed electronic devices. Each entity that will communicate has to be equipped with a Bluetooth circuit, where the chip works as a transceiver. To be able to communicate, all Bluetooth entities have to work together. One entity will act as a master, which controls the other entities that are participating in the communication. These controlled entities are known as “slaves”. Bluetooth is developed to be a robust and reliable link, which can operate in noisy environments at a low cost [17].

3.2 Bluetooth Radio Layer

The Bluetooth radio layer is the lowest defined layer in the standard. Figure 2 exhibits the lowest layer in the Bluetooth stack. The transceiver operates in the license free 2.4 GHz Industrial Scientific Medicine (ISM) band. In a majority of countries, this frequency band reaches from 2400 – 2438.5 MHz. The Bluetooth wireless technology uses a frequency-hopping/time division duplex scheme. It can hop to a maximum of 79 different frequencies; a space of 1 MHz separates each hopping frequency. The time between two hops is called a slot, and the Bluetooth entity makes 1600 hops per second. I.e. each slot uses different frequencies, and remains at a specific frequency for 625 µs. Some countries, however, have limitations in their frequency range, which forces the Bluetooth entity to adhere to a reduced hopping range [21].

Host controller interface

Figure 2: The lowest defined layers in the Bluetooth protocol stack.

Gaussian Frequency Shift Keying for modulation is used in the transceivers and is available in three different classes.

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Power Class 1: Has a maximum transmission power of 100mW and a minimum of 1mW. Power control is mandatory. A class 1 entity usually ranges up to 100 m without impediments.

Power Class 2: Has a maximum transmission power of 2.5mW and minimum of 0.25mW. Power control is optional. A class 2 entity can range up to 10 m without obstacles.

Power Class 3: The sole requirement is that an entity has a maximum transmission power below 1mW. [18]

3.3 Bluetooth Baseband Layer

The Bluetooth baseband layer is fairly complex [18, 22]; it handles the “pseudo random frequency hopping sequence” which limits interference and handles all access to the medium. (The hopping sequence is defined by the master address while the master clock determines the phase in the hopping sequence.)

The Bluetooth baseband layer also defines different physical links and a large numbers of packet formats. Figure 3 shows different frequency sections for slot packets allocating 1, 3 and 5 slots. Each slot has a duration of 625 µs. The master sends data at a frequency fk, which only allocates frequencies at an even number. The slave subsequently responds in the frequency fk+1, hence the slaves use only odd numbered frequencies. This odd and even schedule shows a point of Time Division Duplex (TDD) and is used for separating the transmission directions in Bluetooth.

The multi-slot packets defined in the Bluetooth standard, as described above, cover three or five slots in order to achieve a higher data rate. When a master or a slave sends a multi-slot packet, the transmitting entity will remain at the same frequency during the transmission. Under these circumstances, no frequency hopping is performed within the packet transmission. During a multi-slot packet transmission, the clock is running in every entity, thus changing the frequencies. Therefore, after the transmission has been completed, the radio entities will remain synchronized with the other entities that are participating in the communication. Devices that share the same “pseudo random frequency hopping sequence” are also sharing the same channel.

Figure 3: Master and slave frequency hopping.

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When two or more devices are using the same channel, they will form a Piconet [20]. In the Piconet there is one master and up to 7 slave devices. The hopping sequence is unique for each Piconet. Figure 4 shows different types of piconets.

Figure 4: Different piconet composition in A and B; a scatternet example in C.

Multiple Piconets with overlapping coverage areas form a scatternet. Each Piconet can only have one single master; however, slaves can participate in different Piconets simultaneously on a time-division multiplex basis. In addition, a master can participate as a slave in another Piconet. This is only possible when the different Piconets participating in the scatternet are not frequency synchronized.

3.3.1 Physical Links

In the Bluetooth baseband, two types of links are defined [18, 20, 22]; the SCO and the Asynchronous Connection-Less (ACL) link.

The SCO link is a point-to-point connection between a master and a specific slave participating in the same Piconet. The link has reserved slots and can therefore be considered a circuit-switched connection. SCO packets are never retransmitted and are intended for speech transmissions. Every SCO link has a transmission capacity of 64 kB/s.

The ACL link is a point-to-multipoint packet-switched connection between the master and all active slaves participating in the Piconet. In slots not reserved for a SCO link, a master can establish an ACL connection on a peer-to-peer basis. This applies to any slave, even a slave that is already engaged in an SCO link. For all of the ACL links there is a packet retransmission possibility, which assures data integrity.

3.3.2 Packet format and types

The data packets presented in the Piconet channel consist of three different fields: access code, header and payload. The access code and header have a fixed size; 72 and 54 bits respectively. The payload can carry 0-2745 bits[18]. Figure 5 shows the different fields in the packets.

Figure 5: Standard Bluetooth packet entities.

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The packet types in the Piconet are related to the physical links they are being used in.

Four different packages are defined for the SCO link; HV1, HV2, HV3 and a DV-packet [17, 18, 20]. HV stands for High-quality Voice. The HV1 packet carries 10 information bytes, which represents 1.25ms of speech. The HV1 packet is encoded using 1/3 Forward Error Correction (FEC). The HV2 packet carries 20 information bytes (2.5 ms speech) and is encoded using 2/3 FEC, while the HV3 packet, which is also encoded with 2/3 FEC, carries 30 information bytes (3.75 ms speech). These bytes are not protected by FEC. A HV packet is never retransmitted. The DV is a combined voice and data packet, which carries 10 bytes of voice information and up to 150 bits of data. The voice field of the payload is not FEC protected, and like the HV packet, it is never retransmitted. The data field is protected by a 16-bit CRC and encoded with 2/3 FEC.

The ACL link has seven different packages defined as DM1, DM3, DM5, DH1, DH3, DH5, and AUX1. The Data-Medium (DM) packets only carry data information. DM1, DM3 and DM5 cover one, three and five time slots respectively. In the same order they contain 18, 123 and 226 information bytes. All packets have a 16-bit Cyclic Redundancy Code (CRC), and 2/3 FEC. The Data-High rate packets are similar to the DM packets, except for the lack of FEC encoding in the payload information. As a result, the DH1, DH2 and DH3 packets can carry up to 28, 185 and 341 information bytes. In the same manner as the DM packets, the DH packets cover one, three and five time slots respectively. [18, 20]

The AUX1 packet looks like the DH1 packet but has no CRC code. The AUX1 packet can carry up to 30 information bytes, and it covers only one slot [18].

3.4 Additional Layers in the Bluetooth Protocol Stack

The Link Manager Protocol

The Link Manager Protocol (LMP) is responsible for establishing a link between two Bluetooth devices and controlling the baseband packet sizes. It is also responsible for the power control modes and the security issues in a Bluetooth connection. LMP messages between two devices are filtered out and interpreted; they have higher priority than a data packet and are never delayed [17].

L2CAP

The tasks for Logical Link Control and Adaptation Protocol (L2CAP) include multiplexing for higher layer protocols, reassembling and separating packets and QoS. It allows higher layers to transmit packets with a length up to 64 kilobytes and it only supports ACL packets [17].

Service Discovery Protocol

The Service Discovery Protocol (SDP) is a very important framework that discovers information about other devices, such as which services are allowed and what characteristics of the detected services can be queried. After a device has detected the services in the surrounding area, the user can choose to establish a connection with the device [17].

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RFCOMM

The Radio Frequency Communication (RFCOMM) is a protocol that emulates the RS-232 serial port. Due to its dependency on the underlying L2CAP layer, which is used for multiplexing, the RFCOMM layer is forced to use the ACL link for transmission. The higher layer in the Bluetooth protocol stack, e.g. the Object Exchange Protocol, manages the transportation capacities of the RFCOMM layer [17].

3.5 Bluetooth Audio

The Bluetooth air interface can have either a 64 kB/s log PCM format (A-law or µ-law) or a 64 kB/s Continuous Variable Slope Delta modulation (CVSD)[17, 18, 20]. What type of air interface will be used depends on the input data stream to the entity. Table 1 summarizes the interfaces supported by the air interface.

Voice Codecs

Linear CVSD A-law 8-bit logarithmic µ-law

Table 1: Voice coding plan supported on the air interface.

3.5.1 CVSD

CVSD modulation is a method used to convert a speech signal into a digital format [22]. It takes advantage of the fact that voice signals do not change unexpectedly [17]. The CVSD modulation schedule tries to follow the waveform. The sgn(x) function returns 1 if x ≥ 0, however, if x < 0, then the function will return -1 instead. These numbers represent 0 and 1 respectively on the air interface. I.e. the CVSD algorithm uses a prediction value; if the input value is larger than the predictor, a 1 is generated as output in the air interface, whereas a 0 is generated if the input value is lower than the prediction value. This process is illustrated in figure 7.

The step size control is applied to reduce slope overhead effects [20]. The step size is adjusted according to the average signal slope. The input to the CVSD encoder in the Bluetooth standard has to be a 64 kB/s linear PCM. Figure 6 exhibits the CVSD encoder. Figure 7 shows the decoder and figure 8 describes the accumulator [20].

Figure 6: View of the CVSD encoder with syllabic compounding [22].

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Figure 7: View of the CVSD decoder with syllabic compounding[22].

Figure 8: View of the Accumulator action[22].

Notice the constants in the figure on page 14. The CVSD encoder output is b(k), the accumulator substance is y(k) and the step size is d(k). Also pay attention to the decay factors, where the representative of the step size decay is β and the decay factor for the accumulator is denoted by h. These decay factors are definite as shown in table 2. The step size parameter is denoted by α, which monitors the slope by considering the four most recent output bits.

Let

)()(ˆ khykx = (1)

The different steps in the CVSD algorithm are then updated according to the following equations:

}{ ,)1(ˆ)(sgn)( −−= kxkxkb (2)

=otherwise ,0

equal, are bitsoutput four last in the bitsfour if ,1α

}{

}{

=−=+−

=,0 ,),1(max

1, ,,)1(min)(

min

maxmin

αβα

dkdddkd

kd (3)

}{

}{

<≥

=0. (k)y ,y(k),ymax0.(k)y ,),(ˆmin

)(min

maxykyky (4)

where:

).()()1(ˆ)(ˆ kdkbkxky +−=

The minimum and maximum step sizes are denoted by dmax and dmin , and ymax and ymin denotes the accumulator’s negative and positive saturation values respectively [22].

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The bits are transmitted in the same order as they are generated over the air. Table 2 below shows the different values that the parameters must use, and the parameter values are settled into different locations that are determined by a 16-bit signed number output from the accumulator.

Parameter Value Parameter Value h 1-1/32 β 1-1/1024 dmin 10 ymin -215 or –215+1 dmax 1280 ymax 215-1

Table 2: CVSD parameter values [22].

3.5.2 PCM

The first step when converting an analog speech signal to a digital one is to filter out the high frequency components in the signal. This is possible because most of the energy in spoken language measures between 200-2800 Hz [26]. A band-limiting filter is used to reduce aliasing. The second step is to read (sample) the amplitude of the analog curve in a manner that results in good audio quality. The sampling will result in a Pulse Amplitude Modulation (PAM) where each pulse corresponds to an amplitude in the analog curve. The sampling frequency is determined by the Nyquist criterion [26, 5], which says that the sampling frequency has to be at least twice as high as the highest frequency in the original signal.

signal voiceanalog theofBandwith BW

Frequency Sampling FS)(2

==

BWFS }

Figure 9: 16-bit PCM sampling with 8 kHz samplings frequency.

The next step in the process is the quantization step. Each input sample is mapped into the quantization interval that is the closest match to the amplitude height. If the quantization interval does not match the actual amplitude of the input signal, an error is introduced into the PCM. This error is called quantization noise. A way to reduce the error is to increase the quantization intervals. The last step in the Analog Digital (AD) transformation is the coding phase, where each quantization value is expressed as a binary code, each code consisting of 16-bits.

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The µ-law and the A-law are two different types of compression schemes, which compress the 16-bit linear PCM data down to eight-bit logarithmic data.

Since the air-interface supports a 64 kb/s information stream, it is possible to apply either the A-law PCM or µ-law PCM compression. If the line interface uses A-law and the air interface uses µ-law or vice versa, a conversion from A-law to µ-law is made. Both A-law and µ-law follow the ITU-T recommendations G.711[22]. If the PCM is represented in 16-bit, i.e. linear PCM, CVSD modulation is used instead of the A-law PCM or µ-law PCM.

3.6 Free2Move F2M03AC2 Module

The F2M03AC2 module is of power class 2. The module is a surface mountable Bluetooth system, and is equipped with two different interfaces for audio. One is an analog voice interface and the other is a PCM digital audio interface. The module can be equipped with a number of different firmware versions [6]. The module in this Master’s Thesis is equipped with a standalone headset firmware.

Figure 10: Free2Move F2M03AC2 module.

3.6.1 PCM interface

To transfer PCM-data in a wired manner, five different wires represent the PCM data-bus, and if two entities will share PCM-data between each other, they have to be configured in a master to slave relationship. I.e. in a wired solution, one entity will have the master role in the connection and the other entity will be the slave. The master’s role is to synchronize the slave to the clock of the master, and to generate sync pulses.

As mentioned earlier the PCM is a standard method to digitalize human voice patterns for transmission in digital channels. The F2M03AC2 has hardware support for transmitting continual PCM data. The data will not pass through the HCI layer of the protocol stack; it is only managed in the radio- and baseband layers. The SCO links in the F2M03AC2 are designated to send and receive streaming mono audio and voice data particularly, and the module can handle up to three different SCO connections at one time.

When the F2M03AC2 entity operates as a master in the PCM interface it can generate an output clock at 128, 256 or 512 kHz (it only generates clock pulses during the time when a SCO link is established). When configured as a slave it supports input clocks up to 2048kHz.

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The F2M03AC2 follows the Bluetooth standard v1.1 and supports four different types of sampling formats; 13- and 16-bit linear PCM, and 8-bit A-law or µ-law. The two latter are coded formats with 8000 samples per second. In the first three of the four primary slots following the PCM sync, the module is able to transmit or receive.

The module is equipped with a headset firmware that supports a 16-bit linear PCM format, which force the module to use CVSD transformation. The different types of PCM formats that the module supports are directed by the firmware, which is flashed into the module.

There are two types of synchronization pulses in the F2M03AC2 module; Long frame synchronization and short frame synchronization. In the long frame synchronization the rising edge of the synchronization pulse indicates the start of a data word. The long frame synchronization is always 8 bits long. In the short frame synchronization, however, the falling edge of synchronization pulse indicates the start of the PCM word. The short frame synchronization is always 1 bit long, and as mentioned, the device will only produce the synchronization pulses when the F2M03AC2 is configured as a master [6].

It is also possible to configure which bit shall be sent first in the data word; the most significant or the least significant [6]. Figure 11 gives an overview of the different types of synchronization pulses and the different word lengths which can be managed in the F2M03AC2 module. Before and after the PCM input shown in the figure below, the data is undefined. This is because the F2M03AC2 module is unable to control the actions performed on the other side of the connection before and after a data word is sent [6].

Figure 11: Long and short synchronization-pulses, 8- and 16-bit words.

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4 Perceptual Evaluation Of Speech Quality

The purpose of this section is to give an overview of the Perceptual Evaluation of Speech Quality (PESQ) [12] algorithm, as well as how the audio quality measurements are performed and what the goal of the algorithm is.

PESQ is an objective method for a subjective quality evaluation of mobile networks. This means that PESQ is a mechanism to calculate how the general public experiences the speech quality in the mobile network. PESQ does this by mimicking the human speech perception. It evaluates the distorted speech signal (the signal that is transmitted via the cellular network) by comparing it to the original undistorted reference signal.

Figure 12: View of the PESQ algorithm.

PESQ is divided into two parts; a psychoacoustic domain and cognitive domain [3]. The psychoacoustic domain mimics how humans experience speech. This procedure produces a PESQ-score extending from 1 to 4.5, where a value of 1 indicates a very hard distorted speech signal and a value of 4.5 asserts that the measured speech has no distortion [3, 10].

PESQ has several advantages comparing to older audio quality algorithms. Unlike Perceptual Speech Quality Measure (PSQM) [4], PESQ has a time align procedure, which handles Voice over Internet Protocol (VoIP) and handovers. PESQ is also superior to PSQM in that it removes filtering effects. PESQ is however very sensible to all types of audio transformation and react noticeably to added noise.

4.1 Main output from PESQ MOS-LQO

The Mean Option Score (MOS) [11] is frequently used for a subjective evaluation of speech-encoders and -decoders. In a MOS test, a listener grades a speech sample normally five to eight seconds long, by assigning it to one of the following categories in table 3:

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MOS Intention 5 Excellent 4 Good 3 Fair 2 Poor 1 Bad

Table 3: Relationship, MOS values and intention of the values.

The intentions in the MOS table are only specified in terms of excellent, good, fair, poor and bad. There are no reference audio samples for the listener to relate them to. Therefore, each listener individually decides what constitutes a “fair” speech sample. This loose definition of MOS makes it very sensible to utilize alternative listening procedures, as the listener’s prior experiences, equipment quality et cetera can affect the listener’s interpretation of the MOS values.

The output from the PESQ algorithm is a MOS-Listening Quality Objective (MOS-LQO) [10] and not a MOS value. This means that the MOS value is transformed to a MOS-LQO value according to the ITU-T p.862.1 standard.

4.2 Psychoacoustic domain

As mentioned, the PESQ algorithm is divided into different parts. The most important steps in the psychoacoustic region of the algorithm are described below. They are also illustrated in figure 13.

Figure 13: Psychoacoustic part of the PESQ algorithm.

Scale: When performing system tests, the gain of the system may vary considerably. For instance, how much the system gains is affected by whether the system uses an ISDN-line or if an analog two-wired interface is used to perform the measurements. For this reason, the transmitted speech and reference speech are both scaled as a means to compensate for the overall gain in the network.

Time align: Transmission delays may occur in a mobile network. They can change the transmitted sentences either in a single speech reference or between two speech references. The delays occur as a result of handovers or VoIP. Both the transmitted speech sentence and the reference sentence are time aligned; therefore all parts of the transmitted sentence continuously correspond to the reference and vice versa.

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Mimic ear resolution: The speech signal is converted into the frequency domain. Next, the Hertz scale is warped into the critical band domain by attempting to mimic how the ear treats different frequencies. Thus higher frequencies are given a lower resolution.

Remove filter influence: Filtering in the PSTN or mobile network may have a negative effect on the PESQ score, because severe filtering disturbing to the listener. To decrease the filter influence, the transfer function is measured and this information is used to equalize the reference.

Remove gain variations: Gain variations may occur because of the Automatic Gain Control (AGC) units in the network. The effect of gain variations is removed.

Mimic ear-brain loudness perception: In order to imitate how the human ear transforms intensity into discerned loudness, the intensity of the spectrum is warped.

4.3 Cognitive domain

Figure 14: Cognitive domain of the PESQ algorithm.

Perceptual subtraction: In order to obtain a disturbance density signal, the volume representation is subtracted from the reference and transmitted signals. The brain’s understanding of differences in volume is taken into account.

Identify bad intervals: An incorrect time alignment for a speech interval can sometimes cause very bad disturbances, affecting the disturbance density signal. By re-computing the time alignment and rest of the PESQ algorithm, it is possible to replace the bad interval with an improved version if the altered interval has a better disturbance signal.

Asymmetry processing: Noise can be added to the original speech by speech codec, resulting in clearly audible distortion. An asymmetric disturbance density signal, including added disturbances, is calculated by the asymmetry processing.

Aggregate disturbances for all of the speech: By adding the disturbance signals and the asymmetric disturbance signals in the frequency plane, signals representing the amount of speech distortion are achieved. These signals are representative of the speech distortion during a very short time interval. When these short periods of time are summed to blocks of 320ms, they constitute what is referred to as split second disturbances.

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By combining the average split second disturbance with the average split second asymmetrical disturbance for the entire speech reference, a PESQ_MOS score can be calculated.

Transform to MOS-LQO: The PESQ-score [10] is transformed into the MOS_LQO score according to ITU p.826.1.

MOS-LQO: MOS_LQO is similar to the MOS scale. The MOS-LQO scale goes from 1 to 4.5, where 1 is the worst value and 4.5 is the best.

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5 Implementation

This section aims to describe the bigger problems that have occurred during the work with this Master’s Thesis. The main input devices and hardware used for the work on this Master’s Thesis are the MTU 700, a SonyEricsson V800 (SE V800) equipped with Ericsson TEMS software, and a Free2Move Bluetooth module F2M03AC2. Two Broadcom USB Bluetooth dongles were also utilized.

5.1 Design software required for the F2M03 module

The Bluetooth module F2M03AC2 was connected to an evaluation board [6] designed to make the module more user friendly. Using request, respond and indication protocols [9], the module could be controlled via a serial port. All sent requests had to receive a response before a new request could be sent. An indication could be sent from the F2M03AC2 module at any time. It indicated any changes that had affected the module. Figure 15 shows what a restart of the module could look like.

Figure 15: The PC resets and restarts the F2M03AC2.

A program has been produced in order to enable governing of the F2M03AC2 module via a PC. The module protocol communicates with hexadecimal values that describe the status of the module. This hexadecimal protocol communication is translated into a text arrangement, and it is presented in a graphical user interface. The main task for the software is to control the setup of the audio link in the prototype, and to give feedback to the user about the status of the audio link. The software contains the possibility of forcing the module to change the audio packet types in the audio link. The default packet type is a HV1 packet.

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The packets in the protocol are divided into three parts. The first part is a command value consisting of 1 byte of information. This provides the module with commands that are waiting to be executed, but it can also contain information from the module about the tasks that have already been completed. The second part of the packet is a length indicator that tells how many bytes the command parameters consist of. This part of the packet, like the command value, constitutes 1 byte. The third part contains command parameters describing whether the commands have been processed under normal circumstances or if there were any deviating occurrences. Figure 16 shows the structure of a packet in the protocol.

Figure 16: Packet format, command length and comand parameters area.

5.2 Audio in a TEMS SonyEricsson V800

All MTU:s are equipped with a mobile phone, which is flashed with TEMS phone software. There are numerous differences between TEMS and the original software, and as a result of this, the phone in the MTU can give more information than a mobile phone distributed to the end user market.

5.2.1 Audio Path

The SE V800 supports several audio modes [2]. The mode used in this Master’s Thesis is a normal voice mode. It includes functions that are able to perform audio decoding, audio mixing and filtering. All of these transformations are completed before the digital audio signal reaches the Bluetooth circuit. Most of the audio processing units are turned off, in order to keep the incoming audio picture as consistent as possible.

Some of the phone band filtering and voice coding cannot be completely turned off due to aspects of the mobile network that are out of our control, therefore this will to some extent affect the audio coming from the mobile. Figure 17 is a diagram that compares the frequency scan of a reference sentence in the MTU and the same sentence after it was sent through a mobile network. The figure 17 also displays the effects of the phone band filters and voice codec. Figure 18 shown below is a model of the path the audio takes from the mobile network to the Bluetooth circuit in the SE V800.

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-90

-80

-70

-60

-50

-40

-301 51 101 151 201 251 301 351 401 451 501

Hz * 4

dB

Original SentenceSentence Sent

Figure 17: A comparison between an original sentence and a sentence sent through the mobile network.

Figure 18: Simplified Audio Path in the SE V800.

5.2.2 Acoustic Parameters

In the file system inside the SE V800, several files belong to the audio configuration [1]. These are so called acoustic parameter files. Most of these files manage the routing of the audio path. Two of these files have been modified during the work on this Master’s Thesis. One corresponds to the access type i.e. this parameter determines how the audio is routed in the mobile, and is also responsible for the time and location from where and how the data can be reached in the mobile. The main task for this file is to provide the Bluetooth entity in the SE V800 with a constant PCM-data stream.

The second parameter file that has been altered, manages how the SE V800 behaves when its clamshell is either opened or closed. When the mobile is inside the MTU the clamshell has to be closed due to the lack of room. During the work with the prototype, it was instead required that the clamshell stay open. This was necessary in order to be able to pair the phone with another Bluetooth entity originating from the MTU.

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5.3 Building prototype

During the designing phase of the prototype, circuit schemes designated for the MTU and the F2M03AC2 module were studied. In particular, two different options for routing the new PCM-data stream were of interest. The first option required an investigation of a new way of routing the PCM-data into the DSP, but because it was difficult to estimate the amount of time that would be required to create the new route, and because of the strict time limitations for this Master’s Thesis, this option was cancelled at an early stage. Therefore, the remaining option was to route the audio using the existing way from the phone to the DSP. The existing wires connecting the mobile phone to the MTU were removed at this stage and the F2M03AC2 PCM-bus was directly connected to the DSP PCM-bus.

The 16-bit linear PCM-interface from the SE V800 was copied to the F2M03AC2 module so that it would fit the existing DSP software. However, the PCM-interface in the F2M03AC2 module was not copied in its entirety due to compatibility problems, and therefore the differences between the SE V800 and the F2M03AC2 module in the PCM interface had to be managed by the DSP software. The existing DSP software was replaced by a modified version to suit the interface in the F2M03AC2 module.

Notice that the SE V800 phone card was never removed from the MTU; only the wired PCM-bus between the phone and DSP was taken out. The phone in the MTU had two tasks; it was used to handle calls to- and from the CG during all tests and it also managed one end of the Bluetooth connection. Figure 19 gives a schematic picture of the exchange from the SE V800 PCM-bus to the new F2M03AC2 PCM-bus.

Figure 19: Exchanging the SE V800 with the F2M03 module to create a new connection with the DSP.

5.4 AQM in a MTU including PESQ

The ordinary AQM procedure between the CG and a MTU is performed in a master to slave manner. Both the master and the slave follow a half duplex timetable where the master (CG) and the slave (MTU) alternate between playing and recording speech sentences. The master decides the playing and the recording scheme and the slave has to follow it [3].

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Every sentence in the TA system is 5.5 seconds long. The CG plays the first sentence for 5.5 seconds and the MTU records the transmitted speech sentence. In the following 5.5 seconds, the roles are reversed, i.e. the MTU plays a sentence and the CG records it. Hence, each play and record cycle will take 11 seconds. [3]

For every recording, three quality scores are calculated [3]. Two of these scores are Frequent AQM values; one score is calculated for the first half of the recorded sentence, while the other reasonably calculates a score for the remaining part of the sentence. As the sentence is 5.5 seconds long, each half will constitute 2.75 seconds. The third value that is calculated is the PESQ score, which is calculated based on the entire sentence.

On occasion, the measurements do not result in any scores, because the recordings contain more than 25% of silence [12]. The PESQ algorithm cannot synchronize these sentences.

The scores on the CG side are based on the uplink, while the MTU manages the measurement scores from the downlink. All generated scores are saved to log files. Figure 20 gives a schematic picture of the measurement procedure between the CG and the MTU.

Figure 20: The relationship between the CG and the MTU in an audio quality measurement.

5.5 AQM using the Prototype in the MTU

All test measurements where the prototype has been involved, have been performed in a manner similar to the original measurement procedures as described above under the heading “AQM in a MTU including PESQ”. However, during measurements involving the prototype, the CG has only played sentences and the MTU only recorded and performed AQM. I.e. The main difference between the original procedures and the ones used for the prototype is that the prototype has only made downlink measurements. This is simply because it is much easier to acquire measurement data from the MTU than it is from the CG.. Figure 21 shows how the sentences are played and recorded when making AQM with the prototype.

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Figure 21 Interaction between the prototype MTU and the CG.

One of the log files is the Trace log, which is sent to one of the serial ports in the MTU during the AQM. The Trace log is transmitted to a PC and is presented to the user via the MS-HyperTerminal program. This log contains both the Frequent AQM and the PESQ score for all downlink sentences used when making AQM. The names of the recorded sentences are also displayed. The recorded sentences in the MTU are saved on a Compact Flash disc, where each sentence receives an individual name. Thus, if any PESQ score deviates from the average score, it will be possible to capture the recorded sentence for further analysis.

5.5.1 PESQ-tester

PESQ-tester is a program intended for a PC, which uses the same algorithm as the PESQ software included in the MTU. The advantage of the PESQ-tester program is that, on top of the Frequent AQM score and PESQ score, it is also able to produce two vectors. One is a symmetric vector, that shows how much of the original sentence is withdrawn during the transmission. The other is an asymmetric vector containing the disturbance density signals, i.e. it describes how much noise is added to the transmitted speech.

The sentences recorded during AQM with the MTU are collected and inserted as arguments to the PESQ-tester. The PESQ-tester program takes three arguments on a command line. The first argument communicates how the reference- and transmitted speech sentences are sampled. The other two arguments consist of the reference and distorted (recorded) sentences respectively.

The output vectors, that are a result of the tests run in PESQ-tester, are often described in a chart, which make it is easier to pinpoint the exact locations of the symmetric and asymmetric parts in the transmitted sentence. Figure 22 shows a chart of the symmetric and asymmetric values taken from a measurement using the prototype.

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Figure 22 PESQ-testers symmetric and asymmetric values.

5.6 Interference

The asymmetric values displayed in the graph in figure 22, shows the noise added during the transmission from the CG to the DSP of the MTU. This transmission is carried out via the mobile network and the Bluetooth prototype link in the MTU. This noise is presented as interference in the audio picture, and most of the interference is rather easily filtered out using a digital filter [25] and noise reduction tools.

Figure 23 shows the same sentence as in figure 22, but after it has been processed with a noise reduction function and a fast fourier transformation filter. However, some parts of the interference are impossible to filter out. These parts consistently appear at the same position in the sentence and have a negative effect on the PESQ score. Some parts of the interference may be introduced via the unshielded wires extending from the F2M03 module to the DSP, whereas some audio packets may have been corrupted in the air transmission between the two Bluetooth entities. The persistent interferences, which continuously occur in certain positions in the sentences, indicate that some type of audio transformation is affecting the PESQ score in a negative way.

Figure 23 Symmetric and asymmetric values after filter and noise processing.

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5.7 Using USB Bluetooth devices

To remove any interference that may have been introduced by the unshielded wires in the prototype, two USB Bluetooth entities were purchased and installed on two computers. A SCO connection was established between the entities, and sentences similar to the CG and MTU correlation of 5.5 second transmissions were sent. However, the time slots between the transmissions were longer than for the CG and MTU correlation.

The tests using the USB Bluetooth devices followed the same pattern as the prototype. The parts that had repeatedly been corrupted in tests with the prototype were also corrupted in the tests with the USB devices. These results show that the Bluetooth transmission has a harmful influence on the PESQ score. Figure 24 illustrates the symmetric and asymmetric PESQ-tester values of a transmission between two USB Bluetooth devices. Figure 25 shows the symmetric and asymmetric values after filter and noise processing are performed on the transmitted sentence.

Figure 24 Asymmetric and symmetric values with a USB Bluetooth transmission.

Figure 25 Symmetric and asymmetric values after filter and noise processing.

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The waveforms of the corrupted parts in the transmitted and recorded sentences were analysed. The sample intervals surrounding 42 and 272 produced high values on the asymmetric vector in every test. Both of these sample intervals show the same pattern of corruption. The waveform was flat in the corrupted parts, which seems to generate bad output MOS_LQO from the PESQ algorithm. The human ear is unable to distinguish these flat sections on the waveform, as they represent only a small fraction of the samples that constitute one second. Figure 26 exhibits the aforementioned flat parts in the corrupted sentence.

Figure 26 Transmitted corrupted waveform.

5.8 Transmission studies

The parts of the transmitted sentences that resulted in large values on the asymmetric vector were analysed in depth. The focal point of this analysis was to discover the elements responsible for transforming the audio stream in Bluetooth. This information was valuable in order to learn the reason why negative values on the asymmetric vector always appeared on the same sample interval every time a test has been performed.

After a closer examination of the radio layer in the Bluetooth stack, the source of the audio impairment was located. The Bluetooth entities in the prototype, as well as the PCM-interface format that exists between them, all use a 16 bit linear PCM format. This format is managed by the CVSD codec, which performs encoding and decoding in the linear PCM [7], as described in chapter three.

This CVSD codec does however have certain known limitations. The CVSD algorithm is constructed based on the assumption that a voice signal does not change abruptly. Therefore, if the slope of input voice stream changes too fast in a way that the CVSD algorithm is unequipped for, the results produced by the CVSD codec will be unreliable. Unsuccessful CVSD transformation may introduce quantization noise into the Bluetooth link, which will affect the PESQ score negatively. [7]

Figure 27 shows how the CVSD quantization noise may occur in the transformation from linear PCM to the CVSD data stream.

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Figure 27 CVSD quantization noise.

Figure 28 shows a change in the average amplitude values of the sentences. This represents how the output from the PESQ algorithm has changed according to the different average Root Mean Square (RMS) powers.

2.82.9

33.13.23.33.43.53.63.73.83.9

-11 -13 -15 -17 -19 -21 -23 -25 -27 -29

dB

MOS_LQO

Figure 28 Relationship between the amplitude and MOS_LQO score.

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6 Conclusion

The Bluetooth audio link created using the prototype, as well as the Bluetooth connection set up between the two PCs used for this Master’s Thesis, both affected the audio recorded in the MTU negatively to the extent that the audio could not be used for AQM with PESQ.

The TA system and especially the MTU are designated to measure the audio quality in the mobile network. If an audio link within the MTU is of a poor enough quality that measurement results become unsatisfactory, then the MTU will focus on solving other tasks than it is intended for. Instead of producing a measurement of the quality of the mobile network, the MTU will instead provide a score reflecting the quality of the mobile network merged with the Bluetooth link. This is not a task that the MTU is intended for today.

6.1 Further work

This prototype only used the linear PCM interface in the Bluetooth SCO-link, which means that the data is transformed by the CVSD codec, and the data is never retransmitted. A task for developers in the future can be to study whether it is possible to redirect the audio stream and send it to the ACL link [21], as the ACL link does not transform the audio and is able to retransmit audio packets. However, the ACL link will introduce delays in- and between the sentences, and as a result, retransmissions will occur when packets are lost or corrupted upon arrival at the receiver. The PESQ algorithm will hopefully be more tolerant to such delays, which is not entirely unlikely considering its toleration of handovers and VoIP.

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7 Acknowledgement

First of all I would like to thank my external supervisor Ulf Marklund at Ericsson TEMS for his guidance and support during the work on this project. I would also like to direct my great appreciation to Per Johansson at Ericsson TEMS, who has provided me with numerous valuable ideas and suggestions. My internal supervisor, Jerry Eriksson, deserves a special mention for answering my questions about thesis formalities. Lastly, I would also like to thank all of the employees at Ericsson TEMS who have been more or less involved in my project, for making this period a very enjoyable time.

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8 Terminology

ACL Asynchronous Connection-Less

AQM Audio quality measurement

CCD Customer Care Department

CG Call generator, one of the processes in the TA system

CVSD Continuous Variable Slope Delta

DSP Digital Signal Processor

HTU Handheld Test Unit

ISM Industrial Scientific Medicine

L2CAP Logical Link Control and Adaptation Protocol

LMP Link Manger Protocol

LQO Listener Quality Option

MOS Mean Option Score

MTU Mobile Test Unit

PCM Pulse Code Modulation

PESQ Perceptual Evaluation of Speech Quality

PSQM Perceptual Speech Quality measurement

PSTN Public Switch Telephone Network

QoS Quality of Service

RFCOMM Radio Frequency Communication

SCO Synchronous Connection-Oriented

TLH TEMS Logfile Handler

VoIP Voice over Internet Protocol

VQM Voice Quality Measurement

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9 References

[1] Acoustic Parameters Ericsson Mobile Platform E100, G200, U100 Description (internal document)

[2] Audio path Ericsson Mobile Platform E100, G200, U100 Description (internal document)

[3] Audio Quality Measurement in TEMS Automatic –PESQ

[4] Audio Quality Measurement in TEMS Automatic –PSQM White Paper

[5] Att förstå Tele Kommunikation Ericsson Telecom, Telia AB ISBN 91-44-37801-7, pp 70-76

[6] Class 2 BluetoothTM Module – F2M03AC2 Datasheet Rev:10 January 2005

[7] Continuously Variable Slope Delta Modulation: A tutorial. Web site 1 Sep 2005 http://www.cmlmicro.com

[8] Design Specification TEMS Automatic System (internal document)

[9] Host Hands-Free Message Interface Free2Move Rev:08 April 2005

[10] ITU-T P.862.1 Mapping function for transforming of P862 to MOS-LQO

[11] ITU-T P.800 Mean Option Score

[12] ITU-T P.862 Perceptual evaluation of speech quality (PESQ), and objective method for end-to-end speech quality assessment of narrow-band telephone networks and speech codecs

[13] ITU-T G.711 Pulse Code Modulation (PCM) of voice frequencies.

[14] Bluetooth: Carrying Voice over ACL Links Rohit Kapoor, Ling-Jyh Chen, Yeng-Zhong Lee, Mario Gerla. 3803 H, Boelter Hall, University of California, Los Angeles

[15] Modern telekommunikation Gunnar Karlsson ISBN 91-44-00118-5

[16] Bluetooth Revealed Brent A. Miller, Chatschik Bisdikian ISBN 0.13-090294-2

[17] Bluetooth Demystified Nathan J. Muller ISBN 0-07-136323-8, pp 23, 59, 70-71, 101, 104-106

[18] Mobile Communications Second Edition Jochen Schiller ISBN 0-321-12381-6, pp. 269-293

[19] Specification for mixed digital –analog ASIC (internal document)

[20] Specification of the Bluetooth System Wireless connections made easy Core Version 1.1 February 22, 2001

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[21] Specification of the Bluetooth System Wireless connections made easy Core Version 1.1 February 22, 2001 Part A

[22] Specification of the Bluetooth System Wireless connections made easy Core Version 1.1 February 22, 2001 Part B pp. 45-77 ,81, 85-91, 139-142

[23] TEMS Automatic UMTS MTU700 Configuration Guide

[24] USER MANUAL Free2Move Evaluation Board. Rev:24 February 2005

[25] Digital Filters Lars Wanhammar and Håkan Johansson Department of Electrical Enginering Lindköpings universitet 2002 pp. 13-18

[26] Waveform Coding Techniques http://www.cisco.com/warp/public/788/signalling/waveform_coding.html#topic1, 26 September 2005