<p>Communication tools such as text-based instant messaging, voice and video relay services, real-time video chat and mobile SMS and MMS have successfully been used among Deaf people. Several years of field research with a local Deaf community revealed that disadvantaged South African Deaf&nbsp / people preferred to communicate with both Deaf and hearing peers in South African Sign Language as opposed to text. Synchronous video chat and video&nbsp / relay services provided such opportunities. Both types of services are commonly available in developed regions, but not in developing countries like South&nbsp / Africa. This thesis reports on a workaround approach to design and develop an asynchronous video communication tool that adapted synchronous video&nbsp / &nbsp / codecs to store-and-forward video delivery. This novel asynchronous video tool provided high quality South African Sign Language video chat at the&nbsp / expense of some additional latency. Synchronous video codec adaptation consisted of comparing codecs, and choosing one to optimise in order to&nbsp / minimise latency and preserve video quality. Traditional quality of service metrics only addressed real-time video quality and related services. There was no&nbsp / uch standard for asynchronous video communication. Therefore, we also enhanced traditional objective video quality metrics with subjective&nbsp / assessment metrics conducted with the local Deaf community.</p>

Voice over Internet Protocol (VoIP) provides a new telephony approach where the voice traffic passes over Internet Protocol shared traffic networks. VoIP is a significant application of the converged network principle. The research aim is to model VoIP over a hybrid Universal Mobile Telecommunications System (UMTS) network and to identify an improved approach to applying the ITU-T Recommendation G.107 (E-Model) to understand possible Quality of Service (QoS) outcomes for the hybrid UMTS network. This research included Modeling the hybrid UMTS network and carrying out simulations of different traffic types transmitted over the network. The traffic characteristics were analysed and compared with results from the literature. VoIP traffic was modelled over the hybrid UMTS network and the VoIP traffic was generated to represent different loads on the network from light to medium and heavy VoIP traffic. The VoIP over hybrid UMTS network traffic results were characterized and used in conjunction with the E-Model to identify VoIP QoS outcomes. The E-Model technique was implemented and results achieved were compared with results for other network types highlighted in the literature. The research identified an approach that permits accurate Modeling of VoIP QoS over a hybrid UMTS network. Accurate results should allow network design to facilitate new approaches to achieving an optimal network implementation for VoIP.

En este trabajo de título se diseña y elabora un programa para ejecutarse en un teléfono celular, que permite junto con simular la trama completa de una llamada VoIP, evaluar distintos tipos de reconstrucciones para la voz en caso de sufrir pérdida de paquetes IP. Hoy en día, la penetración que tienen los teléfonos celulares y las redes de datos como Internet, han dado paso a un sinnúmero de posibilidades y aplicaciones que hacen uso combinado de estas tecnologías. El envío de audio utilizando dispositivos móviles y paquetes IP, representa todo un desafío pues es considerado una tecnología emergente y se espera que en el futuro tenga un gran impacto dada la transición a la tecnología 3G que recién se está iniciando en Chile. La transmisión de datos sobre Internet sufre de pérdidas de paquetes, lo que significa que hay un porcentaje de los elementos de comunicación que no llega a su destino. El envío de voz y audio no está excento de este problema y la documentación sobre cómo solucionar este problema en dispositivos móviles aún es pobre. Se estudió a fondo el comportamiento de la voz tal como se procesa en una llamada telefónica, utilizando el mismo codec AMR. Con esto se diseñó una aplicación de simulación para toda la trama que sufre una onda de audio, desde que es emitida por una persona hasta que llega a su receptor, pasando justamente por un simulador de pérdidas de paquetes. Finalmente se diseñaron y evaluaron distintos tipo de reconstrucciones para suplir las pérdidas sufridas. En base a las evaluaciones realizadas se obtuvo un prototipo de codec que soporta la pérdida de paquetes de calidad aceptable cuando éstas no superan el 30%. Además la aplicación lograda permite la implementación y evaluación de nuevos prototipos. Se proponen mejoras para esta aplicación con respecto a la optimización de estructuras de datos y memoria, así como también posibles integraciones con otras aplicaciones ya existentes.

<p>As part of an ongoing project at the department of electrical engineering, ISY, at Linköping University, a voice decoder using floating point formats has been the focus of this master thesis. Previous work has been done developing an mp3-decoder using the floating point formats. All is expected to be implemented on a single DSP.The ever present desire to make things smaller, more efficient and less power consuming are the main reasons for this master thesis regarding the use of a floating point format instead of the traditional integer format in a GSM codec. The idea with the low precision floating point format is to be able to reduce the size of the memory. This in turn reduces the size of the total chip area needed and also decreases the power consumption.One main question is if this can be done with the floating point format without losing too much sound quality of the speech. When using the integer format, one can represent every value in the range depending on how many bits are being used. When using a floating point format you can represent larger values using fewer bits compared to the integer format but you lose representation of some values and have to round the values off.From the tests that have been made with the decoder during this thesis, it has been found that the audible difference between the two formats is very small and can hardly be heard, if at all. The rounding seems to have very little effect on the quality of the sound and the implementation of the codec has succeeded in reproducing similar sound quality to the GSM standard decoder.</p>

As part of an ongoing project at the department of electrical engineering, ISY, at Linköping University, a voice decoder using floating point formats has been the focus of this master thesis. Previous work has been done developing an mp3-decoder using the floating point formats. All is expected to be implemented on a single DSP.The ever present desire to make things smaller, more efficient and less power consuming are the main reasons for this master thesis regarding the use of a floating point format instead of the traditional integer format in a GSM codec. The idea with the low precision floating point format is to be able to reduce the size of the memory. This in turn reduces the size of the total chip area needed and also decreases the power consumption.One main question is if this can be done with the floating point format without losing too much sound quality of the speech. When using the integer format, one can represent every value in the range depending on how many bits are being used. When using a floating point format you can represent larger values using fewer bits compared to the integer format but you lose representation of some values and have to round the values off.From the tests that have been made with the decoder during this thesis, it has been found that the audible difference between the two formats is very small and can hardly be heard, if at all. The rounding seems to have very little effect on the quality of the sound and the implementation of the codec has succeeded in reproducing similar sound quality to the GSM standard decoder.

Magister Scientiae - MSc / Communication tools such as text-based instant messaging, voice and video relay services, real-time video chat and mobile SMS and MMS have successfully been used among Deaf people. Several years of field research with a local Deaf community revealed that disadvantaged South African Deaf people preferred to communicate with both Deaf and hearing peers in South African Sign Language as opposed to text. Synchronous video chat and video relay services provided such opportunities. Both types of services are commonly available in developed regions, but not in developing countries like South Africa. This thesis reports on a workaround approach to design and develop an asynchronous video communication tool that adapted synchronous video codecs to store-and-forward video delivery. This novel asynchronous video tool provided high quality South African Sign Language video chat at the expense of some additional latency. Synchronous video codec adaptation consisted of comparing codecs, and choosing one to optimise in order to minimise latency and preserve video quality. Traditional quality of service metrics only addressed real-time video quality and related services. There was no such standard for asynchronous video communication. Therefore, we also enhanced traditional objective video quality metrics with subjective assessment metrics conducted with the local Deaf community. / South Africa

L’algorithme de basculement de bits à descente de gradient probabiliste (Probabilistic Gradient Descent Bit Flipping :PGDBF) est récemment introduit comme un nouveau type de décodeur de décision forte pour le code de contrôle de parité à faible densité (Low Density Parity Check : LDPC) appliqué au canal symétrique binaire. En suivant précisément les étapes de décodage du décodeur déterministe Gradient Descent Bit-Flipping (GDBF), le PGDBF intègre en plus la perturbation aléatoire dans l'opération de basculement des Nœuds de Variables (VNs) et produit ainsi une performance de décodage exceptionnelle qui est meilleure que tous les décodeurs à basculement des bits (BF : Bit Flipping) connus dans la littérature, et qui approche les performances du décodeur de décision souple. Nous proposons dans cette thèse plusieurs implémentations matérielles du PGDBF, ainsi qu'une analyse théorique de sa capacité de correction d'erreurs. Avec une analyse de chaîne de Markov du décodeur, nous montrons qu’en raison de l'incorporation de la perturbation aléatoire dans le traitement des VNs, le PGDBF s'échappe des états de piégeage qui empêchent sa convergence. De plus, avec la nouvelle méthode d'analyse proposée, la performance du PGDBF peut être prédite et formulée par une équation de taux de trames erronées en fonction du nombre des itérations, pour un motif d'erreur donné. L'analyse fournit également des explications claires sur plusieurs phénomènes de PGDBF tels que le gain de re-décodage (ou de redémarrage) sur un motif d'erreur reçu. La problématique de l’implémentation matérielle du PGDBF est également abordée dans cette thèse. L’implémentation classique du décodeur PGDBF, dans laquelle un générateur de signal probabiliste est ajouté au-dessus du GDBF, est introduite avec une augmentation inévitable de la complexité du décodeur. Plusieurs procédés de génération de signaux probabilistes sont introduits pour minimiser le surcoût matériel du PGDBF. Ces méthodes sont motivées par l'analyse statistique qui révèle les caractéristiques critiques de la séquence aléatoire binaire requise pour obtenir une bonne performance de décodage et suggérer les directions possibles de simplification. Les résultats de synthèse montrent que le PGDBF déployé avec notre méthode de génération des signaux aléatoires n’a besoin qu’une très faible complexité supplémentaire par rapport au GDBF tout en gardant les mêmes performances qu’un décodeur PGDBF théorique. Une implémentation matérielle intéressante et particulière du PGDBF sur les codes LDPC quasicyclique (QC-LPDC) est proposée dans la dernière partie de la thèse. En exploitant la structure du QCLPDC, une nouvelle architecture pour implémenter le PGDBF est proposée sous le nom d'architecture à décalage des Nœuds de Variables (VNSA : Variable-Node Shift Architecture). En implémentant le PGDBF par VNSA, nous montrons que la complexité matérielle du décodeur est même inférieure à celle du GDBF déterministe tout en préservant la performance de décodage aussi élevée que celle fournie par un PGDBF théorique. Enfin, nous montrons la capacité de cette architecture VNSA à se généraliser sur d'autres types d'algorithmes de décodage LDPC. / Probabilistic Gradient Descent Bit Flipping (PGDBF) algorithm have been recently introduced as a new type of hard decision decoder for Low-Density Parity-Check Code (LDPC) applied on the Binary Symmetric Channel. By following precisely the decoding steps of the deterministic Gradient Descent Bit-Flipping (GDBF) decoder, PGDBF additionally incorporates a random perturbation in the ipping operation of Variable Nodes (VNs) and produces an outstanding decoding performance which is better to all known Bit Flipping decoders, approaching the performance of soft decision decoders. We propose in this thesis several hardware implementations of PGDBF, together with a theoretical analysis of its error correction capability. With a Markov Chain analysis of the decoder, we show that, due to the incorporation of random perturbation in VN processing, the PGDBF escapes from the trapping states which prevent the convergence of decoder. Also, with the new proposed analysis method, the PGDBF performance can be predicted and formulated by a Frame Error Rate equation as a function of the iteration, for a given error pattern. The analysis also gives a clear explanation on several phenomenons of PGDBF such as the gain of re-decoding (or restarting) on a received error pattern. The implementation issue of PGDBF is addressed in this thesis. The conventional implementation of PGDBF, in which a probabilistic signal generator is added on top of the GDBF, is shown with an inevitable increase in hardware complexity. Several methods for generating the probabilistic signals are introduced which minimize the overhead complexity of PGDBF. These methods are motivated by the statistical analysis which reveals the critical features of the binary random sequence required to get good decoding performance and suggesting the simpli cation directions. The synthesis results show that the implemented PGDBF with the proposed probabilistic signal generator method requires a negligible extra complexity with the equivalent decoding performance to the theoretical PGDBF. An interesting and particular implementation of PGDBF for the Quasi-Cyclic LPDC (QC-LPDC) is shown in the last part of the thesis. Exploiting the structure of QC-LPDC, a novel architecture to implement PGDBF is proposed called Variable-Node Shift Architecture (VNSA). By implementing PGDBF by VNSA, it is shown that the decoder complexity is even smaller than the deterministic GDBF while preserving the decoding performance as good as the theoretical PGDBF. Furthermore, VNSA is also shown to be able to apply on other types of LDPC decoding algorithms.

<p>The thesis project consists on developing an interface for a Nios II processor integrated in a board of Altera (UP3- 2C35F672C6 Cyclone II).</p><p>The main goal is show how the Nios II processor can interact with the other components of the board.The Quartus II software has been used to create to vhdl code of the interfaces, compile it and download it into the board. The Nios II IDE tool is used to build the C/C++ files and download them into the processor.</p><p>It has been prepared an application for the audio codec integrated in the board (Wolfson WM8731 24-bit sigma-delta audio CODEC). The line input of the audio codec receives an analog signal from a laptop, this signal is managed by the control interface of the audio codec. The converters ADCs and DACs are stereo 24-bit sigma delta and they are used with oversampling digital interpolation and decimation filters.</p><p>The digital interface of the audio codec sends the digital signal to the Nios II processor and receives the data from the processor. After building the interfaces for the audio codec and the processor, it has been prepared an application in C++ language for the processor that modifies the volume of the signal.</p><p>The signal come back to the audio codec and it is possible to check the results with headphones or speakers at the line output of the audio codec.</p>

<p>Voice over Internet Protocol (VoIP) has become very popular in recent days andbecome the first choice of small to medium companies for voice and data integration inorder to cut down the cost and use the IT resources in much more efficient way. Anotherpopular technology that is ruling the world after the year 2000 is 802.11 wirelessnetworks. The Organization wants to implement the VoIP on the wireless network. Thewireless medium has different nature and requirement than the 802.3 (Ethernet) andspecial consideration take into account while implementing the VoIP over wirelessnetwork.One of the major differences between 802.11 and 802.3 is the bandwidthavailability. When we implement the VoIP over 802.11, we must use the availablebandwidth is an efficient way that the VoIP application use as less bandwidth as possiblewhile retaining the good voice quality. In our project, we evaluated the differentcompression and decompression (CODEC) schemes over the wireless network for VoIP.To conduct this test we used two computers for comparing and evaluatingperformance between different CODEC. One dedicated system is used as Asterisk server,which is open source PBX software that is ready to use for main stream VoIPimplementation. Our main focus was on the end-to-end delay, jitter and packet loss forVoIP transmission for different CODECs under the different circumstances in thewireless network. The study also analyzed the VoIP codec selection based on the MeanOpinion Score (MOS) delivered by the softphone. In the end, we made a comparisonbetween all the proposed CODECs based on all the results and suggested the one Codecthat performs well in wireless network.</p>

<p>Communication tools such as text-based instant messaging, voice and video relay services, real-time video chat and mobile SMS and MMS have successfully been used among Deaf people.&nbsp / Several years of field research with a local Deaf community revealed that disadvantaged South African Deaf people preferred to communicate with both Deaf and hearing peers in South African&nbsp / Sign Language as opposed to text. Synchronous video chat and video relay services provided such opportunities. Both types of services are commonly available in developed regions, but not in&nbsp / developing countries like South Africa. This thesis reports on a workaround approach to design and develop an asynchronous video communication tool that adapted synchronous video codecs&nbsp / to store-and-forward video delivery. This novel asynchronous video tool provided high quality South African Sign Language video chat at the expense of some additional latency. Synchronous video&nbsp / codec adaptation consisted of comparing codecs, and choosing one to optimise in order to minimise latency and preserve video quality. Traditional quality of service metrics only addressed real-time video quality and related services. There was no such standard for asynchronous video communication. Therefore, we also enhanced traditional objective video quality&nbsp / metrics with subjective assessment metrics conducted with the local Deaf community.&nbsp / </p>