The thesis project consists on developing an interface for a Nios II processor integrated in a board of Altera (UP3- 2C35F672C6 Cyclone II). 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. 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. 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. 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>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>

The past three decades has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for voice applications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. Then I proceed with descriptions of waveform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. At the end the system for computer-based stenographing is presented. Quality research and ways how to improve this system will be provided.

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.

The Voice over Internet Protocol (VoIP) is on its way to surpassing toll quality. Although VoIP shares its transmission channel with other communication traffic, today internet has a wider bandwidth than the legacy Digital Loop Carrier and voice could be digitized higher than traditional 8 kbps, to say 16 kbps. Thus, VoIP should not be limited by the toll quality. However, VoIP quality could go down, as a result of unpredictable traffic congestion and network imperfections. These two situations cause delay jitter and packet loss of VoIP. To overcome these challenges, there are ongoing works for service providers including but not limited to optimizing routing and adding more bandwidth. There are also works by developers at the user’s end, which includes compressing voice packet size and processing playout delay adapted to the network condition. While VoIP planning or off-line quality monitoring and control use overall quality measurements such as mean opinion score (MOS) or R-factor, the real-time quality supervision typically uses the network condition factors only. The control mechanism that is based on network quality could adjust the channel parameter by changing Codec and its parameters, and changing playout delay, etc. to minimize the loss of voice quality. As bandwidth plays a prominent role in IP traffic congestion, compressing the packet header is a possible solution to minimize congestion. Replacing a completed packet header with a smaller header will significantly reduce the packet header size. For instance, with a context, a compressed header will not consist of RTP header and, thus, could reduce 16 bytes from each packet. However, the primary question is how to deal with delay jitter calculation without time stamping. In this research, a delay jitter calculation for VoIP packet without timestamp has been provided. Compressing payload or using high compressing Codecs, is another major solution for preventing quality downgrade with limited bandwidth. The challenge with many Codec and the tradeoff between Codec quality and packet loss due to limited bandwidth has been addressed in this research with a summary of Codec quality evaluation and a bandwidth planning calculation. Although the E-model and its R-factor has been proposed by the International Telecommunication Union (ITU) for VoIP quality measurement, with many network and Codec parameters, it could only be used for offline quality control. Since accessing a live traffic for monitoring live quality is somewhat impossible, at the client side, only packet loss and delay jitter matters. In this research, more in-depth investigation of adaptive playout delay based on jitter prediction has been carried out and recommended as the end user solution for quality improvement. An adaptive playout delay based on Markov model also has been developed in detail and tested with real VoIP network. This development has closed the gap between research and engineering. Therefore, the Markov model could be evaluated and implemented.

Context-awareness is a hot topic in mobile computing currently. A lot of importance is being attached to facilitating the user of various mobile computing devices to provide services that are more “user-centric”. One aspect of context-awareness is to perceive variations in available resources, and to make decisions based on the feedback to enable applications to automatically adapt to the current environment. For Voice over IP (VoIP) software phones (softphones), variations in network performance lead to fluctuations in the quality of the communication. Therefore, by making these softphones more adaptive to the network environment will, to some extent, mask such fluctuations. Dynamic voice and video adaptation derives from the fact that different coder-decoders (CODEC) have different characteristics, even the same CODECs with a different configuration can behave quite differently, in terms of bandwidth consumption, packet size, etc. Minisip is a VoIP client application which was implemented on and targeted for a Linux platform. One of my tasks was to port Minisip to Microsoft’s Windows Mobile operating system, running on an HP IPAQ Pocket PC H5550. Such handheld computer enables the user to communication while they are moving about, thus increasing the probability that the characteristics of the network connection will change. Building upon this port, the next task was to add dynamic voice and video CODEC adaptation. Dynamic voice and video CODEC adaptation on Minisip poses several challenges, for example, in what way can the network performance be determined and what adaptation strategy can achieve high call quality while making efficient utilization of available network resources. In order to make the proper design choices, several estimation models will be discussed, these are used to determine an efficient, un-intrusive, and light weight means of dynamic CODEC selection within Minisip. This thesis only implemented audio CODEC adaptation of Minisip, and the evaluation of the resulting prototype shows that such dynamic adaptation is both feasible and practical; further more, video CODEC adaptation would be a more significant extension to this work in the future. / Context-awareness är ett hett i den nuvarande mobila datavärlden. Det finns ett stort värde i att facilitating användare av olika mobila dator anordningar för att kunna förse branschen med användarvänligare tjänster. En aspekt på Context-awareness är att uppmärksamma variationen i de tillgängliga medel som finns tillhanda, och att ta beslut som är baserade på feedback för att applikationen automatiskt ska anpassa sig till den nuvarande miljön. Variationer i nätverksprestanda påverkar kvaliteten på Voice over IP (VoIP), som är en typ utav softwaretelefon, i hög grad. Dessa kvalitets svängningar kan stabiliseras och döljas i högre grad om softwaretelefonen anpassas till nätverksmiljön. Dynamisk voice och video adaptation härleds från faktum att olika coder-decoders (CODEC) har olika karaktärer, även samma CODEC med en annan konfiguration kan bete sig olikt sig själv om vi talar om bandbredds förbrukning och packet storlekar, etc. Minisip är en VoIP klient som är framtagen för Linux plattformen. En av mina huvuduppgifter var att port Minisip till Microsoft’s Windows Mobila operativsystem genom att köra en HP IPAQ Pocket PC H5550. En sådan bärbar dator möjliggör för användaren att kommunicera fastän denne rör på sig, fastän risken finns för att nätverks kontakten ändras. Baserat på denna port, blev min nästa uppgift att anpassa denna CODEC till dynamiskt ljud och bild. Att anpassa denna CODEC till dynamiskt ljud och bild på Minisip medför många utmaningar t.ex. hur nätverks prestandan kan bestämmas och vilken anpassningsstrategi som kan bidra till högkvalitativa samtal samtidigt som nätverks tillgångarna nyttjas på ett effektivt sätt. Denna tes kan endast genomföras på ljud CODEC anpassning av Minisip, och utvärderingen utav prototypen resulterade i att sådan dynamisk anpassning är både genomförbar och praktisk, en video CODEC anpassning skulle bli ett perfekt uppföljningsprojekt till denna studie.

A Tablet PC is a portable computing device which combines a regular notebook computer with a digitizing screen that interacts with a complementary electronic pen stylus. The pen allows the user to input data by writing on or by tapping the screen. Like a regular notebook computer, the user can also perform tasks using the mouse and keyboard. A Tablet PC gives the users all the features of a regular notebook computer along with the support to recognize, process, and store electronic/digital ink, enabling a user to make and save hand-written notes or data. In institutions of teaching and learning, instructors often use computer-based materials like web pages, PowerPoint® slides, etc., to explain subject matter. The ability to annotate on presentation information using the electronic stylus of a Tablet PC has attracted the attention of the academic community to use the Tablet PC as a potential tool for increasing the effectiveness of presentations in teaching and learning. Tablet PC-based applications such as OneNote®, WindowsJournal® and Classroom Presenter have been developed to enhance note-taking in classrooms based on the fact that a pen stylus is a more natural form of input device for making notes on the computer as compared to the regular keyboard and mouse. Although tools like OneNote®, WindowsJournal® enhanced the note-taking process on the Tablet PC, they lacked the ability to allow the user to directly annotate on the lecture content. Classroom Presenter provides the ability to integrate classroom notes and the presentation material by allowing the instructors and students to annotate over the lecture material. However, all the above tools lacked the ability to allow a user to take notes over the output window of an arbitrary application like Excel, an active simulator or a movies players output. The Tablet PC based tool, WriteOn, developed at Virginia Tech, addresses this drawback. WriteOn, when deployed on the Tablet PC in a classroom environment, allows the instructor to utilize electronic ink to annotate on top of any application window visible on the Tablet PC display screen, including those that play active content like a movie or simulation. WriteOn facilitates a user to annotate over a dynamic application window by activating its virtual transparency surface called the eVellum (electronic vellum). The user can view a movie or an active simulation running in the eVellum background because of its transparent color. The user can deactivate the eVellum to make it invisible by "piercing" it if he/she wishes to access the desktop or an application window under the vellum window. WriteOn provides the instructor with the ability to broadcast a composite of the dynamic lecture content and ink annotations to the students in real-time. The term dynamic lecture contents is meant to indicate that the content being annotated need not be static words on a background, but may also be window contents that are changing in time. Using WriteOn, the students can make their own notes by writing on the eVellum enabled on top of the lecture stream window without losing visibility of the lecture. The instructor/student can save the ink annotations along with base lecture material as a movie file. The ability of WriteOn to improve classroom presentation and student note-taking as shown by initial tests, were pedagogically very useful. However, in order to deploy WriteOn on large scale in classrooms as an active and effective teaching tool of choice, several aspects of the application had to be improved. One aspect of the application that needed improvement was the user interface. The primitive Graphical User Interface (GUI) of the WriteOn tool was not easily usable by instructors and students from non-computer science backgrounds. The second aspect needing improvement was the operational performance of the application in terms of its CPU resource utilization. The WriteOn tool has shown to have operational performance issues during the screen capture process. This research therefore aims to address improvements in the GUI to make it more user friendly and increase the operational performance to the point where the user does not notice degradation of a base lecture application. Incorporation of these improvements has led us to rename the application as WriteOn1.0. WriteOn1.0 implements a picture-based GUI that comprises of two forms: a main form that appears shortly after WriteOn1.0 starts and a toolbar. The WriteOn1.0 toolbar appears in the center of the top edge of the display as soon as the user initiates a task like a screen recording session, by clicking on the appropriate menu button on the main form. The toolbar provides the user, accessibility to perform all the desired activities like annotating, screen recording, presentation broadcast, and piercing of the eVellum by a single-click of the appropriate menu icon. Tool tips that appear when the user points the mouse over a picture icon on the toolbar, explain the task that shall be performed when he/she clicks on the underlying menu icon. WriteOn1.0 introduces a window-like resizable and movable eVellum called the scalable eVellum that it activates in the area of interest specified by the user. Unlike the first implementation of the eVellum which had a fixed location and spanned the entirety of the user's desktop window, the instructor/student define the dimensions of the scalable eVellum and can choose to re-dimension, relocate and pierce through it at any point of time during a session. WriteOn1.0 also introduces the transparent mode of operation wherein the instructor/student, without having to deactivate the scalable eVellum can access any underlying window by a right-click of the mouse on the eVellum surface while the ink annotations are intact on the foreground. WriteOn1.0 addresses the operational performance issues observed during a screen capture session in WriteOn by capturing the activities only in the area of interest of the user for recording and broadcasting. By combining this scheme with a with a lossless screen capture codec called the MSU screen capture codec that has a high-compression ratio and that is optimized for speed for data compression, WriteOn1.0 greatly improves the operational CPU performance of the tool. WriteOn1.0 employs various technologies to implement its features. The improvements to operational performance are implemented by using the MSU screen codec from Moscow State University's Graphics and Media Lab. Microsoft®'s Video for Windows Framework (VfW) and WindowsMedia Player API's are used to realize the module that records the screen activities to an AVI file while DirectShow of DirectX and ConferenceXP API's are used for streaming presentations over a network. WriteOn1.0, with its features like its scalable eVellum, good operational performance and picture-based GUI is aimed at potentially making it a teaching tool of choice across classrooms and changing the method of classroom instruction of courses involving dynamic content. / Master of Science

Submitted by Silvana Teresinha Dornelles Studzinski (sstudzinski) on 2015-07-03T17:52:39Z No. of bitstreams: 1 Iris Corrêa das Chagas Linck.pdf: 1456080 bytes, checksum: ea4a6f659a229e845649c58baaf8cb23 (MD5) / Made available in DSpace on 2015-07-03T17:52:39Z (GMT). No. of bitstreams: 1 Iris Corrêa das Chagas Linck.pdf: 1456080 bytes, checksum: ea4a6f659a229e845649c58baaf8cb23 (MD5) Previous issue date: 2012 / FINEP - Financiadora de Estudos e Projetos / Neste trabalho é desenvolvido um algoritmo híbrido que simula o comportamento do Codificador/Decodificador de vídeo H.264/AVC, ou simplesmente CODEC H.264, utilizado no Sistema Brasileiro de Televisão Digital. O algoritmo proposto tem a finalidade de buscar a melhor configuração possível de seis dos principais parâmetros utilizados para a configuração do CODEC H.264. Este problema é abordado como um problema de otimização combinatória conhecido como Problema de Seleção de Partes e que é classificado como NP-Difícil. O algoritmo híbrido proposto, denominado Simulador de Metaheurísticas aplicado a um CODEC (SMC), foi desenvolvido com base em duas metaheurísticas: Busca Tabu e Algoritmo Genético. Os seis parâmetros de configuração a serem otimizados pelo SMC são: o bit rate; o frame rate; os parâmetros de quantização de quadros tipo B, tipo P e tipo I e a quantidade de quadros tipo B em um grupo de imagens (GOP – Group of Pictures). Os dois primeiros parâmetros mencionados atuam basicamente sobre a qualidade da imagem do vídeo enquanto que os demais parâmetros atuam diretamente na compressão do vídeo. Experimentos e testes foram feitos utilizandose o CODEC H.264 desenvolvido no Projeto Plataforma de Convergência Digital IPTV/TV Digital (DigConv). Nos experimentos o CODEC tem seus parâmetros configurados de acordo com os resultados obtidos pelo SMC. Um vídeo é codificado no CODEC H.264 para que se possa analisar a sua qualidade de imagem e o seu grau de compressão após o processo de codificação. É feita uma correlação entre esses resultados e a Função Objetivo do SMC. A qualidade da imagem é medida através da métrica mais utilizada na literatura, o PSNR (Peak Signal to Noise Ratio), que é calculada pelo próprio CODEC ao final da codificação de um vídeo. Verificouse que à medida que a Função Objetivo aumenta, o CODEC H.264 consegue obter uma melhor qualidade de imagem e um maior grau de compressão de vídeo. / In this work is developed a hybrid algorithm that simulates the behavior of the H.264/AVC video encoder/decoder, or simply H.264 video CODEC, used in the Brazilian System of Digital Television. The proposed algorithm intends to seek the best possible configuration of the six main parameters used for configuring the H.264 video CODEC. This problem is treated as a combinatorial optimization problem known as the Parties Selection Problem, which is classified as NP-Hard. The proposed hybrid algorithm, called Simulator Metaheuristcs applied to a CODEC (SMC), was developed based on two metaheuristics: Tabu Search and Genetic Algorithm. The six configuration parameters to be optimized by the SMC are the bit rate, frame rate, the parameters of quantization tables of type B, type I and type P and the amount of frames type B in a group of pictures (GOP - Group of Pictures).The first two parameters mentioned, work primarily on the quality of the video image while the other parameters act directly on the video compression. Experiments and tests were done using the video CODEC H.264 developed in Digital Convergence Platform IPTV/Digital TV Project (DigConv). DigConv Project. In the experiments the CODEC has its parameters set according to the results obtained by the SMC. Then, a video is encoded by the CODEC in order to analyze the video image quality and the video compression degree reached after the encoding process. It is made a correlation between these results and the objective function of the SMC. The picture quality is measured by the metric most often used in literature, the PSNR (Peak Signal to Noise Ratio), which is calculated by the CODEC at the end of a video encoding process. It was found that as the objective function has increased, the CODEC reached a better image quality and a higher video compression.

With the constant advances in technology, people are using more wireless products, such as earphones or speakers whereas many of them use Bluetooth. With the current advances in Bluetooth technology, consumers and manufacturers have a hard time keeping up with the pace. Thus, when it comes to factors such as battery drainage, CPU usage, and range there is missing knowledge. This study is conducted to find out what effect the different codecs have on these factors, by comparing the two most commonly used codecs SBC and AAC. Using a codec that has lower battery drainage whilst still having a good enough audio quality can have a positive impact on our society and environment. Needing less electricity, lessens the overall energy consumption and directly lowers the energy production. Our results indicate that there is a significant difference in CPU usage but not in battery drainage or range.

Streaming of on-line gaming video is a challenging problem because of the enormous amounts of video data that need to be sent during game playing, especially within the limitations of uplink capabilities. The encoding complexity is also a challenge because of the time delay while on-line gamers are communicating. The main goal of this research study is to propose an enhanced on-line game video streaming system. First, the most common video coding techniques have been evaluated. The evaluation study considers objective and subjective metrics. Three widespread video coding techniques are selected and evaluated in the study; H.264, MPEG-4 Visual and VP- 8. Diverse types of video sequences were used with different frame rates and resolutions. The effects of changing frame rate and resolution on compression efficiency and viewers' satisfaction are also presented. Results showed that the compression process and perceptual satisfaction are severely affected by the nature of the compressed sequence. As a result, H.264 showed higher compression efficiency for synthetic sequences and outperformed other codecs in the subjective evaluation tests. Second, a fast inter prediction technique to speed up the encoding process of H.264 has been devised. The on-line game streaming service is a real time application, thus, compression complexity significantly affects the whole process of on-line streaming. H.264 has been recommended for synthetic video coding by our results gained in codecs comparative studies. However, it still suffers from high encoding complexity; thus a low complexity coding algorithm is presented as fast inter coding model with reference management technique. The proposed algorithm was compared to a state of the art method, the results showing better achievement in time and bit rate reduction with negligible loss of fidelity. Third, recommendations on tradeoff between frame rates and resolution within given uplink capabilities are provided for H.264 video coding. The recommended tradeoffs are offered as a result of extensive experiments using Double Stimulus Impairment Scale (DSIS) subjective evaluation metric. Experiments showed that viewers' satisfaction is profoundly affected by varying frame rates and resolutions. In addition, increasing frame rate or frame resolution does not always guarantee improved increments of perceptual quality. As a result, tradeoffs are recommended to compromise between frame rate and resolution within a given bit rate to guarantee the highest user satisfaction. For system completeness and to facilitate the implementation of the proposed techniques, an efficient game video streaming management system is proposed. Compared to existing on-line live video service systems for games, the proposed system provides improved coding efficiency, complexity reduction and better user satisfaction.