Communication has always been a vital part of our society, and day-to-day communication is increasingly becoming more digital. VoIP (voice over IP) is used for real-time communication, and to be able to send the information over the internet must the speech be compressed to lower the number of bits needed for transmission. Codecs are used to compress the speech, or any other type of data transmitting over a network, which can introduce some noise if lossy compression is used. Depending on the bandwidth, bit rate, and codec used can distortion be minimized which would result in higher perceived speech quality. In the thesis, two codecs, G729D and Opus, were tested and evaluated with two different objective perceive speech quality metrics, POLQA and PESQ. The codecs were also tested with different emulated network scenarios, 2G, 3G, 4G, satellite two-hop, and LAN. Furthermore, Opus was tested with and without VAD (voice activity detection) to see how VAD could affect the perceived speech quality. The different network scenarios did not impact the results of the evaluation, since the main difference between the network scenarios was latency, which POLQA and PESQ do not consider in the evaluation. Opus achieved a higher MOS-LQO (mean opinion score listening quality objective) than G729D. However, when VAD was enabled with Opus for a low bit rate, 8 kbit/s, the MOS-LQO was lower than without VAD.

The emergence of Bluetooth Low Energy in combination with optimized coders has made it possible to transfer digital audio at very low bitrates, paving the way for small devices with longlasting batteries. The aim of this study is to compare the audio codecs LC3 and aptX, as well as peoples’ attitude towards audio quality in different contexts. Two open source implementations of the codecs are evaluated in terms of time for execution. Furthermore, the perceived audio quality of low bitrates are subjectively compared in a listening test in combination with a questionnaire regarding peoples’ attitude towards audio quality. The results show that LC3 is capable of delivering satisfying audio quality at very low bitrates, whilst also outperforming aptX. It will be interesting to see how LC3 will affect transmission latency, battery life and overall QoS once it is established in everyday products

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The thesis Data transmission error-protection with BCH codes deals with a large class of random-error correcting cyclic codes which are able to protect binary data and can be used for example in data storages, high speed modems. Bose, Chaudhuri and Hocquenghem (BCH) codes operate over algebraic structures called Galois fields. The BCH encoding is the same as cyclic encoding and can be done with linear feedback shift register but decoding is more complex and can be done with different algorithms - in this thesis there are two algorithms for decoding Peterson and Berlekam-Massey mentioned. The aim of this thesis is to find BCH code which is able to correct t = 6 independent errors in up to data sequence n = 150 bits, then peruse possible realizations of the codecs and set criteria for the best realization, then design and test this realization. This thesis is split into three main parts. In the first part there are encoding and decoding methods of the BCH code generally described. The second part deals with selecting of the right code and realization. There was chosen BCH (63,30) code and realization with FPGA chip. In the last part is described design of BCH encoder and decoder and compilation in the Altera design software.

Thesis deals with multimedia signal comparison of compression options focused on video and advanced codecs. Specifically it describes the encoding and decoding of video recordings according to the MPEG standard. The theoretical part of the thesis describes characteristic properties of the video signal and justification for the need to use recording and transmission compression. There are also described methods for elimination of encoded video signal redundancy and irrelevance. Further on are discussed ways of measuring the video signal quality. A separate chapter is focused on the characteristics of currently used and promising codecs. In the practical part of the thesis were created functions in Matlab environment. These functions were implemented into graphic user interface that simulates the activity of functional blocks of the encoder and decoder. Based on user-specified input parameters it performs encoding and decoding of any given picture, composed of images in RGB format, and displays the outputs of individual functional blocks. There are implemented algorithms for the initial processing of the input sequence including sub-sampling, as well as DCT, quantization, motion compensation and their inverse operations. Separate chapters are dedicated to the realisation of codec description in the Matlab environment and to the individual processing steps output. Further on are mentioned compress algorithm comparisons and the impact of parameter change onto the final signal. The findings are summarized in conclusion.

Denna rapport behandlar det arbete som under våren 2020 utfördes av en grupp på sju civilingenjörsstudenter i data- och mjukvaruteknik vid Linköpings universitet som en del av kursen TDDD96 – Kandidatprojekt i programvaruutveckling. Projektet utfärdades på efterfrågan av företaget Airpelago, och gick ut på att utveckla en stabil grund för ett molnbaserat videoströmningssystem. Denna rapport beskriver den färdiga produkten, förklarar de beslut som togs, beskriver vilka problem som uppstått och hur dessa lösts, samt diskuterar det slutgiltiga arbetet. Rapporten innehåller även sju individuella bidragskrivna av gruppens enskilda medlemmar / <p>Presentationen genomfördes på distans</p>

(in English) Object of this thesis is to demonstrate compact characteristics and analysis of YouTube portal, explain its importance, content and area of provided services. -- The first chapter is applied to basic characteristics of YouTube portal, the second chapter deals with description of procedures, which are used by the portal. The third chapter is all about video. There are discussed issues about video on PC in general, and afterwards, there are described processes related to videos on YouTube. YouTube users, YouTube partners and projects are mainspring of the fourth chapter. Special subchapter outlines possibilites of YouTube in relation to information studies and librarianship. At the end of the fourth chapter user offences are mentioned. The fifth chapter is composed of introduction of several direct and indirect YouTube competitors. The sixth, and the last chapter, contains author's evaluation of YouTube portal [Author's abstract].

This thesis deals with modern methods of a lossy still image and video compression. Wavelet transformation and SPIHT algorithm also belong to these methods. In second half of this thesis, a videocodec is implemented based on acquired knowledge. This codec uses Daubechies wavelets to analyse an image. Afterwards there is a modified SPIHT algorithm applied on gained coefficients. A lot of effort was put in order to optimize this computation. It is possible to use the created codec in Video for Windows, DirectShow and FFmpeg multimedia frameworks. At the end of this thesis, commonly used codecs are compared with newly created one.

This work deals with module of the FITkit platform, which makes it able to play sound files like mp3, ogg, etc. The module also adds to FITkit some new peripherals: color LCD display with touch screen and USB interface, by which we can connect flash drive.

In IEEE 802.11 WLANs, Link Adaptation mechanisms, which choose the transmission rate of each node, provoke unexpected and random variations on the effective channel capacity. When these changes are towards lower bitrates, inelastic flows, such as VoIP, can suffer from sudden congestion, which results on higher packet delays and losses. In this thesis, a VoIP codec adaptation algorithm is proposed as a solution, based on a cross-layer feedback from RTCP packets and the MAC layer, which can adapt the codecs of active calls to adjust them to the multirate scenario. A combination of this algorithm with a call admission control mechanism is also studied. The results show an important improvement in terms of the QoS of the already active flows as also in the total hotspot's capacity. Additionally, by defining a new Grade of Service related parameter, the Q-Factor, which captures the trade-off between dropping and blocking ratio and perceived speech quality, the codec adaptation algorithm can be tuned to achieve maximum capacity without severely penalizing any of those variables, and hence satisfying both technical and user quality requirements. Finally, a new QoS-enabled AP, which implements these enhancements is designed. / En las redes inalámbricas del estándar IEEE 802.11, los mecanismos de adaptación de enlace que eligen la tasa de transmisión de cada nodo, pueden provocar variaciones aleatorias e inesperadas en la capacidad efectiva del canal. Cuando estos cambios son hacia tasas de transmisión mas bajas, los flujos inelásticos, tales como los de VoIP, pueden de repente sufrir congestión, lo que se traduce en aumento de retrasos y pérdidas de paquetes. En esa tesis, se propone un algoritmo de adaptación de codificadores de voz como solución, basado en técnicas multinivel (cross-layer) que combinan el uso de información de diferentes capas, como los paquetes RTCP y la capa MAC, y que puede adaptar los codecs de las llamadas activas para ajustarlos al escenario "multi-rate". Adicionalmente, la combinación de este algoritmo con un mecanismo de control de admisión de llamadas (CAC) se ha estudiado. Los resultados muestran una importante mejora en términos de QoS de los flujos activos como también en la capacidad total del hotspot. Además, mediante la definición de un nuevo factor, el Q-Factor, que puede captar la compensación entre la tasa de corte y de bloqueo de llamadas y de la calidad percibida por esas, el algoritmo de adaptación de codecs se puede ajustar para lograr la máxima capacidad sin penalizar severamente ninguna de esas variables y así satisfacer los requisitos técnicos de calidad y los usuarios. Por último, un nuevo punto de acceso (AP) habilitado para ofrecer calidad de servicio, ha sido diseñado que lleva a cabo estas mejoras.