Η παρούσα ειδική ερευνητική εργασία εκπονήθηκε στα πλαίσια του Διατμηματικού Προγράμματος Μεταπτυχιακών Σπουδών Ειδίκευσης στα “Συστήματα Επεξεργασίας Σημάτων και Επικοινωνιών” στο Τμήμα Φυσικής του Πανεπιστημίου Πατρών. Αντικείμενο της παρούσας εργασίας είναι η σχεδίαση και ανάπτυξη του MPEG – 4 Simple Profile CODEC στο περιβάλλον Simulink με σκοπό την τελική εκτέλεση του αλγορίθμου DSP που θα προκύψει, στην πλατφόρμα ανάπτυξης TMS320DM6437 EVM. Στο πρώτο κεφάλαιο ορίζεται η έννοια της κωδικοποίησης βίντεο σε πραγματικό χρόνο και περιγράφεται η σύγχυση που επικρατεί γύρω από αυτήν. Επίσης γίνεται μια περιγραφή των επεξεργαστών ψηφιακού σήματος ως προς τα τυπικά χαρακτηριστικά που διαθέτουν, την αρχιτεκτονική τους, την αρχιτεκτονική μνήμης, τα στοιχεία υλικού που διαθέτουν για τη ροή του DSP προγράμματος, ενώ παράλληλα, παρουσιάζεται η ιστορική εξέλιξη των DSPs που οδήγησε στους σύγχρονους DSPs και οι οποίοι, διαθέτουν καλύτερες επιδόσεις από τους προπάτορές τους, και αυτό χάρη στις τεχνολογικές και αρχιτεκτονικές εξελίξεις όπως, οι χαμηλότεροι κανόνες σχεδίασης, η γρήγορη προσπέλαση κρυφής μνήμης δύο επιπέδων, η σχεδίαση του DMA και ενός μεγαλύτερου συστήματος διαύλου. Στο τέλος του κεφαλαίου παρουσιάζεται η αρχιτεκτονική της πλατφόρμας ανάπτυξης TMS320DM6437 EVM καθώς και οι διεπαφές υλικού που διαθέτει για την είσοδο και έξοδο βίντεο/ήχου από αυτήν. Στο δεύτερο κεφάλαιο γίνεται μια εκτενής παρουσίαση των εννοιών που συναντώνται στην κωδικοποίηση βίντεο. Στην αρχή του κεφαλαίου απεικονίζεται το γενικό μοντέλο ενός κωδικοποιητή/αποκωδικοποιητή και βάσει αυτού προχωράμε στην περιγραφή του χρονικού μοντέλου, το οποίο επιβάλλει την πρόβλεψη του τρέχοντος πλαισίου βίντεο χρησιμοποιώντας το προηγούμενο, ενώ παράλληλα, εξηγεί και μεθόδους για την εκτίμηση κίνησης περιοχών (μακρομπλοκ) μέσα στο πλαίσιο ενός βίντεο και το πώς μπορεί να γίνει ο υπολογισμός του σφάλματος κίνησης τους. Στη συνέχεια περιγράφεται το μοντέλο εικόνας το οποίο στην πράξη αποτελείται από τρία συστατικά μέρη: τον μετασχηματισμό (αποσυσχετίζει και συμπιέζει τα δεδομένα), την κβάντιση (μειώνει την ακρίβεια των μετασχηματισμένων δεδομένων) και την ανακατάταξη (ανακατατάσσει τα δεδομένα ούτως ώστε να ομαδοποιήσει μαζί τις σημαντικές τιμές). Οι συντελεστές του μετασχηματισμού μετά την ανακατάταξη και την κωδικοποίηση, μπορούν να κωδικοποιηθούν περαιτέρω με τη χρήση κωδικών μεταβλητού μήκους (Huffman κωδικοποίηση) ή μέσω αριθμητικής κωδικοποίησης. Στο τέλος του κεφαλαίου περιγράφεται το υβριδικό μοντέλο DPCM/DCT CODEC πάνω στον οποίο στηρίζεται και η υλοποίηση του MPEG – 4 Simple Profile CODEC. Στο τρίτο κεφάλαιο ουσιαστικά γίνεται μια περιγραφή των χαρακτηριστικών του MPEG – 4 Simple Profile CODEC, των εργαλείων που χρησιμοποιεί, της έννοιας αντικείμενο που πλέον υπεισέρχεται στην κωδικοποίηση βίντεο καθώς και τα είδη προφίλ και επιπέδων που υποστηρίζει το συγκεκριμένο πρωτόκολλο κωδικοποίησης/αποκωδικοποίησης. Στο τέταρτο κεφάλαιο παρουσιάζεται η υλοποίηση του κωδικοποιητή, του αποκωδικοποιητή του MPEG – 4 Simple Profile CODEC καθώς και των επιμέρους υποσυστημάτων που τους απαρτίζουν. Στο πέμπτο κεφάλαιο περιγράφεται η αλληλεπίδραση του χρήστη με το σύστημα κωδικοποίησης/αποκωδικοποίησης, τι παράμετροι χρειάζονται να δοθούν ως είσοδοι από αυτόν, καθώς και πως είναι δυνατή η χρήση του συγκεκριμένου συστήματος. / This project objective is the design and development of MPEG – 4 Simple Profile CODEC in Simulink environment in order to execute the resulting DSP algorithm on the development platform TMS320DM6437 EVM. The first chapter defines the term of real – time video coding which sometimes is misunderstood by most people. Besides there is a brief description of DSP systems, which includes information about their typical characteristics, their architecture, their memory architecture and the hardware elements provided with in order to support the flow of a DSP program. It is also presented the evolution of DSPs through time, which finally gave the modern DSPs with better performance than their ancestors thanks to the technological and architectonical improvements such as, lower design rules, fast-access two-level cache, (E)DMA circuitry and a wider bus system. At the end of this chapter it is presented the architecture of TMS320DM6437 EVM board and its input/output hardware interfaces for video and sound. At the second chapter there is an extensive presentation of terms found at the science of coding/decoding video. At the beginning of this chapter it is depicted a general model including a video encoder/decoder and this is the reason for the description of temporal model, which includes the prediction of current frame from the previous one, and at the same time it explains the computation methods of macroblock motion estimation and motion compensation. Continuing it is described the image model aparted from three component parts, the transformation (decorrelation and data compression), the quantization (reduces the accuracy of transformed data) and the reordering (reorders data on a way that groups significant values all together). The transform coefficients after reordering and coding, can be further coding by using variable length coding (Huffman coding) or arithmetic coding. At the end of the chapter the hybrid model of DPCM/DCT CODEC is described and this is the one where the implementation of MPEG – 4 Simple Profile CODEC has been set up. At the third chapter there is a description about the characteristics of MPEG – 4 Simple Profile CODEC, the tools used, the “object” term, which appears on video coding/decoding and also what are the profiles and levels supported by the specific video encoding/decoding protocol. Finally it is described how the coding of rectangular frames is done and the Simulink model of MPEG – 4 Simple Profile CODEC which is the base for the implementation of DSP algorithm executed on the development platform. At the forth chapter we present the implementation of MPEG – 4 Simple Profile CODEC encoder/decoder and their partial subsystems. At the fifth chapter it is described the interaction between user and the CODEC, what are the parameters needed to be entered as inputs and how the system can be used.

In digital video compression, apart from storage, successful transmission of the compressed video data over the bandwidth limited erroneous channels is another important issue. To enable a video codec for broadcasting application, it is required to implement the corresponding coding tools (e.g. error-resilient coding, rate control etc.). They are normally non-normative parts of a video codec and hence their specifications are not defined in the standard. In Dirac as well, the original codec is optimized for storage purpose only and so, several non-normative part of the encoding tools are still required in order to be able to use in other types of application. Being the "Research and Developments of the Dirac Video Codec" as the research title, phase I of the project is mainly focused on the error-resilient transmission over a noisy channel. The error-resilient coding method used here is a simple and low complex coding scheme which provides the error-resilient transmission of the compressed video bitstream of Dirac video encoder over the packet erasure wired network. The scheme combines source and channel coding approach where error-resilient source coding is achieved by data partitioning in the wavelet transformed domain and channel coding is achieved through the application of either Rate-Compatible Punctured Convolutional (RCPC) Code or Turbo Code (TC) using un-equal error protection between header plus MV and data. The scheme is designed mainly for the packet-erasure channel, i.e. targeted for the Internet broadcasting application. But, for a bandwidth limited channel, it is still required to limit the amount of bits generated from the encoder depending on the available bandwidth in addition to the error-resilient coding. So, in the 2nd phase of the project, a rate control algorithm is presented. The algorithm is based upon the Quality Factor (QF) optimization method where QF of the encoded video is adaptively changing in order to achieve average bitrate which is constant over each Group of Picture (GOP). A relation between the bitrate, R and the QF, which is called Rate-QF (R-QF) model is derived in order to estimate the optimum QF of the current encoding frame for a given target bitrate, R. In some applications like video conferencing, real-time encoding and decoding with minimum delay is crucial, but, the ability to do real-time encoding/decoding is largely determined by the complexity of the encoder/decoder. As we all know that motion estimation process inside the encoder is the most time consuming stage. So, reducing the complexity of the motion estimation stage will certainly give one step closer to the real-time application. So, as a partial contribution toward realtime application, in the final phase of the research, a fast Motion Estimation (ME) strategy is designed and implemented. It is the combination of modified adaptive search plus semi-hierarchical way of motion estimation. The same strategy was implemented in both Dirac and H.264 in order to investigate its performance on different codecs. Together with this fast ME strategy, a method which is called partial cost function calculation in order to further reduce down the computational load of the cost function calculation was presented. The calculation is based upon the pre-defined set of patterns which were chosen in such a way that they have as much maximum coverage as possible over the whole block. In summary, this research work has contributed to the error-resilient transmission of compressed bitstreams of Dirac video encoder over a bandwidth limited error prone channel. In addition to this, the final phase of the research has partially contributed toward the real-time application of the Dirac video codec by implementing a fast motion estimation strategy together with partial cost function calculation idea.

Des sites de visionnement de contenu audio-vidéo en temps-réel comme YouTube sont devenus très populaires. Le téléchargement des fichiers audio/vidéo consomme une quantité importante de bande passante des réseaux Internet. L’utilisation de codecs à bas débit permet de compresser la taille des fichiers transmis afin de consommer moins de bande passante. La conséquence est une diminution de la qualité de ce qui est transmis. Une diminution de qualité mène à l’apparition de défauts perceptibles dans les fichiers. Ces défauts sont appelés des artifices de compression. L’utilisation d’un algorithme de post-traitement sur les fichiers sonores pourrait augmenter la qualité perçue de la musique transmise en corrigeant certains artifices à la réception, sans toutefois consommer davantage de bande passante. Pour rehausser la qualité subjective des fichiers sonores, il est d’abord nécessaire de déterminer quelles caractéristiques dégradent la qualité perceptuelle. Le présent projet a donc pour objectif le développement d’un algorithme capable de localiser et de corriger de façon non intrusive, un artifice provoqué par des discontinuités et des incohérences au niveau des harmoniques qui dégrade la qualité objective dans les signaux sonores compressés à bas débits (8 – 12 kilobits par seconde).

All multimedia devices now incorporate video CODECs that comply with international video coding standards such as H.264 / MPEG4-AVC and the new High Efficiency Video Coding Standard (HEVC) otherwise known as H.265. Although the standard CODECs have been designed to include algorithms with optimal efficiency, large number of coding parameters can be used to fine tune their operation, within known constraints of for e.g., available computational power, bandwidth, consumer QoS requirements, etc. With large number of such parameters involved, determining which parameters will play a significant role in providing optimal quality of service within given constraints is a further challenge that needs to be met. Further how to select the values of the significant parameters so that the CODEC performs optimally under the given constraints is a further important question to be answered. This thesis proposes a framework that uses machine learning algorithms to model the performance of a video CODEC based on the significant coding parameters. Means of modelling both the Encoder and Decoder performance is proposed. We define objective functions that can be used to model the performance related properties of a CODEC, i.e., video quality, bit-rate and CPU time. We show that these objective functions can be practically utilised in video Encoder/Decoder designs, in particular in their performance optimisation within given operational and practical constraints. A Multi-objective Optimisation framework based on Genetic Algorithms is thus proposed to optimise the performance of a video codec. The framework is designed to jointly minimize the CPU Time, Bit-rate and to maximize the quality of the compressed video stream. The thesis presents the use of this framework in the performance modelling and multi-objective optimisation of the most widely used video coding standard in practice at present, H.264 and the latest video coding standard, H.265/HEVC. When a communication network is used to transmit video, performance related parameters of the communication channel will impact the end-to-end performance of the video CODEC. Network delays and packet loss will impact the quality of the video that is received at the decoder via the communication channel, i.e., even if a video CODEC is optimally configured network conditions will make the experience sub-optimal. Given the above the thesis proposes a design, integration and testing of a novel approach to simulating a wired network and the use of UDP protocol for the transmission of video data. This network is subsequently used to simulate the impact of packet loss and network delays on optimally coded video based on the framework previously proposed for the modelling and optimisation of video CODECs. The quality of received video under different levels of packet loss and network delay is simulated, concluding the impact on transmitted video based on their content and features.

We present an energy conscious location-aware computing system that provides relevant information about the users current location. The location-aware computing system is initialized with a map (in the form of a graph) as well as audio files associated with several locations in the map. The system consists of: GPS receiver module, Serial port, Compact flash module, Stereo codec, Power manager module implementing three sub modules namely, GPS-to-real-world position conversion module (implements algorithm to convert GPS co-ordinates to graph nodes), Nearest-location-search module (implements modified Dijkstras algorithm), and User speed estimation module. The location-aware computing system receives the GPS co-ordinates for the current location from GPS receiver through the serial port. The system converts the GPS co-ordinates to map co-ordinates stored in the Compact Flash card. If the current location matches the landmarks of interest in the site, then the relevant audio details of the current location is played out to the user. The power manager sets the GPS co-ordinates update frequency to avoid keeping the system component on throughout the entire course of travel. The power manager implements an algorithm that works as follows: at any given location, the algorithm predicts the user speed by exponential average approach. The attenuation factor of this approach can be varied to account for the user speed history. The estimated speed is used to predict the time (say T) required to reach the next nearest location determined by Nearest-location-search module implementing modified Dijkstras algorithm. The subsystems are shut-down or switched to low-power mode for time T. After time T, the system will wake up and re-execute the algorithm.

<p>The voice channel in mobile communication systems have high priority and are almost always available. By using the voice channel also for data transmissions it is possible to get the same availability as for voice calls. But due to speech codecs in the voice channel, regular modems can not be used and special techniques are needed to transmit data.</p><p>This thesis presents methods to transmit data over the voice channel in a GSM, UMTS or TETRA network. The focus has been on robust data transmission rather than high data bit rates. Approaches are introduced which improve the reliability for transmissions even for systems with low rate speech codecs and channels with some distortion.</p><p>The results of the thesis are suggestions of symbol patterns and ways to create and adapt symbols for specific application and channel conditions to achieve the desired goal for the application.</p>

<p>The market of portable digital audio players (DAPs) have literally exploded the last couple of years. Other markets has grown as well. PDAs, GPS receivers, mobile phones, and so on. This resulted in more advanced ICs and SoCs becoming publically available, eliminating the need for in-house ASICs, thus enableing smaller actors to enter the markets.</p><p>This thesis explores the possibilities of developing a low power, hand-held device on a very limited budget and strict time scale.</p><p>This thesis report also covers all the steps taken in the development procedure.</p>

Denna undersökning söker ett svar på hur den relativt vana lyssnarens bedömning av ljudkvalitet påverkas av ett så kallat öppet test, där det som bedöms är känd för lyssnaren, jämfört med ett blindtest, där detta objekt är okänt. Frågan appliceras på kvalitetsbedömningen av digitala kodningstekniker, d.v.s. hur lyssnaren påverkas av att valet av kodningsteknik som avlyssnas är känd eller inte. För att ta reda på detta genomfördes ett lyssningstest med nio deltagare. Deltagarna fick betygssätta perceptuellt kodade ljudfiler mot en känd referens, både som ett blindtest samt i ett öppet test. Resultatet är mångtydigt och inga generella slutsatser för hur lyssnaren påverkas av ett öppet test jämfört med ett blindtest går att uppfatta. Resultatet visar dock att påverkan ett öppet test har på lyssnarens bedömning är högst individuell. Lyssningstest i form av blindtest bör därför användas för att uppnå pålitligast resultat.

The voice channel in mobile communication systems have high priority and are almost always available. By using the voice channel also for data transmissions it is possible to get the same availability as for voice calls. But due to speech codecs in the voice channel, regular modems can not be used and special techniques are needed to transmit data. This thesis presents methods to transmit data over the voice channel in a GSM, UMTS or TETRA network. The focus has been on robust data transmission rather than high data bit rates. Approaches are introduced which improve the reliability for transmissions even for systems with low rate speech codecs and channels with some distortion. The results of the thesis are suggestions of symbol patterns and ways to create and adapt symbols for specific application and channel conditions to achieve the desired goal for the application.

<p>Examensarbetet bakom denna rapport utgör en undersökande studie om hur transmission av mediadata i nätverk kan göras effektivare. Det kan åstadkommas genom att viss tilläggsinformation avsedd för att jämna ut datatakten adderas i det realtidsprotokoll, Real Time Protocol, som används för strömmande media. Genom att försöka skicka lika mycket data under alla konsekutiva tidsintervall i sessionen kommer datatakten vid en godtycklig tidpunkt med större sannolikhet att vara densamma som vid tidigare klockslag. En streamingserver kan tolka, hantera och skicka data vidare enligt instruktionerna i protokollets sidhuvud. Datatakten jämnas ut genom att i förtid, under tidsintervall som innehåller mindre data, skicka även senare data i strömmen. Resultatet av detta är en utjämnad datataktskurva som i sin tur leder till en jämnare användning av nätverkskapaciteten.</p><p>Arbetet inkluderar en översiktlig analys av beteendet hos strömmande media, bakgrundsteori om filkonstruktion och nätverksteknologier samt ett förslag på hur mediafiler kan modifieras för att uppfylla syftet med examensarbetet. Resultat och diskussion kan förhoppningsvis användas som underlag för en framtida implementation av en applikation ämnad att förbättra trafikflöden över nätverk.</p>