A novel residual prediction algorithm is proposed for high-bit-rate video coding in this work. We analysis the relationship between the residual data and different quantization parameters, according to the comparison results, we observe that the residual data is raised rapidly when the quality increases. Consequently, in order to reduce the bitrate, we propose a new residual prediction algorithm, it mainly reduce the residual data when the quantization parameter is finer. The proposed algorithm not only reduces the bitrate but also improves the video quality for high-bit-rate coding. Experimental results show that the proposed algorithm outperforms H.264/AVC. Compared to H.264/AVC, the proposed method decreases about 9.66% bitrate in average. The experimental results demonstrated that the second-order-residual prediction algorithm is efficiency for high-bit-rate coding.

This thesis applies the one-dimensional (1-D) and two-dimensional (2-D) cubic-spline interpolation (CSI) schemes to MPEG standard for very low-bit rate video coding. In addition, the CSI scheme is used to implement the scalable video compression scheme in this thesis. The CSI scheme is based on the least-squares method with a cubic convolution function. It has been shown that the CSI scheme yields a very accurate algorithm for smoothing and obtains a better quality of reconstructed image than linear interpolation, linear-spline interpolation, cubic convolution interpolation, and cubic B-spline interpolation. In order to obtain a very low-bit rate video, the CSI scheme is used along with the MPEG-1 standard for video coding. Computer simulations show that this modified MPEG not only avoids the blocking effect caused by MPEG at high compression ratio but also gets a very low-bit rate video coding scheme that still maintains a reasonable video quality. Finally, the CSI scheme is also used to achieve the scalable video compression. This new scalable video compression scheme allows the data rate to be dynamically changed by the CSI scheme, which is very useful when operates under communication networks with different transmission capacities.

Η συγκεκριμένη εργασία έχει ως αντικείμενο τη σύγκριση των ρυθμών μετάδοσης γραμμής VDSL που επιτυγχάνονται με δύο διαφορετικούς τρόπους κατανομής ισχύος: τις μάσκες ισχύος και τον αλγόριθμο iterative waterfilling. Οι μάσκες ισχύος είναι η μέθοδος που χρησιμοποείται σήμερα στις γραμμές DSL, ενώ ο αλγόριθμος iterative waterfilling έχει προταθεί ως εναλλακτική λύση που επιτυγχάνει καλύτερους ρυθμούς μετάδοσης. Για την πραγματοποίηση της σύγκρισης υλοποιήθηκαν προσομοιώσεις της κάθε μεθόδου σε περιβάλλον Matlab και έγινε σύγκριση των αποτελεσμάτων για διάφορες τιμές μήκους γραμμών VDSL και σε διαφορετικές συνθήκες θορύβου. Τα αποτελέσματα δείχνουν πως η μέθοδος iterative waterfilling αυξάνει το ρυθμό μετάδοσης των γραμμών VDSL και είναι πιο αποτελεσματική από τις μάσκες ισχύος σε περιβάλλον αυξημένου θορύβου. / The particular work has as object the comparison of the bit rates for a VDSL line that is achieved with two different ways of power distribution: the power masks and the algorithm iterative waterfilling.The power masks are the method that is used today in DSL lines, while the algorithm iterative waterfilling has been proposed as alternative solution that achieves better bit rates. For the purpose of this comparison were materialised simulations of each method in environment Matlab and it became comparison of results for different lengths of lines VDSL and in different conditions of noise. The results show that the method iterative waterfilling increases the bit rate of lines VDSL and is more effective than the power masks when the noise is big.

[pt] Esta tese propõe novas melhorias para a codificação CELP a baixas taxas de bits. Primeiro, é proposto um algoritmo CELP em que a complexidade do procedimento de busca no dicionário adaptativo é grandemente reduzida, graças a uma modificação introduzida no modelo de síntese CELP. Resultados de simulação mostram que a qualidade da voz codificada com o algoritmo CELP proposto tem qualidade comparável àquela obtida com o algoritmo CELP convencional. As demais contribuições têm o propósito de melhorar a qualidade da voz codificada com o algoritmo CELP a baixas taxas de bits. Uma delas propicia uma codificação mais eficiente da envoltória espectral LPC da voz: é, especificamente, um esquema que combina quantização vetorial e interpolação interbloco dos parâmetros LSF. Com este esquema a envoltória espectral LPC codificada tem boa qualidade a uma taxa de bits tão baixa quanto 1 kb/s. A voz codificada com os algoritmos CELP apresenta freqüentemente distorções em sua envoltória espectral que são causadas por deficiências do sinal de excitação. Esta tese propõe um novo pós-filtro que reduz estas distorções e, com isso, melhora significativamente a qualidade subjetiva da voz codificada. A baixas taxas de bits a estrutura CELP convencional é incapaz de reproduzir com boa qualidade os ataques dos sons sonoros, que são cruciais para uma boa percepção da voz. Nesta tese é descrito um algoritmo CELP que dá prioridade a estes segmentos críticos. Cada bloco da voz é classificado em um dentre dezesseis padrões de sonoridade e cada padrão tem uma configuração de codificação e alocação de bits distintas. Resultados de simulação mostram que a qualidade da voz codificada a 4 kb/s com o algoritmo CELP proposto é significativamente melhor do que aquela conseguida com um codificador CELP convencional, também operando a 4 kb/s. / [en] This work presents new improvements to CELP speech coding at low bit rates. First, a CELP algorithm is proposed in wich the complexity of the adaptive codebook search is gratly decreased. This is achieved by means of a modified model of the CELP synthesizer. Simulation results show that the proposed algorithm can provide speech quality comparable to one obtained with the conventional CELP codec. The rest of contributions aim to improve the quality of speech codec at low bit rates with CELP algorithm. One of them is an efficient scheme for coding the LPC spectral envelope of speech for coding the LPC spectral envelope of speech. The proposed scheme combines vector quantization and interpolation of LSF parameters, and it provides a coded spectral envelope with very good quality at 1 kb/s. Speech coded with CELP codecs frequently displays distortions in its spectral envelope that are produced by deficient excitation. This thesis proposes a new postfilter that enhances the perceptual quality of codec speech by decreasin these distortions. This work presents new improvements to CELP speech coding at low bit rates. First, a CELP algorithm is proposed in wich the complexity of the adaptive codebook search is gratly decreased. This is achieved by means of a modified model of the CELP synthesizer. Simulation results show that the proposed algorithm can provide speech quality comparable to one obtained with the conventional CELP codec. The rest of contributions aim to improve the quality of speech codec at low bit rates with CELP algorithm. One of them is an efficient scheme for coding the LPC spectral envelope of speech for coding the LPC spectral envelope of speech. The proposed scheme combines vector quantization and interpolation of LSF parameters, and it provides a coded spectral envelope with very good quality at 1 kb/s. Speech coded with CELP codecs frequently displays distortions in its spectral envelope that are produced by deficient excitation. This thesis proposes a new postfilter that enhances the perceptual quality of codec speech by decreasin these distortions. Voiced onsets are crucial for a good perception of speech but, at low bit rates, the conventional CELP is unable to reproduce them with good quality. This work presents a CELP algorithm into one of a set of sixteen voicing patterns. A distinct coding configuration and bit allocation are applied to each pattern. Simulation results show that the quality of speech codec with the proposed 4 kb/s CELP codec is significantly bette than the one obtained with conventional 4 kb/s CELP codec.

In the field of communications, a high data rate and low multi-path fading is required for efficient information exchange. Orthogonal Frequency Division Multiplexing (OFDM) is a widely accepted IEEE 802.11n (and many others) standard for usage in communication systems operating in fading dispersive channels. In this thesis, we modeled the OFDM algorithm at the behavioral level in VHDL/Verilog that was successfully synthesized/verified on an FPGA. Due to rapid technology scaling, FPGAs have become popular and are low-cost and high performance alternatives to (semi-) custom ASICs. Further, due to reprogramming flexibility, FPGAs are useful in rapid prototyping. As per the IEEE standard, we implemented both transmitter and receiver with four modulation schemes (BPSK, QPSK, QAM16, and QAM64). We extensively verified the design in simulation as well as on Altera Stratix IV EP4SGX230KF40C2 FPGA (Terasic DE4 Development Board). The synthesized design ran at 100 MHz clock frequency incurring 54 µ sec. end-to-end latency and 8% logic utilization.

During the last decade, the data rates provided by mobile networks have improved to the point that it is now feasible to provide richer services, such as streaming multimedia, to mobile users. However, due to factors such as radio interference and cell load, the throughput available to a client varies over time. If the throughput available to a client decreases below the media’s bit rate, the client’s buffer will eventually become empty. This causes the client to enter a period of rebuffering, which degrades user experience. In order to avoid this, a streaming server may provide the media at different bit rates, thereby allowing the media’s bit rate (and quality) to be modified to fit the client’s bandwidth. This is referred to as adaptive streaming. The aim of this thesis is to devise an algorithm to find the media quality most suitable for a specific client, focusing on how to detect that the user is able to receive content at a higher rate. The goal for such an algorithm is to avoid depleting the client buffer, while utilizing as much of the bandwidth available as possible without overflowing the buffers in the network. In particular, this thesis looks into the difficult problem of how to do adaptation for live content and how to switch to a content version with higher bitrate and quality in an optimal way. This thesis examines if existing adaptation mechanisms can be improved by considering the characteristics of different mobile networks. In order to achieve this, a study of mobile networks currently in use has been conducted, as well as experiments with streaming over live networks. The experiments and study indicate that the increased available throughput can not be detected by passive monitoring of client feedback. Furthermore, a higher data rate carrier will not be allocated to a client in 3G networks, unless the client is sufficiently utilizing the current carrier. This means that a streaming server must modify its sending rate in order to find its maximum throughput and to force allocation of a higher data rate carrier. Different methods for achieving this are examined and discussed and an algorithm based upon these ideas was implemented and evaluated. It is shown that increasing the transmission rate by introducing stuffed packets in the media stream allows the server to find the optimal bit rate for live video streams without switching up to a bit rate which the network can not support. This thesis was carried out during the summer and autumn of 2008 at Ericsson Research, Multimedia Technologies in Kista, Sweden. / Under det senaste decenniet har överföringshastigheterna i mobilnätet ökat så pass mycket att detnu är möjligt att erbjuda användarna mer avancerade tjänster, som till exempel strömmandemultimedia. I mobilnäten varierar dock klientens bandbredd med avseende på tiden på grund avfaktorer som störningar på radiolänken och lasten i cellen. Om en klients överföringshastighetsjunker till mindre än mediets bithastighet, kommer klientens buffert till slut att bli tom. Dettaleder till att klienten inleder en period av ombuffring, vilket försämrar användarupplevelsen. Föratt undvika detta kan en strömmande server erbjuda mediet i flera olika bithastigheter, vilket gördet möjligt för servern att anpassa bithastigheten (och därmed kvalitén) till klientens bandbredd.Denna metod kallas för adaptive strömning. Syftet för detta examensarbete är att utveckla en algoritm, som hittar den bithastighet som är bästlämpad för en specifik användare med fokus på att upptäcka att en klient kan ta emot media avhögre kvalité. Målet för en sådan algoritm är att undvika att klientens buffert blir tom ochsamtidigt utnyttja så mycket av bandbredden som möjligt utan att fylla nätverksbuffertarna. Merspecifikt undersöker denna rapport det svåra problemet med hur adaptering för direktsänd mediakan utföras. Examensarbetet undersöker om existerande adapteringsmekanismer kan förbättras genom attbeakta de olika radioteknologiers egenskaper. I detta arbete ingår både en studie avradioteknologier, som för tillfället används kommersiellt, samt experiment med strömmandemedia över dessa. Resultaten från studien och experimenten tyder på att ökad bandbredd inte kanupptäckas genom att passivt övervaka ”feedback” från klienten. Vidare kommer inte användarenatt allokeras en radiobärare med högre överföringshastighet i 3G-nätverk, om inte den nuvarandebäraren utnyttjas maximalt. Detta innebär att en strömmande server måste variera sinsändningshastighet både för att upptäcka om mer bandbredd är tillgänglig och för att framtvingaallokering av en bärare med högre hastighet. Olika metoder för att utföra detta undersöks ochdiskuteras och en algoritm baserad på dessa idéer utvecklas. Detta examensarbete utfördes under sommaren och hösten 2008 vid Ericsson Research,Multimedia Technologies i Kista, Sverige.

No / We propose a hybrid scheme to implement an object driven, block based algorithm to achieve low bit-rate compression of stereo image pairs. The algorithm effectively combines the simplicity and adaptability of the existing block based stereo image compression techniques with an edge/contour based object extraction technique to determine appropriate compression strategy for various areas of the right image. Unlike the existing object-based coding such as MPEG-4 developed in the video compression community, the proposed scheme does not require any additional shape coding. Instead, the arbitrary shape is reconstructed by the matching object inside the left frame, which has been encoded by standard JPEG algorithm and hence made available at the decoding end for those shapes in right frames. Yet the shape reconstruction for right objects incurs no distortion due to the unique correlation between left and right frames inside stereo image pairs and the nature of the proposed hybrid scheme. Extensive experiments carried out support that significant improvements of up to 20% in compression ratios are achieved by the proposed algorithm in comparison with the existing block-based technique, while the reconstructed image quality is maintained at a competitive level in terms of both PSNR values and visual inspections

Dans le cadre de la nouvelle société de l'information multimédia et communicante, cette thèse propose des contributions méthodologiques et techniques relatives à la représentation, l'animation et la transmission des objets virtuels.<br /><br />Les méthodes existantes sont analysées de façon comparée et les performances des standards multimédias actuels évaluées en termes de réalisme d'animation et de débit de transmission. Pour surmonter les limitations mises en évidence, un nouveau cadre de modélisation et d'animation de personnages virtuels est proposé. Le modèle SMS (Skeleton, Muscle and Skin), fondé sur le concept de contrôleur de déformation d'un maillage, est introduit et sa formulation mathématique développée. Le graphe de scène 3D et le flux de compression associés à SMS sont décrits. L'approche SMS est évaluée dans le cadre d'un nouveau service de transmission télévisuelle d'un signeur virtuel destinés aux déficients auditifs. Le modèle SMS a été promu dans le standard MPEG-4 version 5.

Most existing telephone networks transmit narrowband coded speech which has been bandlimited to 4 kHz. Compared with normal speech, this speech has a muffled quality and reduced intelligibility, which is particularly noticeable in sounds such as /s/, /f/ and /sh/. Speech which has been bandlimited to 8 kHz is often coded for this reason, but this requires an increase in the bit rate. Wideband enhancement is a scheme that adds a synthesized highband signal to narrowband speech to produce a higher quality wideband speech signal. The synthesized highband signal is based entirely on information contained in the narrowband speech, and is thus achieved at zero increase in the bit rate from a coding perspective. Wideband enhancement can function as a post-processor to any narrowband telephone receiver, or alternatively it can be combined with any narrowband speech coder to produce a very low bit rate wideband speech coder. Applications include higher quality mobile, teleconferencing, and internet telephony. This thesis examines in detail each component of the wideband enhancement scheme: highband excitation synthesis, highband envelope estimation, and narrowband-highband envelope continuity. Objective and subjective test measures are formulated to assess existing and new methods for all components, and the likely limitations to the performance of wideband enhancement are also investigated. A new method for highband excitation synthesis is proposed that uses a combination of sinusoidal transform coding-based excitation and random excitation. Several new techniques for highband spectral envelope estimation are also developed. The performance of these techniques is shown to be approaching the limit likely to be achieved. Subjective tests demonstrate that wideband speech synthesized using these techniques has higher quality than the input narrowband speech. Finally, a new paradigm for very low bit rate wideband speech coding is presented in which the quality of the wideband enhancement scheme is improved further by allocating a very small bitstream for highband envelope and gain coding. Thus, this thesis demonstrates that wideband speech can be communicated at or near the bit rate of a narrowband speech coder.

Asynchronous Transfer Mode (ATM) Adaptation Layer Type 2 (AAL2) has been developed to carry low and variable bit rate traffic. It provides high bandwidth efficiency with low packing delay by allowing voice traffic from different AAL2 channels to be multiplexed onto a single ATM virtual channel connection. Examples of where AAL2 are used include the Code Division Multiple Access and the Third Generation mobile telephony networks. The main objective of this thesis is to study traditional and novel AAL2 multiplexing methods and to characterise their performance when carrying low and variable bit rate (VBR) voice traffic. This work develops a comprehensive QoS framework which is used as a basis to study the performance of the AAL2 multiplexer system. In this QoS framework the effects of packet delay, delay variation, subjective voice quality and bandwidth utilisation are all used to determine the overall performance of the end-to-end system for the support of real time voice communications. Extensions to existing AAL2 voice multiplexers are proposed and characterised. In the case where different types of voice applications are presented to the AAL2 multiplexer, it was observed that increased efficiency gains are possible when a priority queuing scheme is introduced into the traditional AAL2 multiplexer system. Studies of the voice traffic characteristics and their effects on the performance of the AAL2 multiplexer are also investigated. It is shown that particular source behaviours can have deleterious effect on the performance of the AAL2 multiplexer. Methods of isolating these voice sources are examined and the performance of the AAL2 multiplexer re-evaluated to show the beneficial effects of a particular source isolation technique. The extent to which statistical multiplexing is possible for real time variable VBR sources is theoretically examined. These calculations highlight the difficulties in multiplexing VBR real time traffic while maintaining guaranteed delay bounds for these sources. Based on these calculations, multiplexing schemes that incorporate data transfers within the real time traffic transfer are proposed as alternatives for utilising unused bandwidth caused by the VBR nature of the voice traffic.