QoS evaluation of Bandwidth Schedulers in IPTV Networks Offered SRD Fluid Video Traffic

Internet protocol TV (IPTV) is predicted to be the key technology winner in the future. Efforts to accelerate the deployment of IPTV centralized model which is combined of VHO, encoders, controller, access network and Home network. Regardless of whether the network is delivering live TV, VOD, or Time-shift TV, all content and network traffic resulting from subscriber requests must traverse the entire network from the super-headend all the way to each subscriber's Set-Top Box (STB). IPTV services require very stringent QoS guarantees When IPTV traffic shares the network resources with other traffic like data and voice, how to ensure their QoS and efficiently utilize the network resources is a key and challenging issue. For QoS measured in the network-centric terms of delay jitter, packet losses and bounds on delay. The main focus of this thesis is on the optimized bandwidth allocation and smooth data transmission. The proposed traffic model for smooth delivering video service IPTV network with its QoS performance evaluation. According to Maglaris et al [5] first, analyze the coding bit rate of a single video source. Various statistical quantities are derived from bit rate data collected with a conditional replenishment inter frame coding scheme. Two correlated Markov process models (one in discrete time and one in continuous time) are shown to fit the experimental data and are used to model the input rates of several independent sources into a statistical multiplexer. Preventive control mechanism which is to be including CAC, traffic policing used for traffic control. QoS has been evaluated of common bandwidth scheduler( FIFO) by use fluid models with Markovian queuing method and analysis the result by using simulator and analytically, Which is measured the performance of the packet loss, overflow and mean waiting time among the network users.

Study of the audio coding algorithm of the MPEG-4 AAC standard and comparison among implementations of modules of the algorithm

Audio coding is used to compress digital audio signals, thereby reducing the amount of bits needed to transmit or to store an audio signal. This is useful when network bandwidth or storage capacity is very limited. Audio compression algorithms are based on an encoding and decoding process. In the encoding step, the uncompressed audio signal is transformed into a coded representation, thereby compressing the audio signal. Thereafter, the coded audio signal eventually needs to be restored (e.g. for playing back) through decoding of the coded audio signal. The decoder receives the bitstream and reconverts it into an uncompressed signal. ISO-MPEG is a standard for high-quality, low bit-rate video and audio coding. The audio part of the standard is composed by algorithms for high-quality low-bit-rate audio coding, i.e. algorithms that reduce the original bit-rate, while guaranteeing high quality of the audio signal. The audio coding algorithms consists of MPEG-1 (with three different layers), MPEG-2, MPEG-2 AAC, and MPEG-4. This work presents a study of the MPEG-4 AAC audio coding algorithm. Besides, it presents the implementation of the AAC algorithm on different platforms, and comparisons among implementations. The implementations are in C language, in Assembly of Intel Pentium, in C-language using DSP processor, and in HDL. Since each implementation has its own application niche, each one is valid as a final solution. Moreover, another purpose of this work is the comparison among these implementations, considering estimated costs, execution time, and advantages and disadvantages of each one.

Analytical expressions for the FM and AM responses of an integrated laser-modulator

The need for low-chirp and compact transmitters for high-bit-rate optical links has led to the development of integrated laser electroabsorption modulators (ILM). We have investigated feedback effects inducing frequency chirp by developing a model treating the ILM as a whole and obtained analytical expressions of the FM and AM responses. The variation of the frequency chirp with the residual facet reflectivity of the modulator section is calculated. The model predicts the unusual peak in the measured frequency responses and has been used to define design rules.

Comportamento ingestivo de novilhos de corte sob diferentes alturas de pastejo do capim Tanzânia1

Conselho Nacional de Desenvolvimento Científico e Tecnológico (CNPq)

Transmissão DSL em modo fantasma: medições e avaliação de desempenho

A demanda por taxa de bits vem aumentando a cada ano, devido principalmente pela incorporação de aplicativos online ao cotidiano das pessoas. Assim, as companhias prestadoras deste tipo de serviço estão sempre investindo em pesquisa por tecnologias que possibilitem o aumento da taxa de bits com um bom custo-benefício. Seguindo esta perspectiva, este trabalho apresenta o modo de transmissão fantasma (Phantom mode), que visa aumentar a taxa total de transmissão de bits em sistemas DSL, quando múltiplos pares de cobre estão disponíveis. Aqui também são mostrados resultados e discussões importantes sobre desempenho do modo de transmissão fantasma. Também são discutidos procedimentos para a execução de medições de modo fantasma em frequências de 100 kHz a 300 MHz, em uma maneira que permite a medição de modo fantasma e modo diferencial com o mesmo setup. Novos resultados são apresentados sobre canal direto, far-end crosstalk (FEXT) e conversão de modo, para um cabo Cat-5e de 50 metros. Dentre outros fatos, foram verificadas grandes diferenças no canal direto de modo fantasma quando foram comparadas medições com cabo enrolado e desenrolado, e que a forma como o modo fantasma _e construção da influência bastante no comportamento das curvas de canal direto, FEXT e conversão de modo. Também são apresentados resultados de estimação da taxa de bits quando são usados modo fantasma e modo diferencial simultaneamente, e observou-se que ha um ganho de ate 60% quando foram usados este dois modos e vectoring.

Estrutura do pasto e comportamento ingestivo de caprinos em capim-tanzânia submetidos a índices de área foliar residual em lotação intermitente

Conselho Nacional de Desenvolvimento Científico e Tecnológico (CNPq)

Comportamento ingestivo de bovinos mantidos em pastos de Brachiaria brizantha cv. Xaraés manejado em diferentes alturas de pastejo

This experiment was conducted to evaluate the effect of grazing heights on daytime behavioral activities of Nellore beef cattle in the rainy season. The experimental area was 12 hectares divided into paddocks of one hectare each. The treatments consisted of four defoliation heights (15, 30, 45 and 60 cm) in pastures of Brachiaria brizantha cv. Xaraes with three replicates each. It was used the continuos grazing method, with variable stocking rate. Forage samples collected on the plots were sent to the laboratory for separation of the botanical components, weighing and determination of dry matter, with the material collected by simulated grazing. The variables: grazing time, idle time and ruminating time were evaluated for 12 consecutive hours on days 15 and 16 February 2011, considering the morning and afternoon periods. It was used a completely randomized design. The height of the canopy significantly influenced the daily grazing time and ruminating time, with a quadratic response as a function of time of defoliation. The bite rate decreased as a function of heights studied. However the chemical composition of the material collected by simulated grazing did not differ between treatments. Xaraes grass swards grazed at around 45 cm height provide greater ease of apprehension by grazing cattle.

Sviluppo di un modulo VHDL per interfacciamento Ethernet-FPGA in applicazioni di high throughput screening

In molti settori della ricerca in campo biologico e biomedico si fa ricorso a tecniche di High Throughput Screening (HTS), tra cui studio dei canali ionici. In questo campo si studia la conduzione di ioni attraverso una membrana cellulare durante fenomeni che durano solo alcuni millisecondi. Allo scopo sono solitamente usati sensori e convertitori A/D ad elevata velocità insieme ad opportune interfacce di comunicazione, ad elevato bit-rate e latenza ridotta. In questa tesi viene descritta l'implementazione di un modulo VHDL per la trasmissione di dati digitali provenienti da un sistema HTS attraverso un controller di rete integrato dotato di un'interfaccia di tipo Ethernet, individuando le possibili ottimizzazioni specifiche per l'applicazione di interesse.

Fiber-Optic Technologies for Wireline and Wireless In-building Networks

This doctoral dissertation aims to establish fiber-optic technologies overcoming the limiting issues of data communications in indoor environments. Specific applications are broadband mobile distribution in different in-building scenarios and high-speed digital transmission over short-range wired optical systems. Two key enabling technologies are considered: Radio over Fiber (RoF) techniques over standard silica fibers for distributed antenna systems (DAS) and plastic optical fibers (POFs) for short-range communications. Hence, the objectives and achievements of this thesis are related to the application of RoF and POF technologies in different in-building scenarios. On one hand, a theoretical and experimental analysis combined with demonstration activities has been performed on cost-effective RoF systems. An extensive modeling on modal noise impact both on linear and non-linear characteristics of RoF link over silica multimode fiber has been performed to achieve link design rules for an optimum choice of the transmitter, receiver and launching technique. A successful transmission of Long Term Evolution (LTE) mobile signals on the resulting optimized RoF system over silica multimode fiber employing a Fabry-Perot LD, central launch technique and a photodiode with a built-in ball lens was demonstrated up to 525m with performances well compliant with standard requirements. On the other hand, digital signal processing techniques to overcome the bandwidth limitation of POF have been investigated. An uncoded net bit-rate of 5.15Gbit/s was obtained on a 50m long POF link employing an eye-safe transmitter, a silicon photodiode, and DMT modulation with bit and power loading algorithm. With the insertion of 3x2N quadrature amplitude modulation constellation formats, an uncoded net-bit-rate of 5.4Gbit/s was obtained on a 50 m long POF link employing an eye-safe transmitter and a silicon avalanche photodiode. Moreover, simultaneous transmission of baseband 2Gbit/s with DMT and 200Mbit/s with an ultra-wideband radio signal has been validated over a 50m long POF link.

Evaluation of new HTTP Adaptive Streaming algorithms: the clients’ and network’s perspectives

Lo streaming è una tecnica per trasferire contenuti multimediali sulla rete globale, utilizzato per esempio da servizi come YouTube e Netflix; dopo una breve attesa, durante la quale un buffer di sicurezza viene riempito, l'utente può usufruire del contenuto richiesto. Cisco e Sandvine, che con cadenza regolare pubblicano bollettini sullo stato di Internet, affermano che lo streaming video ha, e avrà sempre di più, un grande impatto sulla rete globale. Il buon design delle applicazioni di streaming riveste quindi un ruolo importante, sia per la soddisfazione degli utenti che per la stabilità dell'infrastruttura. HTTP Adaptive Streaming indica una famiglia di implementazioni volta a offrire la migliore qualità video possibile (in termini di bit rate) in funzione della bontà della connessione Internet dell'utente finale: il riproduttore multimediale può cambiare in ogni momento il bit rate, scegliendolo in un insieme predefinito, adattandosi alle condizioni della rete. Per ricavare informazioni sullo stato della connettività, due famiglie di metodi sono possibili: misurare la velocità di scaricamento dei precedenti trasferimenti (approccio rate-based), oppure, come recentemente proposto da Netflix, utilizzare l'occupazione del buffer come dato principale (buffer-based). In questo lavoro analizziamo algoritmi di adattamento delle due famiglie, con l'obiettivo di confrontarli su metriche riguardanti la soddisfazione degli utenti, l'utilizzo della rete e la competizione su un collo di bottiglia. I risultati dei nostri test non definiscono un chiaro vincitore, riconoscendo comunque la bontà della nuova proposta, ma evidenziando al contrario che gli algoritmi buffer-based non sempre riescono ad allocare in modo imparziale le risorse di rete.

An optimized QoS scheme for IMS-NEMO in heterogeneous networks

The network mobility (NEMO) is proposed to support the mobility management when users move as a whole. In IP Multimedia Subsystem (IMS), the individual Quality of Service (QoS) control for NEMO results in excessive signaling cost. On the other hand, current QoS schemes have two drawbacks: unawareness of the heterogeneous wireless environment and inefficient utilization of the reserved bandwidth. To solve these problems, we present a novel heterogeneous bandwidth sharing (HBS) scheme for QoS provision under IMS-based NEMO (IMS-NEMO). The HBS scheme selects the most suitable access network for each session and enables the new coming non-real-time sessions to share bandwidth with the Variable Bit Rate (VBR) coded media flows. The modeling and simulation results demonstrate that the HBS can satisfy users' QoS requirement and obtain a more efficient use of the scarce wireless bandwidth.

A low-cost alternative scheme to detect a 100 Gbps PM-DQPSK signal

We propose and demonstrate a low-cost alternative scheme of direct-detection to detect a 100Gbps polarization-multiplexed differential quadrature phase-shift keying (PM-DQPSK) signal. The proposed scheme is based on a delay line and a polarization rotator; the phase-shift keying signal is first converted into a polarization shift keying signal. Then, this signal is converted into an intensity modulated signal by a polarization beam splitter. Finally, the intensity-modulated signal is detected by balanced photodetectors. In order to demonstrate that our proposed receiver is suitable for using as a PM-DQPSK demodulator, a set of simulations have been performed. In addition to testing the sensitivity, the performance under various impairments, including narrow optical filtering, polarization mode dispersion, chromatic dispersion and polarization sensitivity, is analyzed. The simulation results show that our performance receiver is as good as a conventional receiver based on four delay interferometers. Moreover, in comparison with the typical receiver, fewer components are used in our receiver. Hence, implementation is easier, and total cost is reduced. In addition, our receiver can be easily improved to a bit-rate tunable receiver.

Logarithmical hopping encoding: a low computational complexity algorithm for image compression

LHE (logarithmical hopping encoding) is a computationally efficient image compression algorithm that exploits the Weber–Fechner law to encode the error between colour component predictions and the actual value of such components. More concretely, for each pixel, luminance and chrominance predictions are calculated as a function of the surrounding pixels and then the error between the predictions and the actual values are logarithmically quantised. The main advantage of LHE is that although it is capable of achieving a low-bit rate encoding with high quality results in terms of peak signal-to-noise ratio (PSNR) and image quality metrics with full-reference (FSIM) and non-reference (blind/referenceless image spatial quality evaluator), its time complexity is O( n) and its memory complexity is O(1). Furthermore, an enhanced version of the algorithm is proposed, where the output codes provided by the logarithmical quantiser are used in a pre-processing stage to estimate the perceptual relevance of the image blocks. This allows the algorithm to downsample the blocks with low perceptual relevance, thus improving the compression rate. The performance of LHE is especially remarkable when the bit per pixel rate is low, showing much better quality, in terms of PSNR and FSIM, than JPEG and slightly lower quality than JPEG-2000 but being more computationally efficient.

Research in speech communication.

Advances in digital speech processing are now supporting application and deployment of a variety of speech technologies for human/machine communication. In fact, new businesses are rapidly forming about these technologies. But these capabilities are of little use unless society can afford them. Happily, explosive advances in microelectronics over the past two decades have assured affordable access to this sophistication as well as to the underlying computing technology. The research challenges in speech processing remain in the traditionally identified areas of recognition, synthesis, and coding. These three areas have typically been addressed individually, often with significant isolation among the efforts. But they are all facets of the same fundamental issue--how to represent and quantify the information in the speech signal. This implies deeper understanding of the physics of speech production, the constraints that the conventions of language impose, and the mechanism for information processing in the auditory system. In ongoing research, therefore, we seek more accurate models of speech generation, better computational formulations of language, and realistic perceptual guides for speech processing--along with ways to coalesce the fundamental issues of recognition, synthesis, and coding. Successful solution will yield the long-sought dictation machine, high-quality synthesis from text, and the ultimate in low bit-rate transmission of speech. It will also open the door to language-translating telephony, where the synthetic foreign translation can be in the voice of the originating talker.

Improved Techniques for Video Compression and Communication

Thesis (Ph.D.)--University of Washington, 2016-06