A configurable vector processor for accelerating speech coding algorithms

The growing demand for voice-over-packer (VoIP) services and multimedia-rich applications has made increasingly important the efficient, real-time implementation of low-bit rates speech coders on embedded VLSI platforms. Such speech coders are designed to substantially reduce the bandwidth requirements thus enabling dense multichannel gateways in small form factor. This however comes at a high computational cost which mandates the use of very high performance embedded processors. This thesis investigates the potential acceleration of two major ITU-T speech coding algorithms, namely G.729A and G.723.1, through their efficient implementation on a configurable extensible vector embedded CPU architecture. New scalar and vector ISAs were introduced which resulted in up to 80% reduction in the dynamic instruction count of both workloads. These instructions were subsequently encapsulated into a parametric, hybrid SISD (scalar processor)–SIMD (vector) processor. This work presents the research and implementation of the vector datapath of this vector coprocessor which is tightly-coupled to a Sparc-V8 compliant CPU, the optimization and simulation methodologies employed and the use of Electronic System Level (ESL) techniques to rapidly design SIMD datapaths

Media gateway utilizando um GPU

Mestrado em Engenharia de Computadores e Telemátic

Voice over IP

The area that this thesis covers is Voice over IP (or IP Telephony as it is sometimes called) over Private networks and not over the Internet. There is a distinction to be made between the two even though the term is loosely applied to both. IP Telephony over Private Networks involve calls made over private WANs using IP telephony protocols while IP Telephony over the Internet involve calls made over the public Internet using IP telephony protocols. Since the network is private, service is reliable because the network owner can control how resources are allocated to various applications, such as telephony services. The public Internet on the other hand is a public, largely unmanaged network that offers no reliable service guarantee. Calls placed over the Internet can be low in quality, but given the low price, some find this solution attractive. What started off as an Internet Revolution with free phone calls being offered to the general public using their multimedia computers has turned into a telecommunication revolution where enterprises are beginning to converge their data and voice networks into one network. In retrospect, an enterprise\u27s data networks are being leveraged for telephony. The communication industry has come full circle. Earlier in the decade data was being transmitted over the public voice networks and now voice is just another application which is/will be run over the enterprises existing data networks. We shall see in this thesis the problems that are encountered while sending Voice over Data networks using the underlying IP Protocol and the corrective steps taken by the Industry to resolve these multitudes of issues. Paul M. Zam who is collaborating in this Joint Thesis/project on VoIP will substantiate this theoretical research with his practical findings. On reading this paper the reader will gain an insight in the issues revolving the implementation of VoIP in an enterprises private network as well the technical data, which sheds more light on the same. Thus the premise of this joint thesis/project is to analyze the current status of the technology and present a business case scenario where an organization will be able to use this information

Multi-core platforms for audio and multimedia coding algorithms in telecommunications

Tietoliikenteessä käytettävät multimedian koodausalgoritmit eli koodekit kehittyvät jatkuvasti. USAC ja Opus ovat uusia, sekä puheelle että musiikille soveltuvia audiokoodekkeja. Molemmat ovat sijoittuneet korkealle koodekkien äänenlaatua vertailevissa tutkimuksissa. Näiden keskeisiä ominaisuuksia käsitellään kirjallisuuskatsaukseen perustuen. Varsinkin HD-tasoisen videon käsittelyssä käytettävät koodekit vaativat suurta laskentatehoa. Tilera TILEPro64 -moniydinsuorittimen ja sille optimoitujen multimediakoodekkien suorituskykyä testattiin tarkoitukseen suunnitelluilla tietokoneohjelmilla. Tulokset osoittivat, että suoritinytimiä lisättäessä videon koodausalgoritmien suoritusnopeus kasvaa tiettyyn rajaan asti. Testatuilla äänen koodausalgoritmeillä ytimien lisääminen ei parantanut suoritusnopeutta. Tileran moniydinratkaisuja verrattiin lopuksi Freescalen ja Texas Instrumentsin moniydinratkaisuihin. Huolimatta eroista laitteistoarkkitehtuureissa, kyseisten toimittajien kehitystyökaluissa todettiin olevan paljon samoja piirteitä.Multimedia coding algorithms used in telecommunications evolve constantly. Benefits and properties of two new hybrid audio codecs (USAC, Opus) were reviewed on a high level as a literature study. It was found that both have succeeded well in subjective sound quality measurements. Tilera TILEPro64-multicore platform and a related software library was evaluated in terms of performance in multimedia coding. The performance in video coding was found to increase with the number of processing cores up to a certain point. With the tested audio codecs, increasing the number of cores did not increase coding performance. Additionally, multicore products of Tilera, Texas Instruments and Freescale were compared. Development tools of all three vendors were found to have similar features, despite the differences in hardware architectures

Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

Quality of Service Controlled Multimedia Transport Protocol

PhDThis research looks at the design of an open transport protocol that supports a range of services including multimedia over low data-rate networks. Low data-rate multimedia applications require a system that provides quality of service (QoS) assurance and flexibility. One promising field is the area of content-based coding. Content-based systems use an array of protocols to select the optimum set of coding algorithms. A content-based transport protocol integrates a content-based application to a transmission network. General transport protocols form a bottleneck in low data-rate multimedia communicationbsy limiting throughpuot r by not maintainingt iming requirementsT. his work presents an original model of a transport protocol that eliminates the bottleneck by introducing a flexible yet efficient algorithm that uses an open approach to flexibility and holistic architectureto promoteQ oS.T he flexibility andt ransparenccyo mesi n the form of a fixed syntaxt hat providesa seto f transportp rotocols emanticsT. he mediaQ oSi s maintained by defining a generic descriptor. Overall, the structure of the protocol is based on a single adaptablea lgorithm that supportsa pplication independencen, etwork independencea nd quality of service. The transportp rotocol was evaluatedth rougha set of assessmentos:f f-line; off-line for a specific application; and on-line for a specific application. Application contexts used MPEG-4 test material where the on-line assessmenuts eda modified MPEG-4 pl; yer. The performanceo f the QoSc ontrolledt ransportp rotocoli s often bettert hano thers chemews hen appropriateQ oS controlledm anagemenatl gorithmsa re selectedT. his is shownf irst for an off-line assessmenwt here the performancei s compared between the QoS controlled multiplexer,a n emulatedM PEG-4F lexMux multiplexers chemea, ndt he targetr equirements. The performanceis also shownt o be better in a real environmentw hen the QoS controlled multiplexeri s comparedw ith the real MPEG-4F lexMux scheme

Proceedings of the Mobile Satellite Conference

A satellite-based mobile communications system provides voice and data communications to mobile users over a vast geographic area. The technical and service characteristics of mobile satellite systems (MSSs) are presented and form an in-depth view of the current MSS status at the system and subsystem levels. Major emphasis is placed on developments, current and future, in the following critical MSS technology areas: vehicle antennas, networking, modulation and coding, speech compression, channel characterization, space segment technology and MSS experiments. Also, the mobile satellite communications needs of government agencies are addressed, as is the MSS potential to fulfill them

Study of voice quality in IP networks

Orientador: Helio WaldmanDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de ComputaçãoAbstract: This work describes the study of voice quality in IP networks based on a revision of quality of service (QoS) protocols and mechanisms and aspects of the system impact associated with the presence or absence of them; revision of the diverse evaluation methods of voice quality with emphasis in the automatic methods (objective and repetitive) used to analyze the effects in the voice due to diverse factors presented in packet networks, such as packet loss, delay and jitter, as well as the proper voice coding at low rate; revision of the main protocols of signalling for implementation of voice over IP (VoIP) or IP telephony, considering its strong and weak points with regard to implementation facility, scalability and adequacy for some network applications and quality of service (QoS) and accomplishment of tests in simulated and experimental IP networks. The main objective is the characterization of voice service in IP networks taking into account the effect of the network factors in call set-up (connection) and in voice quality . For the simulation of the IP network the Shunra¿s Cloud software was used where it is possible to deal with, in isolated form, the influence of packet loss, fixed delay, delay variation ( jitter), as well as the composed effect of packet loss and jitter. Solutions to such effects are investigated in experimental tests. Results of system simulations are presented and discussed. Degradations in voice due to such effects are evaluated and a practical method to solve them is considered. The experimental results demonstrate the technical feasibility of using voice over IP (or IP telephony) by service providers, as well as private corporations being able to forward voice and data over the same converged IP networkResumo: Este trabalho descreve o estudo da qualidade de voz em redes IP a partir de uma revisão dos protocolos e mecanismos relativos a qualidade de serviço (QoS) e os aspectos do impacto sistêmico na presença ou ausência destes; revisão dos diversos métodos de avaliação da qualidade da voz com ênfase nos métodos automáticos (objetivos e repetitivos) para auxiliar na análise dos efeitos na voz dos diversos fatores presentes em uma rede de pacotes, tais como perda de pacote, atraso e jitter, bem como a própria codificação da voz em baixas taxas; revisão dos principais protocolos de sinalização utilizados para implementação de voz sobre IP (VoIP) ou telefonia sobre IP, evidenciando-se seus pontos fortes e fracos com relação a facilidade de implementação, extensibilidade e adequabilidade para várias aplicações de rede e qualidade de serviço (QoS) e realização de testes em redes IP simulada e experimental. O principal objetivo é a caracterização do serviço de voz em redes IP levando-se em consideração os efeitos dos fatores de rede e gateway no tempo de estabelecimento de uma chamada (conexão) e na qualidade da voz. Para simulação da rede IP foi utilizado o software Cloud da Shunra onde é possível tratar, de forma isolada, a influência da perda de pacote, do atraso fixo, do atraso variável (jitter), bem como do efeito conjunto da perda de pacote e jitter. Soluções a tais efeitos são investigadas em testes experimentais. Resultados de simulações sistêmicas são apresentados e discutidos. As degradações na voz devidas a tais efeitos são avaliadas e um método prático para solucionar é testado. Os resultados experimentais demonstram a viabilidade técnica da utilização da voz sobre IP (ou telefonia IP) pelos provedores de serviço, bem como pelas corporações privadas, podendo trafegar voz e dados em uma mesma rede convergente IPMestradoTelecomunicações e TelemáticaMestre em Engenharia Elétric

Proceedings of the Fifth International Mobile Satellite Conference 1997

Satellite-based mobile communications systems provide voice and data communications to users over a vast geographic area. The users may communicate via mobile or hand-held terminals, which may also provide access to terrestrial communications services. While previous International Mobile Satellite Conferences have concentrated on technical advances and the increasing worldwide commercial activities, this conference focuses on the next generation of mobile satellite services. The approximately 80 papers included here cover sessions in the following areas: networking and protocols; code division multiple access technologies; demand, economics and technology issues; current and planned systems; propagation; terminal technology; modulation and coding advances; spacecraft technology; advanced systems; and applications and experiments

Radio Communications

In the last decades the restless evolution of information and communication technologies (ICT) brought to a deep transformation of our habits. The growth of the Internet and the advances in hardware and software implementations modified our way to communicate and to share information. In this book, an overview of the major issues faced today by researchers in the field of radio communications is given through 35 high quality chapters written by specialists working in universities and research centers all over the world. Various aspects will be deeply discussed: channel modeling, beamforming, multiple antennas, cooperative networks, opportunistic scheduling, advanced admission control, handover management, systems performance assessment, routing issues in mobility conditions, localization, web security. Advanced techniques for the radio resource management will be discussed both in single and multiple radio technologies; either in infrastructure, mesh or ad hoc networks