Análise de desempenho dos codecs suportados pelo Asterisk com e sem criptografia

The main objective is to analyze the performance of some codecs supported by Asterisk with and without encryption using RTP and SRTP, respectively, providing important data for decision-making in the implementation of a VoIP system with Asterisk. Thus, it is possible to realize both codecs as the protocol can be chosen depending on the application, or the system's main feature is the speed packet switching, security level or lower tolerance for unsuccessful calls. For this, tests were made with the codec with and without the use of cryptography to obtain some findings on the use of the same, giving more attention to the response time for the start of a call.

A Novel Scalable Deblocking-Filter Architecture for H.264/AVC and SVC Video Codecs

A highly parallel and scalable Deblocking Filter (DF) hardware architecture for H.264/AVC and SVC video codecs is presented in this paper. The proposed architecture mainly consists on a coarse grain systolic array obtained by replicating a unique and homogeneous Functional Unit (FU), in which a whole Deblocking-Filter unit is implemented. The proposal is also based on a novel macroblock-level parallelization strategy of the filtering algorithm which improves the final performance by exploiting specific data dependences. This way communication overhead is reduced and a more intensive parallelism in comparison with the existing state-of-the-art solutions is obtained. Furthermore, the architecture is completely flexible, since the level of parallelism can be changed, according to the application requirements. The design has been implemented in a Virtex-5 FPGA, and it allows filtering 4CIF (704 × 576 pixels @30 fps) video sequences in real-time at frequencies lower than 10.16 Mhz.

Run-time Scalable Architecture for Deblocking Filtering in H.264/AVC-SVC Video Codecs

Systems relying on fixed hardware components with a static level of parallelism can suffer from an underuse of logical resources, since they have to be designed for the worst-case scenario. This problem is especially important in video applications due to the emergence of new flexible standards, like Scalable Video Coding (SVC), which offer several levels of scalability. In this paper, Dynamic and Partial Reconfiguration (DPR) of modern FPGAs is used to achieve run-time variable parallelism, by using scalable architectures where the size can be adapted at run-time. Based on this proposal, a scalable Deblocking Filter core (DF), compliant with the H.264/AVC and SVC standards has been designed. This scalable DF allows run-time addition or removal of computational units working in parallel. Scalability is offered together with a scalable parallelization strategy at the macroblock (MB) level, such that when the size of the architecture changes, MB filtering order is modified accordingly

Impact of region-of-interest video coding on perceived quality in mobile video

Effective streaming of video can be achieved by providing more bits to the most important region in the frame at the cost of reduced bits in the less important regions. This strategy can be beneficial for delivering high quality videos in mobile devices, especially when the availability of bandwidth is usually low and limited. While the state-of-the-art video codecs such as H.264 may have been optimised for perceived quality, it is hypothesised that users will give more attention to interesting region/object when watching videos. Therefore, giving a higher quality to region of interest (ROI)while reducing quality of other areas may result in improving the overall perceived quality without necessarily increasing the bitrate. In this paper, the impact of ROI-based encoded video on perceived quality is investigated by conducting a user study for varous target bitrates. The results from the user study demonstrate that ROI-based video coding has superior perceived quality compared to normal encoded video at the same bitrate in the lower bitrate range.

QoE modelling for VP9 and H.265 videos on mobile devices

Current mobile devices and streaming video services support high definition (HD) video, increasing expectation for more contents. HD video streaming generally requires large bandwidth, exerting pressures on existing networks. New generation of video compression codecs, such as VP9 and H.265/HEVC, are expected to be more effective for reducing bandwidth. Existing studies to measure the impact of its compression on users’ perceived quality have not been focused on mobile devices. Here we propose new Quality of Experience (QoE) models that consider both subjective and objective assessments of mobile video quality. We introduce novel predictors, such as the correlations between video resolution and size of coding unit, and achieve a high goodness-of-fit to the collected subjective assessment data (adjusted R-square >83%). The performance analysis shows that H.265 can potentially achieve 44% to 59% bit rate saving compared to H.264/AVC, slightly better than VP9 at 33% to 53%, depending on video content and resolution.

Analytical models for capacity estimation of IEEE 802.11 WLANs using DCF for internet applications

We provide analytical models for capacity evaluation of an infrastructure IEEE 802.11 based network carrying TCP controlled file downloads or full-duplex packet telephone calls. In each case the analytical models utilize the attempt probabilities from a well known fixed-point based saturation analysis. For TCP controlled file downloads, following Bruno et al. (In Networking '04, LNCS 2042, pp. 626-637), we model the number of wireless stations (STAs) with ACKs as a Markov renewal process embedded at packet success instants. In our work, analysis of the evolution between the embedded instants is done by using saturation analysis to provide state dependent attempt probabilities. We show that in spite of its simplicity, our model works well, by comparing various simulated quantities, such as collision probability, with values predicted from our model. Next we consider N constant bit rate VoIP calls terminating at N STAs. We model the number of STAs that have an up-link voice packet as a Markov renewal process embedded at so called channel slot boundaries. Analysis of the evolution over a channel slot is done using saturation analysis as before. We find that again the AP is the bottleneck, and the system can support (in the sense of a bound on the probability of delay exceeding a given value) a number of calls less than that at which the arrival rate into the AP exceeds the average service rate applied to the AP. Finally, we extend the analytical model for VoIP calls to determine the call capacity of an 802.11b WLAN in a situation where VoIP calls originate from two different types of coders. We consider N-1 calls originating from Type 1 codecs and N-2 calls originating from Type 2 codecs. For G711 and G729 voice coders, we show that the analytical model again provides accurate results in comparison with simulations.

An optimal transform Architecture for H.264/AVC

This paper presents the design of the area optimized integer two dimensional discrete cosine transform (2-D DCT) used in H.264/AVC codecs. The 2-D DCT calculation is performed by utilizing the separability property, in such a way that 2-D DCT is divided into two 1-D DCT calculation that are joined through a common memory. Due to its area optimized approach, the design will find application in mobile devices. Verilog hardware description language (HDL) in cadence environment has been used for design, compilation, simulation and synthesis of transform block in 0.18 mu TSMC technology.

基于LAN的语音通讯软件设计

文章介绍了网络语音通信的原理、关键技术和实现方法,详细叙述了基于LAN的语音通信软件的开发过程,给出了相应的实例。该软件在音质和时延上都达到了IP电话的效果。

Rapid design of DSP ASIC cores using hierarchical VHDL libraries

Methods are presented for the rapid design of DSP ASICs based on the use of a series of hierarchical VHDL libraries which are portable across many silicon foundries. These allows complex DSP silicon systems to be developed in a small fraction of the time normally required. Resulting designs are highly competitive with those developed using more conventional methods. The approach is illustrated using several examples. These include ADPCM codecs, as well as DCT and FFT cores.

A Steganalysis System Utilizing Temporal Pixel Correlation Of HEVC Video

High Efficiency Video Coding (HEVC) is the most recent video codec coming after currently most popular H.264/MPEG4 codecs and has promising compression capabilities. It is conjectured that it will be a substitute for current video compression standards. However, to the best knowledge of the authors, none of the current video steganalysis methods designed or tested with HEVC video. In this paper, pixel domain steganography applied on HEVC video is targeted for the first time. Also, its the first paper that employs accordion unfolding transformation, which merges temporal and spatial correlation, in pixel domain video steganalysis. With help of the transformation, temporal correlation is incorporated into the system. Its demonstrated for three different feature sets that integrating temporal dependency substantially increased the detection accuracy.

Improved voice activity detection algorithm using wavelet and support vector machine

This paper proposes an improved voice activity detection (VAD) algorithm using wavelet and support vector machine (SVM) for European Telecommunication Standards Institution (ETS1) adaptive multi-rate (AMR) narrow-band (NB) and wide-band (WB) speech codecs. First, based on the wavelet transform, the original IIR filter bank and pitch/tone detector are implemented, respectively, via the wavelet filter bank and the wavelet-based pitch/tone detection algorithm. The wavelet filter bank can divide input speech signal into several frequency bands so that the signal power level at each sub-band can be calculated. In addition, the background noise level can be estimated in each sub-band by using the wavelet de-noising method. The wavelet filter bank is also derived to detect correlated complex signals like music. Then the proposed algorithm can apply SVM to train an optimized non-linear VAD decision rule involving the sub-band power, noise level, pitch period, tone flag, and complex signals warning flag of input speech signals. By the use of the trained SVM, the proposed VAD algorithm can produce more accurate detection results. Various experimental results carried out from the Aurora speech database with different noise conditions show that the proposed algorithm gives considerable VAD performances superior to the AMR-NB VAD Options 1 and 2, and AMR-WB VAD. (C) 2009 Elsevier Ltd. All rights reserved.

MoCHA : arquitetura dedicada para a compensação de movimento em decodificadores de vídeo de alta definição, seguindo o padrão H.264

O padrão H.264 foi desenvolvido pelo JVT, que foi formado a partir de uma união entre os especialistas do VCEG da ITU-T e do MPEG da ISO/IEC. O padrão H.264 atingiu seu objetivo de alcançar as mais elevadas taxas de processamento dentre todos os padrões existentes, mas à custa de um grande aumento na complexidade computacional. Este aumento de complexidade impede, pelo menos na tecnologia atual, a utilização de codecs H.264 implementados em software, quando se deseja a decodi cação de vídeos de alta de nição em tempo real. Essa dissertação propõe uma solução arquitetural de hardware, denominada MoCHA, para compensação de movimento do decodi cador de vídeo de alta de nição, segundo o padrão H.264/AVC. A MoCHA está dividida em três blocos principais, a predição dos vetores de movimento, o acesso à memória e o processamento de amostras. A utilização de uma cache para explorar a redundância dos dados nos acessos à mem ória, em conjunto com melhorias propostas, alcançou economia de acessos à memória superior a 60%, para os casos testados. Quando uma penalidade de um ciclo por troca de linha de memória é imposta, a economia de ciclos de acesso supera os 75%. No processamento de amostras, a arquitetura realiza o processamento dos dois blocos, que dão origem ao bloco bi-preditivo, de forma serial. Dessa forma, são economizados recursos de hardware, uma vez que a duplicação da estrutura de processamento não é requerida. A arquitetura foi validada a partir de simulações, utilizando entradas extraídas de seqüências codi cadas. Os dados extraídos, salvos em arquivos, serviam de entrada para a simulação. Os resultados da simulação foram salvos em arquivos e comparados com os resultados extraídos. O processador de amostras do compensador de movimento foi prototipado na placa XUP Virtex-II Pro. A placa possui um FPGA VP30 da família Virtex-II PRO da Xilinx. O processador PowerPC 405, presente no dispositivo, foi usado para implementar um test bench para validar a operação do processador de amostras mapeado para o FPGA. O compensador de movimento para o decodi cador de vídeo H.264 foi descrito em VHDL, num total de 30 arquivos e cerca de 13.500 linhas de código. A descrição foi sintetizada pelo sintetizador Syplify Pro da Symplicity para o dispositivo XC2VP30-7 da Xilinx, consumindo 8.465 slices, 5.671 registradores, 10.835 LUTs, 21 blocos de memó- ria interna e 12 multiplicadores. A latência mínima para processar um macrobloco é de 233 ciclos, enquanto a máxima é de 590, sem considerar misses na cache. A freqüência máxima de operação foi de 100,5 MHz. A arquitetura projetada é capaz de processar, no pior caso, 36,7 quadros HDTV de 1080 por 1920, inteiramente bi-preditivos, por segundo. Para quadros do tipo P, que não utilizam a bi-predição, a capacidade de processamento sobe para 64,3 quadros por segundo. A arquitetura apresentada para o processamento de quadros bi-preditivos e a hierarquia de memória são, até o momento, inéditas na literatura. Os trabalhos relativos a decodi cadores completos não apresentam a solução para esse processamento. Os resultados apresentados tornam a MoCHA uma solução arquitetural capaz de fazer parte de um decodi cador para vídeos de alta definição.

Instalação de uma rede mesh metropolitana utilizando o padrão IEEE 802.11a e implementação do serviço Voip (Wman-Voip)

Pós-graduação em Engenharia Elétrica - FEIS

Quality of experience in skype video calls

Skype is one of the well-known applications that has guided the evolution of real-time video streaming and has become one of the most used software in everyday life. It provides VoIP audio/video calls as well as messaging chat and file transfer. Many versions are available covering all the principal operating systems like Windows, Macintosh and Linux but also mobile systems. Voice quality decreed Skype success since its birth in 2003 and peer-to-peer architecture has allowed worldwide diffusion. After video call introduction in 2006 Skype became a complete solution to communicate between two or more people. As a primarily video conferencing application, Skype assumes certain characteristics of the delivered video to optimize its perceived quality. However in the last years, and with the recent release of SkypeKit1, many new Skype video-enabled devices came out especially in the mobile world. This forced a change to the traditional recording, streaming and receiving settings allowing for a wide range of network and content dynamics. Video calls are not anymore based on static ‘chatting’ but mobile devices have opened new possibilities and can be used in several scenarios. For instance, lecture streaming or one-to-one mobile video conferences exhibit more dynamics as both caller and callee might be on move. Most of these cases are different from “head&shoulder” only content. Therefore, Skype needs to optimize its video streaming engine to cover more video types. Heterogeneous connections require different behaviors and solutions and Skype must face with this variety to maintain a certain quality independently from connection used. Part of the present work will be focused on analyzing Skype behavior depending on video content. Since Skype protocol is proprietary most of the studies so far have tried to characterize its traffic and to reverse engineer its protocol. However, questions related to the behavior of Skype, especially on quality as perceived by users, remain unanswered. We will study Skype video codecs capabilities and video quality assessment. Another motivation of our work is the design of a mechanism that estimates the perceived cost of network conditions on Skype video delivery. To this extent we will try to assess in an objective way the impact of network impairments on the perceived quality of a Skype video call. Traditional video streaming schemes lack the necessary flexibility and adaptivity that Skype tries to achieve at the edge of a network. Our contribution will lye on a testbed and consequent objective video quality analysis that we will carry out on input videos. We will stream raw video files with Skype via an impaired channel and then we will record it at the receiver side to analyze with objective quality of experience metrics.