Gavril Toderean

Network performance evaluation for RIP, OSPF and EIGRP routing protocols

The main function of a network layer is to route packets from the source machine to the destination machine. Algorithms that are used for route selection and data structure are the main parts for the network layer. In this paper we examine the network performance when using three routing protocols, RIP, OSPF and EIGRP. Video, HTTP and Voice application where configured for network transfer. We also examine the behaviour when using link failure/recovery controller between network nodes. The simulation results are analyzed, with a comparison between these protocols on the effectiveness and performance in network implemented.

A rule-based approach to build a text-to-speech system for Romanian

We present in this article our approach for building a text-to-speech system for Romanian. Main stages of this work were: voice signal analysis, region segmentation, construction of acoustic database, text analysis, unit and prosody detection, unit matching, concatenation and speech synthesis. In our approach we consider word syllables as basic units and stress indicating intrasegmental prosody. A special characteristic of current approach is rule-based processing of both speech signal analyse and text analyse stages.

Analysis and Synthesis of Vowels Using Matlab

We propose a software environment in Matlab, in order to extract the main features from the Romanian vowels and to synthesize the vowels. The used analysis techniques for the estimation of the parameters are: time domain analysis for energy and zero-crossing rate (ZCR), cepstral analysis for fundamental frequency and linear predictive coding (LPC) for formants. Also, to synthesize the vowels, we used the LPC method

Region-based fractal image compression using deterministic search

The paper introduces a new method based on deterministic search to fractal image compression. In order to find a good region-based partitioning, we propose a deterministic search method for finding the blocks to be merged. For each range, a list of best N domains is maintained. When two ranges are to be merged and their common edge disappears, for the new range the best N domains are selected only from the 2/spl times/N domain extension of the two ranges. At each step the edge with minimum collage error increase is deterministically selected and the two corresponding ranges are merged. The process starts with atomic blocks as ranges and ends when the desired number of ranges is achieved. In order to reduce the encoding time, a suboptimal initialization method is also considered. Experimental results prove that our method yields a better rate-distortion curve than the classic quadtree partitioning scheme.

Voice Signal Processing For Speech Synthesis

We present here a software application capable to manipulate and analyse speech signal, extract characteristic parameters needed for speech synthesis and to enhance the speech quality. We also present main signal parameters used in speech synthesis, the facilities of realized application and the experimental results obtained

Agent based dbist dbisr and its web wireless management

This paper presents an attempt of using intelligent agents for testing and repairing a distributed system, whose elements may or may not have embedded BIST (built-in self-test) and BISR (built-in self-repair) facilities. It also discusses a few possible solutions for remotely managing the agent society, including the devices, agents, tests and repair processes. The agents are software modules that perform monitoring, diagnosis and repair of the faults. They form together a society whose members communicate, set goals and solve tasks. We tried multiple agent platforms, including Java 2 Micro Edition, which ensures portability, flexibility, but also a relatively small memory footprint. Presented remote management alternatives for the agent society include Web/PHP, WAP/PHP, Java 2 Micro Edition and embedded Web servers.

Merging the transform step and the quantization step for Karhunen-Loeve transform based image compression

Transform coding is one of the most important methods for lossy image compression. The optimum linear transform - known as Karhunen-Loeve transform (KLT) - was difficult to implement in the classic way. Now, due to continuous improvements in neural networks performance, the KLT method becomes more topical then ever. We propose a new scheme where the quantization step is merged together with the transform step during the learning phase. The new method is tested for different levels of quantization and for different types of quantizers. Experimental results presented in the paper prove that the new proposed scheme always gives better results than the state-of-the-art solution.

Romanian phonetic transcription dictionary for speeding up language technology development

This paper intends to present a machine readable Romanian language pronunciation dictionary called NaviRo. The dictionary contains 138,500 unique words from the DexOnline dictionary together with their phonetic transcriptions in speech assessment method phonetic alphabet. The development of the pronunciation dictionary and the performed validation tests are also described in the paper. NaviRo pronunciation dictionary is freely available on the project website (http://users.utcluj.ro/~jdomokos/naviro) in plain text, Hidden Markov Model Toolkit and Festival speech synthesis system dictionary format. There are also available for download the used grapheme and phoneme sets and the audio samples for the used phonemes. The use of these resources is completely unrestricted for any research purposes in order to speed up Romanian language speech technology research.

Experiments with the prediction and generation of Romanian intonation

During the development of a text-to-speech synthesis system, prosody prediction and generation, is one of the most critical phases. Prosody is composed by the melody, intensity and the speed of the speech. The current paper has in focus the intonation of Romanian language, its prediction and generation within a TTS system. During the development of different prosodic modules, several experimental results have been achieved, which are presented in this work. The process is followed starting from text pre-processing, the determination of sentence types and subtypes, through automatic labeling of sentences, till the labeling and transposing of the intonation curves. The creation and storage of relative intonation curves, as the developed different XML rule systems are also topic of this work.

An Improved Method for Automatic Classification of Speech

In this paper we present a novel method for silence/unvoiced/voiced (SUV) classification of speech signals. The well-known algorithm for locating endpoints in an utterance, based on zero-crossing rate and energy, was our starting point. We added a few supplementary decision criteria to it and we tested it using features like Teager energy and entropy. The experiments that we performed showed that these features performed better than the traditional energy measure for clean speech, but none of them produced a significant improvement in a noisy environment