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EN
Correct posture is a key element in the proper functioning of the entire body. Both defects and postural disorders lead to overload syndromes and degenerative changes in the musculoskeletal system. Different body positions correlate with respiratory parameters, which form the basis in modifying loudness and accentuation when speaking or singing Body posture can affect the quality of the voice signal and its fatigue. As movement and duration intensify, vocal effort increases. What is still open, however, is the problem of speech signal evaluation, especially in order to obtain assessments useful in the context of supporting medical diagnosis, optimizing therapy and monitoring rehabilitation. Meanwhile, such evaluations are what we need in medicine, rehabilitation and sports. This paper presents excerpts from a study of the effects of changes in posture and fatigue in healthy subjects, and those with phonation disorders, on changes in the acoustic parameters of the speech signal.
EN
In this paper a method based on the spectral contrast enhancement is proposed for improvement of the speech intelligibility of esophageal talkers, i.e., of people after surgical larynx operations (e.g., after the extraction of vocal cords). The proposed method is in our opinion one of the best approaches for the improvement of the esophageal speech quality via software and thus it can become an alternative for various mechanic-acoustic devices, which are - up to now - commonly used.
EN
The majority of hearing aid fitting systems are focused on improving speech understanding. A hearing aid fitting problem could be described as matching wide dynamic of speech signal to narrow dynamic of impaired hearing. To solve the problem the majority hearing aids use dynamic processors such as compressors and expanders. The engineered system is focused on obtaining desired hearing aid dynamic characteristics. The aim of performed experiments was to test the developed multimedia computer system and check whether and to what degree the implemented virtual hearing aid algorithm can improve speech intelligibility. A group of hearing impaired patients who have problems with perception of speech in noise was examined. The short overview of carried out tests and the examination results are presented in the paper.
EN
The effect of spectral ehancement in the ear has not yet been fully understood and documented, but it is an important mechanism which seems to contribute to the ear's capability of comprehending speech despite a high level of accompanying background noise. Three algorithms of spectral enhancement have been developed and tested in this work. These algorithms are not based on any physiological orpsychophysical data documenting the spectral enhancement in the ear, as they are hardly available, but they are based on arbitrary assumptions of how this mechanism may affect the time-frequency spectra of the acoustic signals. At the first stage of the simulations, input signals were analysed by the Multiple Trigonometric Bases Transform developed by this author. That analysis procedure approximates the time - frequency resolution of the ear. The main aim of the simulations proposed is to develop a tool for extracting most relevant information out of the audio signal. The examples of graphical presentations of the results of simulations and some flowcharts of the algorithms are presented.
EN
The Digital Speech Aid (DSA), constructed at the Sound and Vision Engineering Department, Gdansk University of Technology, is a device improving speech fluency of stuttering people. It uses a new method for the modification of the auditory feedback loop. In 2002 the Foundation for Polish Science supported clinical evaluation of the DSA. In this way, 100 devices are currently under examination in various areas of Poland. Owing to that, comprehensive assessments of the efficacy of the therapy using DSA can be carried out. The basic information about stuttering and the DSA are presented in this paper. The description of therapy methods is given. The paper placed emphasis on issues related to the assessments of the therapy progress using questionnaires and so called "syllable test". Furthermore, some algorithms applicable to the automatic detection of speech disfluency are discussed.
EN
This paper presents and compares two algorithms used for blind separation of signals mixed with delays (the convolved signals). The aim consists in separation of source signals using only their convolved mixtures. These algorithms are based on the entropy maximization principle. They are implemented in both: time and frequency domain. The time domain offers simpler signal processing but the general performance is worse than in the case of the frequency domain separation. Presentation of these algorithms includes results of the computer simulations.
EN
This paper describes the measurements system of assessments of the quality of speech in rooms. It includes an overview of traditional method of subjective measurements of logatom intelligibility and modified intelligibility test with forced choice. Experiments in which the methods were used are described.
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