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Tytuł:
Phoneme Segmentation Based on Wavelet Spectra Analysis
Autorzy:
Ziółko, B.
Manandhar, S.
Wilson, R. C.
Ziółko, M.
Powiązania:
https://bibliotekanauki.pl/articles/177480.pdf
Data publikacji:
2011
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
speech recognition
speech segmentation
discrete wavelet transform
Opis:
A phoneme segmentation method based on the analysis of discrete wavelet transform spectra is described. The localization of phoneme boundaries is particularly useful in speech recognition. It enables one to use more accurate acoustic models since the length of phonemes provide more information for parametrization. Our method relies on the values of power envelopes and their first derivatives for six frequency subbands. Specific scenarios that are typical for phoneme boundaries are searched for. Discrete times with such events are noted and graded using a distribution-like event function, which represent the change of the energy distribution in the frequency domain. The exact definition of this method is described in the paper. The final decision on localization of boundaries is taken by analysis of the event function. Boundaries are, therefore, extracted using information from all subbands. The method was developed on a small set of Polish hand segmented words and tested on another large corpus containing 16 425 utterances. A recall and precision measure specifically designed to measure the quality of speech segmentation was adapted by using fuzzy sets. From this, results with F-score equal to 72.49% were obtained.
Źródło:
Archives of Acoustics; 2011, 36, 1; 29-47
0137-5075
Pojawia się w:
Archives of Acoustics
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Obiektywne dopasowanie aparatu słuchowego przy użyciu systemu SpeechPro i urządzenia Avant REM Speech+
Objective fitting of hearing aid with SpeechPro and Avant REM Speech+
Autorzy:
Swinarska, D.
Swinarski, R.
Powiązania:
https://bibliotekanauki.pl/articles/270908.pdf
Data publikacji:
2017
Wydawca:
Centralny Ośrodek Badawczo-Rozwojowy Aparatury Badawczej i Dydaktycznej, COBRABiD
Tematy:
Avant REM Speech+
SpeechPro
aparat słuchowy
niedosłuch
hearing aid
deafness
Opis:
Osoby niedosłyszące bardzo często borykają się z problemem nieodpowiednio dobranego aparatu słuchowego. Najczęstszym problemem wśród osób noszących pomoce słuchowe jest niewłaściwe rozumienie mowy w różnych warunkach akustycznych. Obecnie aparaty ustawiane są za pomocą specjalistycznych programów i urządzeń na podstawie subiektywnych odczuć pacjenta. Przedstawiona w niniejszym artykule metoda dopasowania aparatów słuchowych przy użyciu systemu SpeechPro i urządzenia Avant REM Speech+ pozwala na obiektywne dopasowanie aparatu słuchowego każdej osobie niedosłyszącej bez jej czynnego udziału. Wyniki i dyskusja: Urządzenie Avant REM Speech+ całkowicie spełnia swoje zadanie – dopasowuje aparat słuchowy do danego ubytku słuchu bez ingerencji protetyka słuchu oraz subiektywnych odczuć pacjenta. Ustalenie natężenia dźwięku na zewnątrz i wewnątrz ucha pacjenta pozwala protetykowi słuchu dobrać rzeczywiste wzmocnienie niezbędne do danego niedosłuchu. Urządzenie jest bardzo dokładne, zakres błędu wynosi ±/1 dB. Wykres audiogramu oraz charakterystyka dopasowania są zbliżone do siebie, co odpowiada bardzo dokładnemu uzyskaniu skutecznego dopasowania aparatu słuchowego.
Deafness is civilization disease, which affects, not only older people, but also young people and children. The most common problem among hearing aid users is improper speech understanding in various acoustic conditions. The method described in following article is about hearing aid fitting, using SpeechPro and Avant REM Speech+ device allowing objective fitting of hearing aids to any hearing impaired person without active participation. Results and discussion: The Avant REM Speech+ device completely fulfills its purpose – it fits hearing system to the specific hearing loss without intervention of the hearing care professional and the subjective feelings of the patient. The chart of audiogram and fit characteristics are close to each other, which responds to very accurate attainment of effective fit of the hearing instrument.
Źródło:
Aparatura Badawcza i Dydaktyczna; 2017, 22, 3; 215-223
2392-1765
Pojawia się w:
Aparatura Badawcza i Dydaktyczna
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Visualization of stages of determining cepstral factors in speech recognition systems
Autorzy:
Proksa, R.
Powiązania:
https://bibliotekanauki.pl/articles/333103.pdf
Data publikacji:
2009
Wydawca:
Uniwersytet Śląski. Wydział Informatyki i Nauki o Materiałach. Instytut Informatyki. Zakład Systemów Komputerowych
Tematy:
rozpoznawanie mowy
LPCC
MFCC
wyizolowane słowo
sygnały mowy
speech recognition
cepstral coefficients
isolated word
Opis:
The article presents two methods of determination of cepstral parameters commonly applied in digital signal processing, in particular in speech recognition systems. The solutions presented are part of a project aimed at developing applications allowing to control the Windows operating system with voice and the use of MSAA (Microsoft Active Accessibility). The analysed voice signal has been visually presented at each of the crucial stages of developing cepstral coefficients.
Źródło:
Journal of Medical Informatics & Technologies; 2009, 13; 121-128
1642-6037
Pojawia się w:
Journal of Medical Informatics & Technologies
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Automatic speech signal segmentation based on the innovation adaptive filter
Autorzy:
Makowski, R.
Hossa, R.
Powiązania:
https://bibliotekanauki.pl/articles/330096.pdf
Data publikacji:
2014
Wydawca:
Uniwersytet Zielonogórski. Oficyna Wydawnicza
Tematy:
automatic speech segmentation
inter phoneme boundaries
Schur adaptive filtering
detection threshold determination
automatyczna segmentacja mowy
filtracja adaptacyjna
określenie progu detekcji
Opis:
Speech segmentation is an essential stage in designing automatic speech recognition systems and one can find several algorithms proposed in the literature. It is a difficult problem, as speech is immensely variable. The aim of the authors’ studies was to design an algorithm that could be employed at the stage of automatic speech recognition. This would make it possible to avoid some problems related to speech signal parametrization. Posing the problem in such a way requires the algorithm to be capable of working in real time. The only such algorithm was proposed by Tyagi et al., (2006), and it is a modified version of Brandt’s algorithm. The article presents a new algorithm for unsupervised automatic speech signal segmentation. It performs segmentation without access to information about the phonetic content of the utterances, relying exclusively on second-order statistics of a speech signal. The starting point for the proposed method is time-varying Schur coefficients of an innovation adaptive filter. The Schur algorithm is known to be fast, precise, stable and capable of rapidly tracking changes in second order signal statistics. A transfer from one phoneme to another in the speech signal always indicates a change in signal statistics caused by vocal track changes. In order to allow for the properties of human hearing, detection of inter-phoneme boundaries is performed based on statistics defined on the mel spectrum determined from the reflection coefficients. The paper presents the structure of the algorithm, defines its properties, lists parameter values, describes detection efficiency results, and compares them with those for another algorithm. The obtained segmentation results, are satisfactory.
Źródło:
International Journal of Applied Mathematics and Computer Science; 2014, 24, 2; 259-270
1641-876X
2083-8492
Pojawia się w:
International Journal of Applied Mathematics and Computer Science
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Biometric speech signal processing in a system with digital signal processor
Autorzy:
Marciniak, T.
Weychan, R.
Stankiewicz, A.
Dąbrowski, A.
Powiązania:
https://bibliotekanauki.pl/articles/200794.pdf
Data publikacji:
2014
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
biometry
speech processing
digital signal processor
Gaussian mixture models
vector quantization
Opis:
This paper presents an analysis of issues related to the fixed-point implementation of a speech signal applied to biometric purposes. For preparing the system for automatic speaker identification and for experimental tests we have used the Matlab computing environment and the development software for Texas Instruments digital signal processors, namely the Code Composer Studio (CCS). The tested speech signals have been processed with the TMS320C5515 processor. The paper examines limitations associated with operation of the realized embedded system, demonstrates advantages and disadvantages of the technique of automatic software conversion from Matlab to the CCS and shows the impact of the fixed-point representation on the speech identification effectiveness.
Źródło:
Bulletin of the Polish Academy of Sciences. Technical Sciences; 2014, 62, 3; 589-594
0239-7528
Pojawia się w:
Bulletin of the Polish Academy of Sciences. Technical Sciences
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Fusing the electromagnetic articulograph, high-speed video cameras and a 16-channel microphone array for speech analysis
Autorzy:
Mik, Ł.
Lorenc, A.
Król, D.
Wielgat, R.
Święciński, R.
Jędryka, R.
Powiązania:
https://bibliotekanauki.pl/articles/200263.pdf
Data publikacji:
2018
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
electromagnetic articulography
microphone array
vision system
speech analysis
artykulograf elektromagnetyczny
system wizyjny
analiza mowy
Opis:
Electromagnetic articulography (EMA) is one of the instrumental phonetic research methods used for recording and assessing articulatory movements. Usually, articulographic data are analysed together with standard audio recordings. This paper, however, demonstrates how coupling the articulograph with devices providing other types of information may be used in more advanced speech research. A novel measurement system is presented that consists of the AG 500 electromagnetic articulograph, a 16-channel microphone array with a dedicated audio recorder and a video module consisting of 3 high-speed cameras. It is argued that synchronization of all these devices allows for comparative analyses of results obtained with the three components. To complement the description of the system, the article presents innovative data analysis techniques developed by the authors as well as preliminary results of the system’s accuracy.
Źródło:
Bulletin of the Polish Academy of Sciences. Technical Sciences; 2018, 66, 3; 257-266
0239-7528
Pojawia się w:
Bulletin of the Polish Academy of Sciences. Technical Sciences
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
An Experimental Study of Acoustic Comfort in Open Space Banks Based on Speech Intelligibility and Noise Annoyance Measures
Autorzy:
Golmohammadi, R.
Aliabadi, M.
Nezami, T.
Powiązania:
https://bibliotekanauki.pl/articles/176406.pdf
Data publikacji:
2017
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
acoustic comfort
open-space banks
speech intelligibility
noise annoyance
Opis:
Tasks requiring intensive concentration are more vulnerable to noise than routine tasks. Due to the high mental workload of bank employees, this study aimed to evaluate acoustic comfort in open-space banks based on speech intelligibility and noise annoyance metrics. Acoustic metrics including preferred noise criterion (PNC), speech transmission index (STI), and signal to noise ratio (SNR) were measured in seventeen banks (located in Hamadan, a western province of Iran). For subjective noise annoyance assessments, 100-point noise annoyance scales were completed by bank employees during activities. Based on STI (0.56±0.09) and SNR (20.5±8.2 dB) values, it was found that speech intelligibilities in the workstations of banks were higher than the satisfactory level. However, PNC values in bank spaces were 48.2±5.5 dB, which is higher than the recommended limit value for public spaces. In this regard, 95% of the employees are annoyed by background noise levels. The results show irrelevant speech is the main source of subjective noise annoyance among employees. Loss of concentration is the main consequence of background noise levels for employees. The results confirmed that acoustic properties of bank spaces provide enough speech intelligibility, while staff’s noise annoyance is not acceptable. It can be concluded that due to proximity of workstations in open-space banks, access to very short distraction distance is necessary. Therefore, increasing speech privacy can be prioritised to speech intelligibility. It is recommended that current desk screens are redesigned in order to reduce irrelevant speech between nearby workstations. Staff’s training about acoustic comfort can also manage irrelevant speech characteristics during work time.
Źródło:
Archives of Acoustics; 2017, 42, 2; 333-347
0137-5075
Pojawia się w:
Archives of Acoustics
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Prokurorskij professional'nyj sti'
Prosecutor’s professional style
Autorzy:
Duskaeva, Lilijâ R.
Powiązania:
https://bibliotekanauki.pl/articles/615482.pdf
Data publikacji:
2016
Wydawca:
Uniwersytet Opolski
Tematy:
Stanisław Gajda
professional style
speech genres
Opis:
Based on the integrative concept of style, developed in the works of Professor Stanis³aw Gajda, the article justifies the allocation of prosecutorial professional style as a speech variety of business style, which is realized in a system of prescribing, requesting and informing genres. Analysis of the internal differentiation of the Russian prosecutors’ professional style shows that, along with stylistic features that are common to business macro style, this professional style has specific traits – legal evaluation and contrary relations of semantic positions, – which are expressed in each speech genre in a different way. Stylistical and speech analysis of these two genre models shows that typical algorithms of prosecutors’ professional speech activity are reflected in the models. The study shows that the professional style of prosecutors represents one of the types of speech activities in the legal field.
Źródło:
Stylistyka; 2016, 25; 187-202
1230-2287
2545-1669
Pojawia się w:
Stylistyka
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Slovak Morphosyntactic Tagset
Autorzy:
Garabík, R.
Šimková, M.
Powiązania:
https://bibliotekanauki.pl/articles/103837.pdf
Data publikacji:
2012
Wydawca:
Polska Akademia Nauk. Instytut Podstaw Informatyki PAN
Tematy:
Slovak language
corpus
tagset
morphology
part of speech
grammatical categories
Opis:
Morphological annotation constitutes essential, very useful and very common linguistic information presented in corpora, especially for highly inflectional languages. The morphological tagset used in the Slovak National Corpus has been designed with several goals in mind – the tags are compact and easily human-readable, without sacrificing their informational contents. The tags consist of ASCII letters, numbers and several other characters. In general, they have a variable numer of symbols, but their order is obligatory, and each category or specific feature is assigned a particular character, which can be shared among several parts of speech. The tagset is highly functional and pragmatic, although some allowances had to be made to accommodate the traditional analysis of Slovak morphology and part of speech categories.
Źródło:
Journal of Language Modelling; 2012, 0, 1; 41-63
2299-856X
2299-8470
Pojawia się w:
Journal of Language Modelling
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Frequency Selection Based Separation of Speech Signals with Reduced Computational Time Using Sparse NMF
Autorzy:
Varshney, Y. V.
Abbasi, Z. A.
Abidi, M. R.
Farooq, O.
Powiązania:
https://bibliotekanauki.pl/articles/176829.pdf
Data publikacji:
2017
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
sparse NMF
non-negative matrix factorisation
mixed speech recognition
machine learning
Opis:
Application of wavelet decomposition is described to speed up the mixed speech signal separation with the help of non-negative matrix factorisation (NMF). It is assumed that the basis vectors of training data of individual speakers had been recorded. In this paper, the spectrogram magnitude of a mixed signal has been factorised with the help of NMF with consideration of sparseness of speech signals. The high frequency components of signal contain very small amount of signal energy. By rejecting the high frequency components, the size of input signal is reduced, which reduces the computational time of matrix factorisation. The signal of lower energy has been separated by using wavelet decomposition. The present work is done for wideband microphone speech signal and standard audio signal from digital video equipment. This shows an improvement in the separation capability using the proposed model as compared with an existing one in terms of correlation between separated and original signals. Obtained signal to distortion ratio (SDR) and signal to interference ratio (SIR) are also larger as compare of the existing model. The proposed model also shows a reduction in computational time, which results in faster operation.
Źródło:
Archives of Acoustics; 2017, 42, 2; 287-295
0137-5075
Pojawia się w:
Archives of Acoustics
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Mowa gońca w Hekabie Eurypidesa (518-582)
The Messenger's Speech in Euripides' Hecuba (518-582)
Autorzy:
Chodkowski, Robert R.
Powiązania:
https://bibliotekanauki.pl/articles/1963574.pdf
Data publikacji:
1996
Wydawca:
Katolicki Uniwersytet Lubelski Jana Pawła II. Towarzystwo Naukowe KUL
Opis:
The author carries out a literary analysis of the messenger's speech in Hecuba and seeks to grasp its function in this drama. Therefore he presents the narrative situation of the speech, its character of a report and information, but also interpretation and evaluation. He pinpoints that the role of a messenger in this tragedy is fulfilled by the non-anonymous narrator, yet one of the figures actively involved in the action of the play. It is Talthybius, a man marked with great character, who is able not only to relate the event which has taken place beyond the stage space, but also he may give the event a proper dimension. It is through him that Euripides conveys his audience a lesson that in extreme situations one may preserve human dignity, respect for oneself, or even gain recognition from his enemies. Talthybius' report goes beyond the customary function of the messenger speeches, whose main aim to bring to the stage some events from a remote space. In its interpretative and evaluative structure Talthybius' speech confers a deeper sense of a universal character on the event which is being related. In this function it reminds us the choral chants from Aeschylus' tragedies.
Źródło:
Roczniki Humanistyczne; 1996, 44, 3; 71-79
0035-7707
Pojawia się w:
Roczniki Humanistyczne
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Enhancing Speech Recognition in Adverse Listening Environments: The Impact of Brief Musical Training on Older Adults
Autorzy:
Nandakumar, Akhila R
Somashekara, Haralakatta Shivananjappa
Kanagokar, Vibha
Pitchaimuthu, Arivudai Nambi
Powiązania:
https://bibliotekanauki.pl/articles/31339763.pdf
Data publikacji:
2024
Wydawca:
Polska Akademia Nauk. Czasopisma i Monografie PAN
Tematy:
musical training
carnatic music
speech recognition in noise
speech recognition in reverberation
Opis:
The present research investigated the effects of short-term musical training on speech recognition in adverse listening conditions in older adults. A total of 30 Kannada-speaking participants with no history of gross otologic, neurologic, or cognitive problems were divided equally into experimental (M = 63 years) and control groups (M = 65 years). Baseline and follow-up assessments for speech in noise (SNR50) and reverberation was carried out for both groups. The participants in the experimental group were subjected to Carnatic classical music training, which lasted for seven days. The Bayesian likelihood estimates revealed no difference in SNR50 and speech recognition scores in reverberation between baseline and followed-up assessment for the control group. Whereas, in the experimental group, the SNR50 reduced, and speech recognition scores improved following musical training, suggesting the positive impact of music training. The improved performance on speech recognition suggests that short-term musical training using Carnatic music can be used as a potential tool to improve speech recognition abilities in adverse listening conditions in older adults.
Źródło:
Archives of Acoustics; 2024, 49, 1; 3-9
0137-5075
Pojawia się w:
Archives of Acoustics
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Application of Teager Energy Operator on Linear and Mel Scales for Whispered Speech Recognition
Autorzy:
Marković, B. R.
Galić, J.
Mijić, M.
Powiązania:
https://bibliotekanauki.pl/articles/176961.pdf
Data publikacji:
2018
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
Teager energy operator
cepstral mean subtraction
whispered speech recognition
linear scale
mel scale
dynamic time warping
hidden Markov models
Opis:
This paper presents experimental results on whispered speech recognition based on Teager Energy Operator for linear and mel cepstral coefficients including the Cepstral Mean Subtraction normalization technique. The feature vectors taken into consideration are Linear Frequency Cepstral Coefficients, Teager Energy based Linear Frequency Cepstral Coefficients, Mel Frequency Cepstral Coefficients and Teager Energy based Mel Frequency Cepstral Coefficients. A speaker dependent scenario is used. For the recognition process, Dynamic Time Warping and Hidden Markov Models methods are applied. Results show a respectable improvement in whispered speech recognition as achieved by using the Teager Energy Operator with Cepstral Mean Subtraction.
Źródło:
Archives of Acoustics; 2018, 43, 1; 3-9
0137-5075
Pojawia się w:
Archives of Acoustics
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Dziennikarski styl zawodowy. Próba zdefiniowania
Professional journalistic style. An attempt to define
Autorzy:
Duskajewa, Lilia R
Powiązania:
https://bibliotekanauki.pl/articles/967969.pdf
Data publikacji:
2016
Wydawca:
Uniwersytet Łódzki. Wydawnictwo Uniwersytetu Łódzkiego
Tematy:
professional language
speech genres
social problem
Opis:
The article substantiates the distinction of professional journalistic language as a variant of langue representing journalistic style, and implemented in three groups of speech genres – informing, estimating and stimulating. Alongside stylistic features common with journalistic macrostyle, this professional language has its own specific characteristics, expressed through faithfulness to the facts, social estimation and administrative stimulation. Stylistic analysis of even only one genre – stimulating – reveals that the professional and linguistic activities of journalists can be analysed as typical execution algorithms, using the aforementioned method. The article demonstrates that the proposed analysis of speech representations in professional language can be productive for the study of language in action in the journalistic sphere of communication.
Źródło:
Acta Universitatis Lodziensis. Folia Litteraria Polonica; 2016, 32, 2
1505-9057
2353-1908
Pojawia się w:
Acta Universitatis Lodziensis. Folia Litteraria Polonica
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Untersuchung zu der Realisierung des Phonems /r/ im Unbetonten Wortfinalen Silbenanlaut in der Modernen Deutschen Spontanrede
Investigation on the realization of the phoneme /r/ in the unstressed word final syllable onset in modern German spontaneous speech
Autorzy:
Solska, Tetiana
Borovska, Olena
Poseletska, Kateryna
Vyshyvana, Nataliia
Powiązania:
https://bibliotekanauki.pl/articles/2119954.pdf
Data publikacji:
2022
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
spontaneous speech
phoneme /ʁ/
unstressed word final syllable onset
modifications
vocalization
elision
secondary phonetic diphthong
Opis:
The present paper deals with the synchronic variation of the phoneme /ʁ/ in the unstressed word final syllable onset in modern German spontaneous speech. Our research task was to determine the phonetic context, in which the phoneme /ʁ/ undergoes modifications in the above-mentioned position and to establish, whether the intensity and the type of modifications (vocalization or elision of the phoneme /ʁ/) could correlate with the part of speech and with the combinatorial conditions of sound realization. The data collected are based on the acoustic analysis of spontaneous speech (interviews in the media) of 20 German scientists (10 men and 10 women) from the Central and Southern Germany. Our results showed that the phoneme /ʁ/ undergoes intense modifications mainly in the word final position "stressed long vowel + ʁ + schwa vowel + nasal" in various parts of speech: verbs, plural forms of nouns, adjectives, participles, substantivized verbs, possessive pronouns and prepositions. The type of modification of the phoneme /ʁ/ in the relevant position correlates with the sound context. After high and mid vowels [iː], [yː], [uː], [eː], [ɛː], [øː], [oː] vowel realizations as unsyllabic [ɐ̯] clearly dominate over the consonantal as [ʁ], leading to the emergence of centralizing secondary diphthongs [iːɐ̯], [yːɐ̯], [uːɐ̯], [eːɐ̯], [ɛːɐ̯], [øːɐ̯], [oːɐ̯]. In the position after the long [aː] an elision of the allophones of the phoneme /ʁ/ is predominant, which can lead to an overlong articulation of the preceding low vowel as [aːː].
Źródło:
Linguistica Silesiana; 2022, 43; 7-23
0208-4228
Pojawia się w:
Linguistica Silesiana
Dostawca treści:
Biblioteka Nauki
Artykuł
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