Wyniki wyszukiwania dla: speech synthesis
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Transfer learning in imagined speech EEG-based BCIs
PublikacjaThe Brain–Computer Interfaces (BCI) based on electroencephalograms (EEG) are systems which aim is to provide a communication channel to any person with a computer, initially it was proposed to aid people with disabilities, but actually wider applications have been proposed. These devices allow to send messages or to control devices using the brain signals. There are different neuro-paradigms which evoke brain signals of interest...
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Distortion of speech signals in the listening area: its mechanism and measurements
PublikacjaThe paper deals with a problem of the influence of the number and distribution of loudspeakers in speech reinforcement systems on the quality of publicly addressed voice messages, namely on speech intelligibility in the listening area. Linear superposition of time-shifted broadband waves of a same form and slightly different magnitudes that reach a listener from numerous coherent sources, is accompanied by interference effects...
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Objectivization of phonological evaluation of speech elements by means of audio parametrization
PublikacjaThis study addresses two issues related to both machine- and subjective-based speech evaluation by investigating five phonological phenomena related to allophone production. Its aim is to use objective parametrization and phonological classification of the recorded allophones. These allophones were selected as specifically difficult for Polish speakers of English: aspiration, final obstruent devoicing, dark lateral /l/, velar nasal...
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Investigating Noise Interference on Speech Towards Applying the Lombard Effect Automatically
PublikacjaThe aim of this study is two-fold. First, we perform a series of experiments to examine the interference of different noises on speech processing. For that purpose, we concentrate on the Lombard effect, an involuntary tendency to raise speech level in the presence of background noise. Then, we apply this knowledge to detecting speech with the Lombard effect. This is for preparing a dataset for training a machine learning-based...
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Human-computer interactions in speech therapy using a blowing interface
PublikacjaIn this paper we present a new human-computer interface for the quantitative measurement of blowing activities. The interface can measure the air flow and air pressure during the blowing activity. The measured values are stored and used to control the state of the graphical objects in the graphical user interface. In speech therapy children will find easier to play attractive therapeutic games than to perform repetitive and tedious,...
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Comparison of Acoustic and Visual Voice Activity Detection for Noisy Speech Recognition
PublikacjaThe problem of accurate differentiating between the speaker utterance and the noise parts in a speech signal is considered. The influence of utilizing a voice activity detection in speech signals on the accuracy of the automatic speech recognition (ASR) system is presented. The examined methods of voice activity detection are based on acoustic and visual modalities. The problem of detecting the voice activity in clean and noisy...
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Creating new voices using normalizing flows
PublikacjaCreating realistic and natural-sounding synthetic speech remains a big challenge for voice identities unseen during training. As there is growing interest in synthesizing voices of new speakers, here we investigate the ability of normalizing flows in text-to-speech (TTS) and voice conversion (VC) modes to extrapolate from speakers observed during training to create unseen speaker identities. Firstly, we create an approach for TTS...
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Corrupted speech intelligibility improvement using adaptive filter based algorithm
PublikacjaA technique for improving the quality of speech signals recorded in strong noise is presented. The proposed algorithmemploying adaptive filtration is described and additional possibilities of speech intelligibility improvement arediscussed. Results of the tests are presented.
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A non-uniform real-time speech time-scale stretching method
PublikacjaAn algorithm for non-uniform real-time speech stretching is presented. It provides a combination of typical SOLA algorithm (Synchronous Overlap and Add ) with the vowels, consonants and silence detectors. Based on the information about the content and the estimated value of the rate of speech (ROS), the algorithm adapts the scaling factor value. The ability of real-time speech stretching and the resultant quality of voice were...
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Investigations of speech signal parameters with regard to articulation influences
PublikacjaW pracy zostało podjęte zagadnienie parametryzacji sygnału mowy w kontekście ekstrakcji cech biometrycznych. Analizowane parametry to parametry cepstralne (cepstrum liniowe i mel-cepstrum, czyli MFCC), parametry liniowej predykcji (LPC) oraz momenty widmowe i parametr F0. Zastosowano analize w krótkich stałych segmentach sygnału z zastosowaniem dużego zakładkowania, tzw. ''implicite segmentation''. Umożliwiło to zaobserwowanie...
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System of speech signal processing and visualisation for linguistic purposes
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Digital analysis of ethnic speech – extraction of information code
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On the EM algorithm for the estimation of speech AR parameters in noise
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Evaluation and Irony in Text in the Light of Speech Act Theory
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Audiovisual speech recognition for training hearing impaired patients
PublikacjaPraca przedstawia system rozpoznawania izolowanych głosek mowy wykorzystujący dane wizualne i akustyczne. Modele Active Shape Models zostały wykorzystane do wyznaczania parametrów wizualnych na podstawie analizy kształtu i ruchu ust w nagraniach wideo. Parametry akustyczne bazują na współczynnikach melcepstralnych. Sieć neuronowa została użyta do rozpoznawania wymawianych głosek na podstawie wektora cech zawierającego oba typy...
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Advanced speech archiving and restoration system for aviation applications
PublikacjaW referacie przedstawiono opracowany System Rejestracji I Rekonstrukcji Mowy dla potrzeb lotnictwa. System ten umożliwia jednoczesny zapis, archiwizację i poprawę zrozumiałości sygnału mowy pochodzącego z wielu różnych kanałów komunikacji radiowej. Głównym celem systemu jest rejestracja i rekonstrukcja komunikatów słownych wymienianych drogą radiową pomiędzy pilotem samolotu a stacją kontroli lotów - jest to niezwykle istotne w...
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Application of hybrid signals processors to speech and hearing aids
PublikacjaDzięki postępowi w technice Cyfrowych Procesorów Sygnałowych (ang. DSP) stało się możliwe budowanie miniaturowych protez słuchu i mowy. Mimo niewielkich wymiarów procesory te są w stanie wykonywać złożone algorytmy. Ich dodatkową zaletą jest łatwość zmiany oprogramowania, a co za tym idzie łatwość zmiany dziedziny zastosowań. W pracy skupiono się na zagadnieniach związanych z projektowanie i implementacją algorytmów mających zastosowanie...
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New approach to localization of clicks in archive speech signals.
PublikacjaPrzedstawiono problem lokalizacji zniekształceń impulsowych w archiwalnych sygnałach mowy. Pokazano, że detekcja oparta na dwuzakresowym modelu autoregresyjnym i przetwarzanie dwukierunkowe pozwala uzyskać znaczącą poprawę działania w stosunku do istniejących metod lokalizacji zniekształceń.
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Detection of dialogue in movie soundtrack for speech intelligibility enhancement
PublikacjaA method for detecting dialogue in 5.1 movie soundtrack based on interchannel spectral disparity is presented. The front channel signals (left, right, center) are analyzed in the frequency domain. The selected partials in the center channel signal, which yield high disparity with left and right channels, are detected as dialogue. Subsequently, the dialogue frequency components are boosted to achieve increased dialogue intelligibility....
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Noise profiling for speech enhancement employing machine learning models
PublikacjaThis paper aims to propose a noise profiling method that can be performed in near real-time based on machine learning (ML). To address challenges related to noise profiling effectively, we start with a critical review of the literature background. Then, we outline the experiment performed consisting of two parts. The first part concerns the noise recognition model built upon several baseline classifiers and noise signal features...