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Search results for: dysarthric speech
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Interpretable Deep Learning Model for the Detection and Reconstruction of Dysarthric Speech
PublicationWe present a novel deep learning model for the detection and reconstruction of dysarthric speech. We train the model with a multi-task learning technique to jointly solve dysarthria detection and speech reconstruction tasks. The model key feature is a low-dimensional latent space that is meant to encode the properties of dysarthric speech. It is commonly believed that neural networks are black boxes that solve problems but do not...
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Evaluation of Lombard Speech Models in the Context of Speech in Noise Enhancement
PublicationThe Lombard effect is one of the most well-known effects of noise on speech production. Speech with the Lombard effect is more easily recognizable in noisy environments than normal natural speech. Our previous investigations showed that speech synthesis models might retain Lombard-effect characteristics. In this study, we investigate several speech models, such as harmonic, source-filter, and sinusoidal, applied to Lombard speech...
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High Speed Milling vibration surveillance with optimal spindle speed based on optimal speeds map
PublicationThe paper presents the method of the surveillance of the self-excited chatter vibration. At first, the workpiece modal parameters are estimated based on experimental data which leads to verification of computational model. Then, for selected surface points optimal spindle speeds are calculated. By considering sufficient amount of points it is possible to build a map of optimal spindle speeds. Experimental results show that this...
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Estimation of the excitation variances of speech and noise AR-models for enhanced speech coding
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Optimizing Medical Personnel Speech Recognition Models Using Speech Synthesis and Reinforcement Learning
PublicationText-to-Speech synthesis (TTS) can be used to generate training data for building Automatic Speech Recognition models (ASR). Access to medical speech data is because it is sensitive data that is difficult to obtain for privacy reasons; TTS can help expand the data set. Speech can be synthesized by mimicking different accents, dialects, and speaking styles that may occur in a medical language. Reinforcement Learning (RL), in the...
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Intelligent processing of stuttered speech.
PublicationW artykule zaprezentowano kilka metod analizy i automatycznego zliczania potknięć artykulacyjnych, związanych z jąkaniem się, opartych na wykorzystaniu algorytmów uczących się sztucznych sieci neuronowych i zbiorów przybliżonych.
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Speech Intelligibility Measurements in Auditorium
PublicationSpeech intelligibility was measured in Auditorium Novum on Technical University of Gdansk (seating capacity 408, volume 3300 m3). Articulation tests were conducted; STI and Early Decay Time EDT coefficients were measured. Negative noise contribution to speech intelligibility was taken into account. Subjective measurements and objective tests reveal high speech intelligibility at most seats in auditorium. Correlation was found between...
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Language Models in Speech Recognition
PublicationThis chapter describes language models used in speech recognition, It starts by indicating the role and the place of language models in speech recognition. Mesures used to compare language models follow. An overview of n-gram, syntactic, semantic, and neural models is given. It is accompanied by a list of popular software.
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Comparison of Language Models Trained on Written Texts and Speech Transcripts in the Context of Automatic Speech Recognition
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A system for multitask noisy speech enhancement.
PublicationW artykule przedstawiono ogolną charakterystyke opracowanego systemu rejestracji i rekonstrukcji mowy. Artykuł zawiera opis składników systemu, ktory jest oprogramowaniem zawierającym zaawansowane narzędzia służące poprawie zrozumiałości mowy. Zaimplementowane narzędzia systemu umożliwiają wyszukiwanie nagrań dźwiękowych i ich obróbkę przy pomocy zaimplementowanych pluginów. W artykule przedstawione wykorzystane w systemie algorytmy...
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Multitask Noisy Speech Enhancement System
PublicationW referacie opisano Wielozadaniowy System Poprawy Jakości Sygnału Mowy. Jest to wyspecjalizowany pakiet oprogramowania przeznaczony do rejestrowania sygnału mowy i do poprawy jego jakości oraz zrozumiałości mowy, przy użyciu zaawansowanych procedur cyfrowego przetwarzania sygnału. Pakiet oprogramowania składa się z programów: Rejestrator, Przeglądarka oraz Rekonstruktor. Oprogramowanie to może być użyte w przypadkach, gdy zrozumiałość...
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Transient detection for speech coding applications
PublicationSignal quality in speech codecs may be improved by selecting transients from speech signal and encoding them using a suitable method. This paper presents an algorithm for transient detection in speech signal. This algorithm operates in several frequency bands. Transient detection functions are calculated from energy measured in short frames of the signal. The final selection of transient frames is based on results of detection...
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Novel approaches to wideband speech coding
PublicationDwie metoda kodowania szerokopasmowego mowy zostały zaprezentowane. W pierwszej metodzie wykorzystano algorytm kompresji i ekspansji czasowej sygnału mowy, pozwalający na kodowanie szerokopasmowe sygnału mowy z wykorzystaniem ustandaryzowanych kodeków. Metoda ta jest przewidziana do zastosowania w adaptacyjnych algorytmach kodowania mowy. Drugie z proponowanych rozwiazan dotyczy nowej metody estymacji obwiedni widma sygnalu mowy...
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Broadband interference in speech reinforcement systems
PublicationArtykuł podejmuje niedoceniany problem wpływu liczby i rozkładu głośników w systemach nagłośnienia, na jakość przekazu głosowego, czyli na zrozumiałość mowy w audytoriach. Superpozycji przesuniętych w czasie szerokopasmowych sygnałów o tym samym kształcie i lekko różnych wielkościach, które docierają do słuchacza z licznych spójnych źródeł, towarzyszy zjawisko interferencji prowadzące do głębokiej modyfikacji odbieranych sygnałów...
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Integration of speech enhancement and coding techniques
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Speech synthesis controlled by eye gazing
PublicationA method of communication based on eye gaze controlling is presented. Investigations of using gaze tracking have been carried out in various context applications. The solution proposed in the paper could be referred to as ''talking by eyes'' providing an innovative approach in the domain of speech synthesis. The application proposed is dedicated to disabled people, especially to persons in a so-called locked-in syndrome who cannot...
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Speech Analytics Based on Machine Learning
PublicationIn this chapter, the process of speech data preparation for machine learning is discussed in detail. Examples of speech analytics methods applied to phonemes and allophones are shown. Further, an approach to automatic phoneme recognition involving optimized parametrization and a classifier belonging to machine learning algorithms is discussed. Feature vectors are built on the basis of descriptors coming from the music information...
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Speech recognition system for hearing impaired people.
PublicationPraca przedstawia wyniki badań z zakresu rozpoznawania mowy. Tworzony system wykorzystujący dane wizualne i akustyczne będzie ułatwiał trening poprawnego mówienia dla osób po operacji transplantacji ślimaka i innych osób wykazujących poważne uszkodzenia słuchu. Active Shape models zostały wykorzystane do wyznaczania parametrów wizualnych na podstawie analizy kształtu i ruchu ust w nagraniach wideo. Parametry akustyczne bazują na...
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Comprehensive Evaluation of Statistical Speech Waveform Synthesis
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Methodology and technology for the polymodal allophonic speech transcription
PublicationA method for automatic audiovisual transcription of speech employing: acoustic, electromagnetical articulography and visual speech representations is developed. It adopts a combining of audio and visual modalities, which provide a synergy effect in terms of speech recognition accuracy. To establish a robust solution, basic research concerning the relation between the allophonic variation of speech, i.e., the changes in the articulatory...
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Methodology and technology for the polymodal allophonic speech transcription
PublicationA method for automatic audiovisual transcription of speech employing: acoustic and visual speech representations is developed. It adopts a combining of audio and visual modalities, which provide a synergy effect in terms of speech recognition accuracy. To establish a robust solution, basic research concerning the relation between the allophonic variation of speech, i.e. the changes in the articulatory setting of speech organs for...
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Applying the Lombard Effect to Speech-in-Noise Communication
PublicationThis study explored how the Lombard effect, a natural or artificial increase in speech loudness in noisy environments, can improve speech-in-noise communication. This study consisted of several experiments that measured the impact of different types of noise on synthesizing the Lombard effect. The main steps were as follows: first, a dataset of speech samples with and without the Lombard effect was collected in a controlled setting;...
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Improved method for real-time speech stretching
Publicationn algorithm for real-time speech stretching is presented. It was designed to modify input signal dependently on its content and on its relation with the historical input data. The proposed algorithm is a combination of speech signal analysis algorithms, i.e. voice, vowels/consonants, stuttering detection and SOLA (Synchronous-Overlap-and-Add) based speech stretching algorithm. This approach enables stretching input speech signal...
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Influence of modulation detection threshold on speech intelligibility
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Communication Platform for Evaluation of Transmitted Speech Quality
PublicationA voice communication system designed and implemented is described. The purpose of the presented platform was to enable a series of experiments related to the quality assessment of algorithms used in the coding and transmitting of speech. The system is equipped with tools for recording signals at each stage of processing, making it possible to subject them to subjective assessments by listening tests or, objective evaluation employing...
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Silence/noise detection for speech and music signals
PublicationThis paper introduces a novel off-line algorithm for silence/noise detection in noisy signals. The main concept of the proposed algorithm is to provide noise patterns for further signals processing i.e. noise reduction for speech enhancement. The algorithm is based on frequency domain characteristics of signals. The examples of different types of noisy signals are presented.
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New generation speech aid for stuttering people
PublicationWspółczesne Cyfrowe Procesory Sygnałowe (ang. DSP) mają niewielkie wymiary, ale są w stanie re-alizować złożone algorytmy. Ich dodatkową zaletą jest łatwość wymiany oprogramowania, a co za tym idzie łatwość zmiany dziedziny zastosowań. Wykorzystując możliwości procesów stało się możliwe budowanie miniaturowych protez słuchu i mowy. W referacie skupiono się na zagadnieniach związanych z projekto-wanie i implementacją algorytmów...
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New generation speech aid for stuttering people
PublicationWspółczesne Cyfrowe Procesory Sygnałowe (ang. DSP) mają niewielkie wymiary, ale są w stanie re-alizować złożone algorytmy. Ich dodatkową zaletą jest łatwość wymiany oprogramowania, a co za tym idzie łatwość zmiany dziedziny zastosowań. Wykorzystując możliwości procesów stało się możliwe budowanie miniaturowych protez słuchu i mowy. W referacie skupiono się na zagadnieniach związanych z projekto-wanie i implementacją algorytmów...
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Transient detection algorithms for speech coding applications
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Multimodal English corpus for automatic speech recognition
PublicationA multimodal corpus developed for research of speech recognition based on audio-visual data is presented. Besides usual video and sound excerpts, the prepared database contains also thermovision images and depth maps. All streams were recorded simultaneously, therefore the corpus enables to examine the importance of the information provided by different modalities. Based on the recordings, it is also possible to develop a speech...
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Real-time speech-rate modification experiments
PublicationAn algorithm designed for real-time speech time scale modification (stretching) is proposed, providing a combination of typical synchronous overlap and add based time scale modification algorithm and signal redundancy detection algorithms that allow to remove parts of the speech signal and replace them with the stretched speech signal fragments. Effectiveness of signal processing algorithms are examined experimentally together...
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Building Knowledge for the Purpose of Lip Speech Identification
PublicationConsecutive stages of building knowledge for automatic lip speech identification are shown in this study. The main objective is to prepare audio-visual material for phonetic analysis and transcription. First, approximately 260 sentences of natural English were prepared taking into account the frequencies of occurrence of all English phonemes. Five native speakers from different countries read the selected sentences in front of...
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Tensor Decomposition for Imagined Speech Discrimination in EEG
PublicationMost of the researches in Electroencephalogram(EEG)-based Brain-Computer Interfaces (BCI) are focused on the use of motor imagery. As an attempt to improve the control of these interfaces, the use of language instead of movement has been recently explored, in the form of imagined speech. This work aims for the discrimination of imagined words in electroencephalogram signals. For this purpose, the analysis of multiple variables...
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Constructing a Dataset of Speech Recordingswith Lombard Effect
PublicationThepurpose of therecordings was to create a speech corpus based on the ISLEdataset, extended with video and Lombard speech. Selected from a set of 165sentences, 10, evaluatedas having thehighest possibility to occur in the context ofthe Lombard effect,were repeated in the presence of the so-called babble speech to obtain Lombard speech features. Altogether,15speakers were recorded, and speech parameterswere...
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Improving the quality of speech in the conditions of noise and interference
PublicationThe aim of the work is to present a method of intelligent modification of the speech signal with speech features expressed in noise, based on the Lombard effect. The recordings utilized sets of words and sentences as well as disturbing signals, i.e., pink noise and the so-called babble speech. Noise signal, calibrated to various levels at the speaker's ears, was played over two loudspeakers located 2 m away from the speaker. In...
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Creation of a map of optimal spindle speeds for a purpose of surveillance of high speed milling of flexible details
PublicationDrgania samowzbudne typu chatter często towarzyszą procesom frezowania szybkościowego. Ich skutkiem są nieodwracalne uszkodzenia powierzchni obrabianej. Mogą one również doprowadzić do intensywnego zużycia narzędzi skrawających, a nawet samej obrabiarki. W pracy przedstawiono metodę umożliwiającą tworzenie mapy optymalnych prędkości obrotowych wrzeciona. Stosowanie podczas procesu frezowania optymalnej prędkości obrotowej wrzeciona...
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New approach to localization of clicks in archive speech signals.
PublicationPrzedstawiono 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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Advanced speech archiving and restoration system for aviation applications
PublicationW 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
PublicationDzięki 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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Automatic Image and Speech Recognition Based on Neural Network
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Audiovisual speech recognition for training hearing impaired patients
PublicationPraca 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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A Method of Real-Time Non-uniform Speech Stretching
PublicationDeveloped method of real-time non-uniform speech stretching is presented.The proposed solution is based on the well-known SOLA algorithm(Synchronous Overlap and Add). Non-uniform time-scale modification isachieved by the adjustment of time scaling factor values in accordance with thesignal content. Dependently on the speech unit (vowels/consonants), instantaneousrate of speech (ROS), and speech signal presence, values of the scalingfactor...
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Comparison of various speech time-scale modificartion methods
PublicationThe objective of this work is to investigate the influence of the different time-scale modification (TSM) methods on the quality of the speech stretched up using the designed non-uniform real-time speech time-scale modification algorithm (NU-RTSM). The algorithm provides a combination of the typical TSM algorithm with the vowels, consonants, stutter, transients and silence detectors. Based on the information about the content and...
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Investigations of speech signal parameters with regard to articulation influences
PublicationW 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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Detection of dialogue in movie soundtrack for speech intelligibility enhancement
PublicationA 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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Visual Lip Contour Detection for the Purpose of Speech Recognition
PublicationA method for visual detection of lip contours in frontal recordings of speakers is described and evaluated. The purpose of the method is to facilitate speech recognition with visual features extracted from a mouth region. Different Active Appearance Models are employed for finding lips in video frames and for lip shape and texture statistical description. Search initialization procedure is proposed and error measure values are...
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Time-domain prosodic modifications for text-to-speech synthesizer
PublicationAn application of prosodic speech processing algorithms to Text-To-Speech synthesis is presented. Prosodic modifications that improve the naturalness of the synthesized signal are discussed. The applied method is based on the TD-PSOLA algorithm. The developed Text-To-Speech Synthesizer is used in applications employing multimodal computer interfaces.