Filtry
wszystkich: 1870
wyświetlamy 1000 najlepszych wyników Pomoc
Wyniki wyszukiwania dla: SPEECH SIGNAL PARAMETERIZATION
-
Rediscovering Automatic Detection of Stuttering and Its Subclasses through Machine Learning—The Impact of Changing Deep Model Architecture and Amount of Data in the Training Set
PublikacjaThis work deals with automatically detecting stuttering and its subclasses. An effective classification of stuttering along with its subclasses could find wide application in determining the severity of stuttering by speech therapists, preliminary patient diagnosis, and enabling communication with the previously mentioned voice assistants. The first part of this work provides an overview of examples of classical and deep learning...
-
Michał Mazur dr inż.
OsobyAktualne zainteresowania inżynieria mechaniczna, robotyka, drgania mechaniczne, analiza modalna, sterowanie, systemy czasu rzeczywistego Wybrane publikacje Kaliński K., Galewski M., Mazur M., Chodnicki M, 2017, Modelling and Simulation Of A New Variable Stiffness Holder for Milling Of Flexible Details, Polish Maritime Research, vol 24, ss. 115-124 Kaliński K. J., Mazur M.: Optimal control at energy performance index of the mobile...
-
New Algorithms for Adaptive Notch Smoothing
PublikacjaThe problem of extraction/elimination of a nonstationary complex sinusoidal signal buried in noise is considered. This problem is usually solved using adaptive notch filtering (ANF)algorithms. It is shown that accuracy of signal estimation can be increased if the results obtained from ANF are further processed using a cascade of appropriately designed filters. The resulting adaptive notch smoothing (ANS) algorithms can be employed...
-
Jacek Stefański prof. dr hab. inż.
OsobyJacek Stefański ukończył studia na Wydziale Elektroniki Politechniki Gdańskiej (PG) w 1993 r. W 2000 r. uzyskał stopień doktora nauk technicznych w dyscyplinie telekomunikacja, w 2012 r. stopień doktora habilitowanego, natomiast w 2020 r. tytuł profesora nauk inżynieryjno-technicznych. Obecnie pracuje na stanowisku profesora w Katedrze Systemów i Sieci Radiokomunikacyjnych (KSiSR) PG. W latach 2005-2009 był zatrudniony w Instytucie...
-
Variable Ratio Sample Rate Conversion Based on Fractional Delay Filter
PublikacjaIn this paper a sample rate conversion algorithm which allows for continuously changing resampling ratio has been presented. The proposed implementation is based on a variable fractional delay filter which is implemented by means of a Farrow structure. Coefficients of this structure are computed on the basis of fractional delay filters which are designed using the offset window method. The proposed approach allows us to freely...
-
Quality Evaluation of Novel DTD Algorithm Based on Audio Watermarking
PublikacjaEcho cancellers typically employ a doubletalk detection (DTD) algorithm in order to keep the adaptive filter from diverging in the presence of near-end speech signal or other disruptive sounds in the microphone signal. A novel doubletalk detection algorithm based on techniques similar to those used for audio signal watermarking was introduced by the authors. The application of the described DTD algorithm within acoustic echo cancellation...
-
New approach for determining the QoS of MP3-coded voice signals in IP networks
PublikacjaPresent-day IP transport platforms being what they are, it will never be possible to rule out conflicts between the available services. The logical consequence of this assertion is the inevitable conclusion that the quality of service (QoS) must always be quantifiable no matter what. This paper focuses on one method to determine QoS. It defines an innovative, simple model that can evaluate the QoS of MP3-coded voice data transported...
-
A Comparison of STI Measured by Direct and Indirect Methods for Interiors Coupled with Sound Reinforcement Systems
PublikacjaThis paper presents a comparison of STI (Speech Transmission Index) coefficient measurement results carried out by direct and indirect methods. First, acoustic parameters important in the context of public address and sound reinforcement systems are recalled. A measurement methodology is presented that employs various test signals to determine impulse responses. The process of evaluating sound system performance, signals enabling...
-
Selected results of signal measurements in a ship power station with two generators working in parallel with the use of the Estimator/Analyzer instrument
Dane BadawczeThe presented dataset is part of research focusing on the assessment of metrological properties of the instrument, Estimator/ Analyser (A/E v.2), developed and made at the Faculty of Electrical Engineering, Department of Marine Electrical Power Engineering, of Gdynia Maritime University. The instrument performs a set of measurement functions that...
-
Zbigniew Czaja dr hab. inż.
OsobyZbigniew Czaja was born in Czluchow, Poland, in 1970. He graduated from Gdansk University of Technology, Faculty of Electronics, Telecommunications and Informatics in 1995. The Ph.D. degree in electronics was received in 2001 from the same university. He worked as assistant professor at this university from 2002 to 2013. In 2014, he qualified as an associate professor, and in 2017, as a professor. His research interests are in...
-
JOURNAL OF RECEPTORS AND SIGNAL TRANSDUCTION
Czasopisma -
Signal Image and Video Processing
Czasopisma -
Improving listeners' experience for movie playback through enhancing dialogue clarity in soundtracks
PublikacjaThis paper presents a method for improving users' quality of experience through processing of movie soundtracks. The dialogue clarity enhancement algorithms were introduced for detecting dialogue in movie soundtrack mixes and then for amplifying the dialogue components. 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...
-
Germline DNA Retention in Murine and Human Rearranged T Cell Receptor Gene Coding Joints: Alternative Recombination Signal Sequences and V(D)J Recombinase Errors
Publikacja -
Analysis-by-synthesis paradigm evolved into a new concept
PublikacjaThis work aims at showing how the well-known analysis-by-synthesis paradigm has recently been evolved into a new concept. However, in contrast to the original idea stating that the created sound should not fail to pass the foolproof synthesis test, the recent development is a consequence of the need to create new data. Deep learning models are greedy algorithms requiring a vast amount of data that, in addition, should be correctly...
-
A modified method of vibration surveillance by using the optimal control at energy performance index
PublikacjaA method of vibration surveillance by using the optimal control at energy performance index has been creatively modified. The suggested original modification depends on consideration of direct relationship between the measured acceleration signal and the optimal control command. The paper presents the results of experiments and Hardware- in-the-loop simulations of a new active vibration reduction algorithm based on the energy...
-
Comparative analysis of various transformation techniques for voiceless consonants modeling
PublikacjaIn this paper, a comparison of various transformation techniques, namely Discrete Fourier Transform (DFT), Discrete Cosine Transform (DCT) and Discrete Walsh Hadamard Transform (DWHT) are performed in the context of their application to voiceless consonant modeling. Speech features based on these transformation techniques are extracted. These features are mean and derivative values of cepstrum coefficients, derived from each transformation....
-
Selection of Features for Multimodal Vocalic Segments Classification
PublikacjaEnglish speech recognition experiments are presented employing both: audio signal and Facial Motion Capture (FMC) recordings. The principal aim of the study was to evaluate the influence of feature vector dimension reduction for the accuracy of vocalic segments classification employing neural networks. Several parameter reduction strategies were adopted, namely: Extremely Randomized Trees, Principal Component Analysis and Recursive...
-
New semi-causal and noncausal techniques for detection of impulsive disturbances in multivariate signals with audio applications
PublikacjaThis paper deals with the problem of localization of impulsive disturbances in nonstationary multivariate signals. Both unidirectional and bidirectional (noncausal) detection schemes are proposed. It is shown that the strengthened pulse detection rule, which combines analysis of one-step-ahead signal prediction errors with critical evaluation of leave-one-out signal interpolation errors, allows one to noticeably improve detection results...
-
A Novel Approach to the Assessment of Cough Incidence
PublikacjaIn this paper we consider the problem of identication of cough events in patients suffering from chronic respiratory diseases. The information about frequency of cough events is necessary to medical treatment. The proposed approach is based on bidirectional processing of a measured vibration signal - cough events are localized by combining the results of forward-time and backward-time analysis. The signal is at rst transformed...
-
Dynamic mass measurement in checkweighers using a discrete time-variant low-pass filter
PublikacjaConveyor belt type checkweighers are complex mechanical systems consisting of a weighing sensor (strain gauge load cell, electrodynamically compensated load cell), packages (of different shapes, made of different materials) and a transport system (motors, gears, rollers). Disturbances generated by the vibrating parts of such a system are reflected in the signal power spectra in a form of strong spectral peaks, located usually in...
-
IEEE Automatic Speech Recognition and Understanding Workshop
Konferencje -
European Signal Processing Conference
Konferencje -
Playback detection using machine learning with spectrogram features approach
PublikacjaThis paper presents 2D image processing approach to playback detection in automatic speaker verification (ASV) systems using spectrograms as speech signal representation. Three feature extraction and classification methods: histograms of oriented gradients (HOG) with support vector machines (SVM), HAAR wavelets with AdaBoost classifier and deep convolutional neural networks (CNN) were compared on different data partitions in respect...
-
Multichannel self-optimizing narrowband interference canceller
PublikacjaThe problem of cancellation of a nonstationary sinusoidal interference, acting at the output of an unknown multivariable linear stable plant, is considered. No reference signal is assumed to be available. The proposed feedback controller is a nontrivial extension of the SONIC (self-optimizing narrowband interference canceller) algorithm, developed earlier for single-input, single-output plants. The algorithm consists of two loops:...
-
Elimination of Impulsive Disturbances From Stereo Audio Recordings Using Vector Autoregressive Modeling and Variable-order Kalman Filtering
PublikacjaThis paper presents a new approach to elimination of impulsive disturbances from stereo audio recordings. The proposed solution is based on vector autoregressive modeling of audio signals. Online tracking of signal model parameters is performed using the exponential ly weighted least squares algo- rithm. Detection of noise pulses an d model-based interpolation of the irrevocably distorted sampl es is realized using an adaptive, variable-order...
-
Introduction to the special issue on machine learning in acoustics
PublikacjaWhen we started our Call for Papers for a Special Issue on “Machine Learning in Acoustics” in the Journal of the Acoustical Society of America, our ambition was to invite papers in which machine learning was applied to all acoustics areas. They were listed, but not limited to, as follows: • Music and synthesis analysis • Music sentiment analysis • Music perception • Intelligent music recognition • Musical source separation • Singing...
-
Signal Processing: An International Journal (SPIJ)
Czasopisma -
A simple way of increasing estimation accuracy of generalized adaptive notch filters
PublikacjaGeneralized adaptive notch filters are used for identification/tracking of quasi-periodically varying dynamic systems and can be considered an extension, to the system case, of classical adaptive notch filters. It is shown that frequency biases, which arisein generalized adaptive notch filtering algorithms, can be significantly reduced by incorporating in the adaptive loop an appropriately chosen decision delay. The resulting performance...
-
Grzegorz Szwoch dr hab. inż.
OsobyGrzegorz Szwoch urodził się w 1972 roku w Gdańsku. W latach 1991-1996 studiował na wydziale Elektroniki Politechniki Gdańskiej. W roku 1996 ukończył studia w Zakładzie Inżynierii Dźwięku (obecnie Katedra Systemów Multimedialnych), broniąc pracę dyplomową pt. Modelowanie fizyczne wybranych instrumentów muzycznych. W tym samym roku dołączył do zespołu badawczego Katedry jako uczestnik Studium Doktoranckiego. Od stycznia 2001 roku...
-
ISCA Tutorial and Research Workshop Automatic Speech Recognition
Konferencje -
Approximate models and parameter analysis of the flow process in transmission pipelines
Publikacjathe paper deals with the problem of early leak detection in transmission pipelines. First we present the derivation of state-space equations of the flow process in the pipelines. This description is then aggregated in order to obtain a principal model. Next, the problem of process model parameterization is addressed, taking into account the maximization of a model stability margin. The location of the maximum is determined using...
-
International Conference on Image and Signal Processing
Konferencje -
International Workshop on Multimedia Signal Processing
Konferencje -
Elimination of Impulsive Disturbances From Archive Audio Signals Using Bidirectional Processing
PublikacjaIn this application-oriented paper we consider the problem of elimination of impulsive disturbances, such as clicks, pops and record scratches, from archive audio recordings. The proposed approach is based on bidirectional processing—noise pulses are localized by combining the results of forward-time and backward-time signal analysis. Based on the results of specially designed empirical tests (rather than on the results of theoretical analysis),...
-
A fast time-frequency multi-window analysis using a tuning directional kernel
PublikacjaIn this paper, a novel approach for time-frequency analysis and detection, based on the chirplet transform and dedicated to non-stationary as well as multi-component signals, is presented. Its main purpose is the estimation of spectral energy, instantaneous frequency (IF), spectral delay (SD), and chirp rate (CR) with a high time-frequency resolution (separation ability) achieved by adaptive fitting of the transform kernel. We...
-
Biometria i przetwarzanie mowy 2023
Kursy Online{mlang pl} Celem kursu jest zapoznanie studentów z: metodami ustalania i potwierdzania tożsamości ludzi na podstawie mierzalnych cech organizmu cechami mowy ludzkiej, w szczególności polskiej metodami rozpoznawania mowy metodami syntezy mowy {mlang} {mlang en} The aim of the course is to familiarize the students with: methods of identification and verification of identity of people based on measurable features of their...
-
Biometria i przetwarzanie mowy 2024
Kursy Online{mlang pl} Celem kursu jest zapoznanie studentów z: metodami ustalania i potwierdzania tożsamości ludzi na podstawie mierzalnych cech organizmu cechami mowy ludzkiej, w szczególności polskiej metodami rozpoznawania mowy metodami syntezy mowy {mlang} {mlang en} The aim of the course is to familiarize the students with: methods of identification and verification of identity of people based on measurable features of their...
-
Generalized adaptive notch smoothing revisited
PublikacjaThe problem of identification of quasi-periodically varying dynamic systems is considered. This problem can be solved using generalized adaptive notch filtering (GANF) algorithms. It is shown that the accuracy of parameter estimates can be significantly increased if the results obtained from GANF are further processed using a cascade of appropriately designed filters. The resulting generalized adaptive notch smoothing (GANS) algorithm...
-
Damage localisation in a stiffened plate structure using a propagating wave
PublikacjaThe paper presents an application of changes in propagating waves for damage detection in a stiffened aluminium plate. The experimental investigation was conducted on an aluminium plate with riveted two L-shape stiffeners. The wave has been excited with a piezoelectric transducer and measured with the Laser Scanning Doppler Vibrometer. Recorded signals were analysed using the special signal processing techniques developed for damage...
-
On DoA estimation for rotating arrays using stochastic maximum likelihood approach
PublikacjaThe flexibility needed to construct DoA estimators that can be used with rotating arrays subject to rapid variations of the signal frequency is offered by the stochastic maximum likelihood approach. Using a combination of analytic methods and Monte Carlo simulations, we show that for low and moderate source correlations the stochastic maximum likelihood estimator that assumes noncorrelated sources has accuracy comparable to the...
-
Ultrawideband transmission in physical channels: a broadband interference view
PublikacjaThe superposition of multipath components (MPC) of an emitted wave, formed by reflections from limiting surfaces and obstacles in the propagation area, strongly affects communication signals. In the case of modern wideband systems, the effect should be seen as a broadband counterpart of classical interference which is the cause of fading in narrowband systems. This paper shows that in wideband communications, the time- and frequency-domain...
-
Automatic labeling of traffic sound recordings using autoencoder-derived features
PublikacjaAn approach to detection of events occurring in road traffic using autoencoders is presented. Extensions of existing algorithms of acoustic road events detection employing Mel Frequency Cepstral Coefficients combined with classifiers based on k nearest neighbors, Support Vector Machines, and random forests are used. In our research, the acoustic signal gathered from the microphone placed near the road is split into frames and converted...
-
The instantaneous frequency rate spectogram
PublikacjaAn accelerogram of the instantaneous phase of signal components referred to as an instantaneous frequency rate spectrogram (IFRS) is presented as a joint time-frequency distribution. The distribution is directly obtained by processing the short-time Fourier transform (STFT) locally. A novel approach to amplitude demodulation based upon the reassignment method is introduced as a useful by-product. Additionally, an estimator of energy...
-
Iterative learning approach to active noise control of highly autocorrelated signals with applications to machinery noise
PublikacjaThis paper discusses the design and application of iterative learning control (ILC) and repetitive control (RC) for high modal density systems. Typical examples of these systems are structural and acoustical systems considered in active structural acoustic control (ASAC) and active noise control (ANC) applications. The application of traditional ILC and RC design techniques, which are based on a parametric system model, on systems...
-
Hybrid SONIC: joint feedforward–feedback narrowband interference canceler
PublikacjaSONIC (self-optimizing narrowband interference canceler) is an acronym of a recently proposed active noise control algorithm with interesting adaptivity and robustness properties. SONIC is a purely feedback controller, capable of rejecting nonstationary sinusoidal disturbances (with time-varying amplitude and/or frequency) in the presence of plant (secondary path) uncertainty. We show that although SONIC can work reliably without...
-
Detection of Lexical Stress Errors in Non-Native (L2) English with Data Augmentation and Attention
PublikacjaThis paper describes two novel complementary techniques that improve the detection of lexical stress errors in non-native (L2) English speech: attention-based feature extraction and data augmentation based on Neural Text-To-Speech (TTS). In a classical approach, audio features are usually extracted from fixed regions of speech such as the syllable nucleus. We propose an attention-based deep learning model that automatically de...
-
International Conference on Information, Communications and Signal Processing
Konferencje -
Localization of impulsive disturbances in audio signals using template matching
PublikacjaIn this paper, a new solution to the problem of elimination of impulsive disturbances from audio signals, based on the matched filtering technique, is proposed. The new approach stems from the observation that a large proportion of noise pulses corrupting audio recordings have highly repetitive shapes that match several typical “patterns”. In many cases a representative set of exemplary pulse waveforms can be extracted from the...
-
Zastosowanie spowalniania wypowiedzi w celu poprawy rozumienia mowy przez dzieci w szkole
PublikacjaThis paper presents a time-scale modification algorithms that could be used for hearing impairment therapy supported by real-time speech stretching. In this paper the OLA based algorithms and Phase Vocoder were described. In the experimental part usability of those algorithms for real-time speech stretching was discussed