Search results for: SPEECH SIGNAL PARAMETERIZATION - Bridge of Knowledge

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Search results for: SPEECH SIGNAL PARAMETERIZATION

Search results for: SPEECH SIGNAL PARAMETERIZATION

  • Speech Intelligibility Measurements in Auditorium

    Publication

    Speech 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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  • Tensor Decomposition for Imagined Speech Discrimination in EEG

    Publication

    - LECTURE NOTES IN COMPUTER SCIENCE - Year 2018

    Most 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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  • Language Models in Speech Recognition

    Publication

    - Year 2022

    This 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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  • Investigating Noise Interference on Speech Towards Applying the Lombard Effect Automatically

    Publication

    - Year 2022

    The 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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  • MECHANICAL SYSTEMS AND SIGNAL PROCESSING

    Journals

    ISSN: 0888-3270

  • Applying the Lombard Effect to Speech-in-Noise Communication

    Publication

    - Electronics - Year 2023

    This 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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  • Constructing a Dataset of Speech Recordingswith Lombard Effect

    Publication

    - Year 2020

    Thepurpose 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...

  • SIGNAL PROCESSING

    Journals

    ISSN: 0165-1684 , eISSN: 1872-7557

  • Structures for parameterization, meshing and data exchange of topologically related surfaces of a ship hull

    Publication

    - Year 2010

    This paper presents proposal of data structures for storage and processing of a parametric three-dimensional model of a midship hull sections. The model consists of coarse surfaces like: decks, frames, girders, stiffeners, brackets, partitions etc. bounded by topological relations. All workshop details are omitted as the model is intended for numeric calculations. Proposed data structures are prepared to facilitate changes in the...

  • Distortion of speech signals in the listening area: its mechanism and measurements

    Publication

    - Year 2014

    The 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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  • A Study of Cross-Linguistic Speech Emotion Recognition Based on 2D Feature Spaces

    Publication
    • G. Tamulevicius
    • G. Korvel
    • A. B. Yayak
    • P. Treigys
    • J. Bernataviciene
    • B. Kostek

    - Electronics - Year 2020

    In this research, a study of cross-linguistic speech emotion recognition is performed. For this purpose, emotional data of different languages (English, Lithuanian, German, Spanish, Serbian, and Polish) are collected, resulting in a cross-linguistic speech emotion dataset with the size of more than 10.000 emotional utterances. Despite the bi-modal character of the databases gathered, our focus is on the acoustic representation...

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  • Radio Reception Signal

    e-Learning Courses
    • A. Marczak
    • M. Gajewska

  • Digital Signal Processing

    e-Learning Courses
    • T. Stefański

  • A Novel Method for Intelligibility Assessment of Nonlinearly Processed Speech in Spaces Characterized by Long Reverberation Times

    Publication

    Objective assessment of speech intelligibility is a complex task that requires taking into account a number of factors such as different perception of each speech sub-bands by the human hearing sense or different physical properties of each frequency band of a speech signal. Currently, the state-of-the-art method used for assessing the quality of speech transmission is the speech transmission index (STI). It is a standardized way...

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  • Speech Analytics Based on Machine Learning

    Publication

    In 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 synthesis controlled by eye gazing

    Publication

    - Year 2010

    A 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...

  • Acoustic Sensing Analytics Applied to Speech in Reverberation Conditions

    Publication

    The paper aims to discuss a case study of sensing analytics and technology in acoustics when applied to reverberation conditions. Reverberation is one of the issues that makes speech in indoor spaces challenging to understand. This problem is particularly critical in large spaces with few absorbing or diffusing surfaces. One of the natural remedies to improve speech intelligibility in such conditions may be achieved through speaking...

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  • Transfer learning in imagined speech EEG-based BCIs

    Publication

    - Biomedical Signal Processing and Control - Year 2019

    The 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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  • Elimination of clicks from archive speech signals using sparse autoregressive modeling

    Publication

    This paper presents a new approach to elimination of impulsivedisturbances from archive speech signals. The proposedsparse autoregressive (SAR) signal representation is given ina factorized form - the model is a cascade of the so-called formantfilter and pitch filter. Such a technique has been widelyused in code-excited linear prediction (CELP) systems, as itguarantees model stability. After detection of noise pulses usinglinear...

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  • Weakly-Supervised Word-Level Pronunciation Error Detection in Non-Native English Speech

    Publication
    • D. Korzekwa
    • J. Lorenzo-trueba
    • T. Drugman
    • S. Calamaro
    • B. Kostek

    - Year 2021

    We propose a weakly-supervised model for word-level mispronunciation detection in non-native (L2) English speech. To train this model, phonetically transcribed L2 speech is not required and we only need to mark mispronounced words. The lack of phonetic transcriptions for L2 speech means that the model has to learn only from a weak signal of word-level mispronunciations. Because of that and due to the limited amount of mispronounced...

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  • Interpretable Deep Learning Model for the Detection and Reconstruction of Dysarthric Speech

    Publication
    • D. Korzekwa
    • R. Barra-Chicote
    • B. Kostek
    • T. Drugman
    • M. Łajszczak

    - Year 2019

    We 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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  • Improvement of speech intelligibility in the presence of noise interference using the Lombard effect and an automatic noise interference profiling based on deep learning

    Publication
    • K. Kąkol

    - Year 2023

    The Lombard effect is a phenomenon that results in speech intelligibility improvement when applied to noise. There are many distinctive features of Lombard speech that were recalled in this dissertation. This work proposes the creation of a system capable of improving speech quality and intelligibility in real-time measured by objective metrics and subjective tests. This system consists of three main components: speech type detection,...

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  • Material for Automatic Phonetic Transcription of Speech Recorded in Various Conditions

    Publication

    Automatic speech recognition (ASR) is under constant development, especially in cases when speech is casually produced or it is acquired in various environment conditions, or in the presence of background noise. Phonetic transcription is an important step in the process of full speech recognition and is discussed in the presented work as the main focus in this process. ASR is widely implemented in mobile devices technology, but...

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  • A study on signal processing methods applied to hearing aids

    Publication

    - Year 2016

    This paper presents a short survey on current technology available in hearing aids with a focus on digital signal processing techniques used. First, factors influencing the hearing aid effectiveness are introduced. Then, examples of the present DSP methods and strategies are provided. Also, a description of current limitations of hearing aids and future trends of development are shown. Finally, the notion of computational auditory...

  • An audio-visual corpus for multimodal automatic speech recognition

    review of available audio-visual speech corpora and a description of a new multimodal corpus of English speech recordings is provided. The new corpus containing 31 hours of recordings was created specifically to assist audio-visual speech recognition systems (AVSR) development. The database related to the corpus includes high-resolution, high-framerate stereoscopic video streams from RGB cameras, depth imaging stream utilizing Time-of-Flight...

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  • Intra-subject class-incremental deep learning approach for EEG-based imagined speech recognition

    Publication

    - Biomedical Signal Processing and Control - Year 2023

    Brain–computer interfaces (BCIs) aim to decode brain signals and transform them into commands for device operation. The present study aimed to decode the brain activity during imagined speech. The BCI must identify imagined words within a given vocabulary and thus perform the requested action. A possible scenario when using this approach is the gradual addition of new words to the vocabulary using incremental learning methods....

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  • IEEE TRANSACTIONS ON SIGNAL PROCESSING

    Journals

    ISSN: 1053-587X , eISSN: 1941-0476

  • Ranking Speech Features for Their Usage in Singing Emotion Classification

    Publication

    - Year 2020

    This paper aims to retrieve speech descriptors that may be useful for the classification of emotions in singing. For this purpose, Mel Frequency Cepstral Coefficients (MFCC) and selected Low-Level MPEG 7 descriptors were calculated based on the RAVDESS dataset. The database contains recordings of emotional speech and singing of professional actors presenting six different emotions. Employing the algorithm of Feature Selection based...

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  • Classifying type of vehicles on the basis of data extracted from audio signal characteristics

    The aim of this study is to find and optimize a feature vector for an automatic recognition of the type of vehicles, extracted form an audio signal. First, the influence of weather-based conditions of road surface on spectral characteristic of the audio signal recorded from a passing vehicle in close proximity to the road is discussed. Next, parameterization of the recorded audio signal is performed. For that purpose, the MIRtoolbox,...

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  • System Supporting Speech Perception in Special Educational Needs Schoolchildren

    Publication

    - Year 2012

    The system supporting speech perception during the classes is presented in the paper. The system is a combination of portable device, which enables real-time speech stretching, with the workstation designed in order to perform hearing tests. System was designed to help children suffering from Central Auditory Processing Disorders.

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  • Silence/noise detection for speech and music signals

    Publication

    - Year 2008

    This 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.

  • Virtual keyboard controlled by eye gaze employing speech synthesis

    Publication

    - Year 2010

    The article presents the speech synthesis integrated into the eye gaze tracking system. This approach can significantly improve the quality of life of physically disabled people who are unable to communicate. The virtual keyboard (QWERTY) is an interface which allows for entering the text for the speech synthesizer. First, this article describes a methodology of determining the fixation point on a computer screen. Then it presents...

  • Virtual Keyboard controlled by eye gaze employing speech synthesis

    The article presents the speech synthesis integrated into the eye gaze tracking system. This approach can significantly improve the quality of life of physically disabled people who are unable to communicate. The virtual keyboard (QWERTY) is an interface which allows for entering the text for the speech synthesizer. First, this article describes a methodology of determining the fixation point on a computer screen. Then it presents...

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  • Corrupted speech intelligibility improvement using adaptive filter based algorithm

    Publication

    A 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.

  • Comparison of the exponential thermal transient parameterization methods with the SMTP method in the unipedicled DIEP flap computer modelling and simulation

    Publication

    - QIRT Journal - Year 2018

    The aim of this paper is to compare the spatial contrast of the image descriptors obtained via three different thermal transient parameterization methods in Active Dynamic Thermography. The thermal constants and amplitude values of the one- and two- exponential parametrization are compared to the Simplified Magnitude-Temporal Parametrization method (SMTP). The comparison is performed using the data obtained by simulating the cold...

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  • SYNTHESIZING MEDICAL TERMS – QUALITY AND NATURALNESS OF THE DEEP TEXT-TO-SPEECH ALGORITHM

    The main purpose of this study is to develop a deep text-to-speech (TTS) algorithm designated for an embedded system device. First, a critical literature review of state-of-the-art speech synthesis deep models is provided. The algorithm implementation covers both hardware and algorithmic solutions. The algorithm is designed for use with the Raspberry Pi 4 board. 80 synthesized sentences were prepared based on medical and everyday...

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  • A survey of automatic speech recognition deep models performance for Polish medical terms

    Among the numerous applications of speech-to-text technology is the support of documentation created by medical personnel. There are many available speech recognition systems for doctors. Their effectiveness in languages such as Polish should be verified. In connection with our project in this field, we decided to check how well the popular speech recognition systems work, employing models trained for the general Polish language....

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  • A non-uniform real-time speech time-scale stretching method

    Publication

    An 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...

  • Building Knowledge for the Purpose of Lip Speech Identification

    Consecutive 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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  • Subjective Quality Evaluation of Speech Signals Transmitted via BPL-PLC Wired System

    Publication
    • P. Falkowski-Gilski
    • G. Debita
    • M. Habrych
    • B. Miedziński
    • P. Jedlikowski
    • B. Polnik
    • J. Wandzio
    • X. Wang

    - Year 2020

    The broadband over power line – power line communication (BPL-PLC) cable is resistant to electricity stoppage and partial damage of phase conductors. It maintains continuity of transmission in case of an emergency. These features make it an ideal solution for delivering data, e.g. in an underground mine environment, especially clear and easily understandable voice messages. This paper describes a subjective quality evaluation of...

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  • Noise profiling for speech enhancement employing machine learning models

    Publication

    - Journal of the Acoustical Society of America - Year 2022

    This 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...

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  • Emotions in polish speech recordings

    Open Research Data
    open access

    The data set presents emotions recorded in sound files that are expressions of Polish speech. Statements were made by people aged 21-23, young voices of 5 men. Each person said the following words / nie – no, oddaj - give back, podaj – pass, stop - stop, tak - yes, trzymaj -hold / five times representing a specific emotion - one of three - anger (a),...

  • Communication Platform for Evaluation of Transmitted Speech Quality

    A 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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  • Results of tests on speech intelligibility in reverberant conditions

    Open Research Data

    The dataset contains the results of tests that aimed to provide a relationship between the rate of speech (RoS) and reverberation conditions characterized by the Speech Transmission Index (STI).

  • Automated detection of pronunciation errors in non-native English speech employing deep learning

    Publication

    - Year 2023

    Despite significant advances in recent years, the existing Computer-Assisted Pronunciation Training (CAPT) methods detect pronunciation errors with a relatively low accuracy (precision of 60% at 40%-80% recall). This Ph.D. work proposes novel deep learning methods for detecting pronunciation errors in non-native (L2) English speech, outperforming the state-of-the-art method in AUC metric (Area under the Curve) by 41%, i.e., from...

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  • Quality Evaluation of Speech Transmission via Two-way BPL-PLC Voice Communication System in an Underground Mine

    Publication

    - Archives of Acoustics - Year 2023

    In order to design a stable and reliable voice communication system, it is essential to know how many resources are necessary for conveying quality content. These parameters may include objective quality of service (QoS) metrics, such as: available bandwidth, bit error rate (BER), delay, latency as well as subjective quality of experience (QoE) related to user expectations. QoE is expressed as clarity of speech and the ability...

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  • Digital Signal Processing - 22/23

    e-Learning Courses
    • T. Stefański

     Po ukończeniu kursu, student projektuje podstawowe algorytmy cyfrowego przetwarzania sygnałów - filtrów cyfrowych FIR i IIR, i estymuje widmo za pomocą FFT.Student opisuje architektury i ścieżki danych procesorów stało-przecinkowych i zmienno-przecinkowych. Student tłumaczy podstawy arytmetyki procesorów i podaje przykłady zastosowań.

  • Digital Signal Processing-23/24

    e-Learning Courses

     Po ukończeniu kursu, student projektuje podstawowe algorytmy cyfrowego przetwarzania sygnałów - filtrów cyfrowych FIR i IIR, i estymuje widmo za pomocą FFT.Student opisuje architektury i ścieżki danych procesorów stało-przecinkowych i zmienno-przecinkowych. Student tłumaczy podstawy arytmetyki procesorów i podaje przykłady zastosowań.

  • The Use of Wavelet Analysis to Denoising of Electrocardiography Signal

    Publication

    The electrocardiography examination, due to its accessibility and simplicity, has an important role in diagnostics of the heart ailments. It enables quick detection of various heart defects, undetectable by other kinds of diagnostic tools, so it is very popular. Nevertheless, the measured signal is exposed to a different disturbances. Among them, the electromagnetic interferences, drift of reference electrode and high frequency...

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  • Mowa nienawiści (hate speech) a odpowiedzialność dostawców usług internetowych w orzecznictwie sądów europejskich

    Publication

    - Year 2015

    The article analyses the phenomenon of hate speech in the Internet contrasted with the problem of responsability of Internet Service Providers for cases of such abuses of freedom of expression. The text provides an analysis of jurisprudence of two European Courts. On the one hand it presents the position of the European Court of Human Rights on the problem of hate speech: its definition and the liability for it as an exception...