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Wyniki wyszukiwania dla: SPEECH ANALYSIS
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Improving signal quality in speech codec using hybrid perceptual-parametric algorithm. [Poprawa jakości sygnału w kodekach mowy przy użyciu hybrydowego, parametryczno-perceptualnego algorytmu kodowania]
PublikacjaPrzedstawiono hybrydową, parametryczno-perceptualną architekturę kodeka. Podstawowa struktura kodeka parametrycznego CELP została wzbogacona o kodowanie perceptualne. Celem hybrydyzacji kodeka jest uzyskanie znaczącej poprawy subiektywnej jakości zdekodowanego sygnału. Zaproponowano dwie hybrydowe struktury. Pierwsza polega na perceptualnym kodowaniu dźwięcznych elementów sygnału rezydualnego kodeka CELP. Druga metoda dzieli sygnał...
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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...
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Discovery of Stylistic Patterns in Business Process Textual Descriptions: IT Ticket Case
PublikacjaGrowing IT complexity and related problems, which are reflected in IT tickets,create a need for new qualitative approaches. The goal isto automate the extraction of main topics described in tickets in order to provide high quality support for the IT process workers and enablea smooth service delivery to the end user. Present paper proposes a method of knowledge extraction in a form of stylistic patterns in business...
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English Language Learning Employing Developments in Multimedia IS
PublikacjaIn the realm of the development of information systems related to education, integrating multimedia technologies offers novel ways to enhance foreign language learning. This study investigates audio-video processing methods that leverage real-time speech rate adjustment and dynamic captioning to support English language acquisition. Through a mixed-methods analysis involving participants from a language school, we explore the impact...
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Automatic Emotion Recognition in Children with Autism: A Systematic Literature Review
PublikacjaThe automatic emotion recognition domain brings new methods and technologies that might be used to enhance therapy of children with autism. The paper aims at the exploration of methods and tools used to recognize emotions in children. It presents a literature review study that was performed using a systematic approach and PRISMA methodology for reporting quantitative and qualitative results. Diverse observation channels and modalities...
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Separability Assessment of Selected Types of Vehicle-Associated Noise
PublikacjaMusic Information Retrieval (MIR) area as well as development of speech and environmental information recognition techniques brought various tools in-tended for recognizing low-level features of acoustic signals based on a set of calculated parameters. In this study, the MIRtoolbox MATLAB tool, designed for music parameter extraction, is used to obtain a vector of parameters to check whether they are suitable for separation of...
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Towards More Realistic Probabilistic Models for Data Structures: The External Path Length in Tries under the Markov Model
PublikacjaTries are among the most versatile and widely used data structures on words. They are pertinent to the (internal) structure of (stored) words and several splitting procedures used in diverse contexts ranging from document taxonomy to IP addresses lookup, from data compression (i.e., Lempel- Ziv'77 scheme) to dynamic hashing, from partial-match queries to speech recognition, from leader election algorithms to distributed hashing...
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Badanie rozkładów parametrów sygnału mowy w zastosowaniach do prognozowania prawdopodobieństwa popełnienia błędów w systemach identyfikacji mówców = Examining distribution of speech signal parameters for the prognosis of error probability in speaker verification systems
PublikacjaPrzedmiotem pracy jest system identyfikacji mówców w sposób zależny od tekstu ("text dependent''). Dokonano analizy wielu różnych wypowiedzi kilkudziesięciu mówców. Zastosowana metoda parametryzacji to metoda oparta na wynikach analizy cepstralnej sygnału mowy. Zdefiniowane zostały nowe parametry skojarzone z elementarnymi zdarzeniami w procesie weryfikacji mówców. Na tej podstawie dokonano estymacji funkcji gęstości prawdopodobieństwa...
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A palatal prosthesis from archaeological research in the St Francis of Assisi church in Cracow (Poland)
PublikacjaThe hard palate is a septum that not only prevents food from entering between the oral and nasal cavity, but also plays an important role during breathing or speech. The presence of cavities within it negatively affects the comfort of life of people with this type of impairment. Hence, in the literature one can find examples of the use of hard palate prostheses to restore the separation between the nasal and oral cavity. During...
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Adaptacja akustyczna pomieszczenia wykładowego - studium przypadku
PublikacjaW niniejszej pracy przedstawiono analizę rozkładu pola akustycznego sali wykładowej znajdującej się w budynku Wydziału Elektroniki i Telekomunikacji Politechniki Gdańskiej. Badania przeprowadzono metodą pomiarową oraz symulacyjną z wykorzystaniem programu Odeon. Wybór parametrów oceny akustyki wnętrz sugerowany jest wymaganiami stawianymi pomieszczeniom lekcyjnym z zaznaczeniem multimedialnego charakteru wykładów prowadzonych...
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Investigating Feature Spaces for Isolated Word Recognition
PublikacjaMuch attention is given by researchers to the speech processing task in automatic speech recognition (ASR) over the past decades. The study addresses the issue related to the investigation of the appropriateness of a two-dimensional representation of speech feature spaces for speech recognition tasks based on deep learning techniques. The approach combines Convolutional Neural Networks (CNNs) and timefrequency signal representation...
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Marking the Allophones Boundaries Based on the DTW Algorithm
PublikacjaThe paper presents an approach to marking the boundaries of allophones in the speech signal based on the Dynamic Time Warping (DTW) algorithm. Setting and marking of allophones boundaries in continuous speech is a difficult issue due to the mutual influence of adjacent phonemes on each other. It is this neighborhood on the one hand that creates variants of phonemes that is allophones, and on the other hand it affects that the border...
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Enhanced voice user interface employing spatial filtration of signals from acoustic vector sensor
PublikacjaSpatial filtration of sound is introduced to enhance speech recognition accuracy in noisy conditions. An acoustic vector sensor (AVS) is employed. The signals from the AVS probe are processed in order to attenuate the surrounding noise. As a result the signal to noise ratio is increased. An experiment is featured in which speech signals are disturbed by babble noise. The signals before and after spatial filtration are processed...
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New Applications of Multimodal Human-Computer Interfaces
PublikacjaMultimodal computer interfaces and examples of their applications to education software and for the disabled people are presented. The proposed interfaces include the interactive electronic whiteboard based on video image analysis, application for controlling computers with gestures and the audio interface for speech stretching for hearing impaired and stuttering people. Application of the eye-gaze tracking system to awareness...
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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
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Instantaneous complex frequency for pipeline pitch estimation
PublikacjaIn the paper a pipeline algorithm for estimating the pitch of speech signal is proposed. The algorithm uses instantaneous complex frequencies estimated for four waveforms obtained by filtering the original speech signal through four bandpass complex Hilbert filters. The imaginary parts of ICFs from each channel give four candidates for pitch estimates. The decision regarding the final estimate is made based on the real parts of...
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XVIII Międzynarodowe Sympozjum Inżynierii i Reżyserii Dźwięku
PublikacjaThe subjective assessment of speech signals takes into account previous experiences and habits of an individual. Since the perception process deteriorates with age, differences should be noticeable among people from dissimilar age groups. In this work, we investigated the difference of speech quality assessment between high school students and university students. The study involved 60 participants, with 30 people in both the adolescents...
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Human voice modification using instantaneous complex frequency
PublikacjaThe paper presents the possibilities of changing human voice by modifying instantaneous complex frequency (ICF) of the speech signal. The proposed method provides a flexible way of altering voice without the necessity of finding fundamental frequency and formants' positions or detecting voiced and unvoiced fragments of speech. The algorithm is simple and fast. Apart from ICF it uses signal factorization into two factors: one fully...
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Investigating Feature Spaces for Isolated Word Recognition
PublikacjaThe study addresses the issues related to the appropriateness of a two-dimensional representation of speech signal for speech recognition tasks based on deep learning techniques. The approach combines Convolutional Neural Networks (CNNs) and time-frequency signal representation converted to the investigated feature spaces. In particular, waveforms and fractal dimension features of the signal were chosen for the time domain, and...
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Voice command recognition using hybrid genetic algorithm
PublikacjaAbstract: Speech recognition is a process of converting the acoustic signal into a set of words, whereas voice command recognition consists in the correct identification of voice commands, usually single words. Voice command recognition systems are widely used in the military, control systems, electronic devices, such as cellular phones, or by people with disabilities (e.g., for controlling a wheelchair or operating a computer...
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Trzej prorocy: Sołżenicyn, Friedman, Dugin. Część pierwsza: Sołżenicyn
PublikacjaArtykuł przedstawia na tle biograficznym dzieło i myśl profetyczną Aleksandra Sołżenicyna. Podstawą jej analizy jest mowa z okazji przyznania autorowi Oddziału chorych na raka literackiej Nagrody Nobla oraz jego wykład na temat stanu cywilizacji Zachodu wygłoszony na Uniwersytecie Harvarda – zatytułowany Zmierzch odwagi. Proroctwa Sołżenicyna dotyczące Zachodu pokazane są w kontekście jego pracy Jak odbudować Rosję? W artykule...
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Auditory-visual attention stimulator
PublikacjaNew approach to lateralization irregularities formation was proposed. The emphasis is put on the relationship between visual and auditory attention stimulation. In this approach hearing is stimulated using time scale modified speech and sight is stimulated by rendering the text of the currently heard speech. Moreover, displayed text is modified using several techniques i.e. zooming, highlighting etc. In the experimental part of...
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INVESTIGATION OF THE LOMBARD EFFECT BASED ON A MACHINE LEARNING APPROACH
PublikacjaThe Lombard effect is an involuntary increase in the speaker’s pitch, intensity, and duration in the presence of noise. It makes it possible to communicate in noisy environments more effectively. This study aims to investigate an efficient method for detecting the Lombard effect in uttered speech. The influence of interfering noise, room type, and the gender of the person on the detection process is examined. First, acoustic parameters...
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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...
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Prof. Haitham Abu-Rub - A Visit to Poland's Gdansk University of Technology
PublikacjaReport on visit of Prof. Haitham Abu-Rub in Gdansk University of Technology. Speech on the Smart Grid Centre. Visit in the new smart grid laboratory of the GUT, the Laboratory for Innovative Power Technologies and Integration of Renewable Energy Sources (LINTE^2).
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Mispronunciation Detection in Non-Native (L2) English with Uncertainty Modeling
PublikacjaA common approach to the automatic detection of mispronunciation in language learning is to recognize the phonemes produced by a student and compare it to the expected pronunciation of a native speaker. This approach makes two simplifying assumptions: a) phonemes can be recognized from speech with high accuracy, b) there is a single correct way for a sentence to be pronounced. These assumptions do not always hold, which can result...
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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...
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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...
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A comparative study of English viseme recognition methods and algorithm
PublikacjaAn elementary visual unit – the viseme is concerned in the paper in the context of preparing the feature vector as a main visual input component of Audio-Visual Speech Recognition systems. The aim of the presented research is a review of various approaches to the problem, the implementation of algorithms proposed in the literature and a comparative research on their effectiveness. In the course of the study an optimal feature vector...
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A comparative study of English viseme recognition methods and algorithms
PublikacjaAn elementary visual unit – the viseme is concerned in the paper in the context of preparing the feature vector as a main visual input component of Audio-Visual Speech Recognition systems. The aim of the presented research is a review of various approaches to the problem, the implementation of algorithms proposed in the literature and a comparative research on their effectiveness. In the course of the study an optimal feature vector construction...
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POPRAWA OBIEKTYWNYCH WSKAŹNIKÓW JAKOŚCI MOWY W WARUNKACH HAŁASU
PublikacjaCelem pracy jest modyfikacja sygnału mowy, aby uzyskać zwiększenie poprawy obiektywnych wskaźników jakości mowy po zmiksowaniu sygnału użytecznego z szumem bądź z sygnałem zakłócającym. Wykonane modyfikacje sygnału bazują na cechach mowy lombardzkiej, a w szczególności na efekcie podniesienia częstotliwości podstawowej F0. Sesja nagraniowa obejmowała zestawy słów i zdań w języku polskim, nagrane w warunkach ciszy, jak również w...
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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...
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Evaluation Criteria for Affect-Annotated Databases
PublikacjaIn this paper a set of comprehensive evaluation criteria for affect-annotated databases is proposed. These criteria can be used for evaluation of the quality of a database on the stage of its creation as well as for evaluation and comparison of existing databases. The usefulness of these criteria is demonstrated on several databases selected from affect computing domain. The databases contain different kind of data: video or still...
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Minimizing Distribution and Data Loading Overheads in Parallel Training of DNN Acoustic Models with Frequent Parameter Averaging
PublikacjaIn the paper we investigate the performance of parallel deep neural network training with parameter averaging for acoustic modeling in Kaldi, a popular automatic speech recognition toolkit. We describe experiments based on training a recurrent neural network with 4 layers of 800 LSTM hidden states on a 100-hour corpora of annotated Polish speech data. We propose a MPI-based modification of the training program which minimizes the...
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Vocalic Segments Classification Assisted by Mouth Motion Capture
PublikacjaVisual features convey important information for automatic speech recognition (ASR), especially in noisy environment. The purpose of this study is to evaluate to what extent visual data (i.e. lip reading) can enhance recognition accuracy in the multi-modal approach. For that purpose motion capture markers were placed on speakers' faces to obtain lips tracking data during speaking. Different parameterizations strategies were tested...
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Modeling and Simulation for Exploring Power/Time Trade-off of Parallel Deep Neural Network Training
PublikacjaIn the paper we tackle bi-objective execution time and power consumption optimization problem concerning execution of parallel applications. We propose using a discrete-event simulation environment for exploring this power/time trade-off in the form of a Pareto front. The solution is verified by a case study based on a real deep neural network training application for automatic speech recognition. A simulation lasting over 2 hours...
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A Device for Measuring Auditory Brainstem Responses to Audio
PublikacjaStandard ABR devices use clicks and tone bursts to assess subjects’ hearing in an objective way. A new device was developed that extends the functionality of a standard ABR audiometer by collecting and analyzing auditory brainstem responses (ABR). The developed accessory allows for the use of complex sounds (e.g., speech or music excerpts) as stimuli. Therefore, it is possible to find out how efficiently different types of sounds...
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Secured wired BPL voice transmission system
PublikacjaDesigning a secured voice transmission system is not a trivial task. Wired media, thanks to their reliability and resistance to mechanical damage, seem an ideal solution. The BPL (Broadband over Power Line) cable is resistant to electricity stoppage and partial damage of phase conductors, ensuring continuity of transmission in case of an emergency. It seems an appropriate tool for delivering critical data, mostly clear and understandable...
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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),...
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Chirp Rate and Instantaneous Frequency Estimation: Application to Recursive Vertical Synchrosqueezing
PublikacjaThis letter introduces new chirp rate and instantaneous frequency estimators designed for frequency-modulated signals. These estimators are first investigated from a deterministic point of view, then compared together in terms of statistical efficiency. They are also used to design new recursive versions of the vertically synchrosqueezed short-time Fourier transform, using a previously published method (D. Fourer, F. Auger, and...
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The Innovative Faculty for Innovative Technologies
PublikacjaA leaflet describing Faculty of Electronics, Telecommunications and Informatics, Gdańsk University of Technology. Multimedia Systems Department described laboratories and prototypes of: Auditory-visual attention stimulator, Automatic video event detection, Object re-identification application for multi-camera surveillance systems, Object Tracking and Automatic Master-Slave PTZ Camera Positioning System, Passive Acoustic Radar,...
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Determining Pronunciation Differences in English Allophones Utilizing Audio Signal Parameterization
PublikacjaAn allophonic description of English plosive consonants, based on audio-visual recordings of 600 specially selected words, was developed. First, several speakers were recorded while reading words from a teleprompter. Then, every word was played back from the previously recorded sample read by a phonology expert and each examined speaker repeated a particular word trying to imitate correct pronunciation. The next step consisted...
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Automatic music signal mixing system based on one-dimensional Wave-U-Net autoencoders
PublikacjaThe purpose of this paper is to show a music mixing system that is capable of automatically mixing separate raw recordings with good quality regardless of the music genre. This work recalls selected methods for automatic audio mixing first. Then, a novel deep model based on one-dimensional Wave-U-Net autoencoders is proposed for automatic music mixing. The model is trained on a custom-prepared database. Mixes created using the...
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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...
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Detection and localization of selected acoustic events in 3D acoustic field for smart surveillance applications
PublikacjaA method for automatic determination of position of chosen sound events such as speech signals and impulse sounds in 3-dimensional space is presented. The events are localized in the presence of sound reflections employing acoustic vector sensors. Human voice and impulsive sounds are detected using adaptive detectors based on modified peak-valley difference (PVD) parameter and sound pressure level. Localization based on signals...
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Detection and localization of selected acoustic events in acoustic field for smart surveillance applications
PublikacjaA method for automatic determination of position of chosen sound events such as speech signals and impulse sounds in 3-dimensional space is presented. The evens are localized in the presence of sound reflections employing acoustic vector sensors. Human voice and impulsive sounds are detected using adaptive detectors based on modified peak-valley difference (PVD) parameter and sound pressure level. Localization based on signals...
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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...
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Impact of the glazed roof on acoustics of historic interiors
PublikacjaThe paper discusses the adverse acoustic phenomena occurring in the semi-open interiors (courtyards, yards) covered with a glass roof. Particularly negative is the rever-beration noise, which leads to the degradation of the utility functions of the resulting spaces. It involves the drastically reducing the intelligibility of speech, loss of natural sounding of music, problems with the sound system, as well as disturbances in the...
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Subjective and Objective Comparative Study of DAB+ Broadcast System
PublikacjaBroadcasting services seek to optimize their use of bandwidth in order to maximize user’s quality of experience. They aim to transmit high-quality digital speech and music signals at the lowest bitrate. They intend to offer the best quality under available conditions. Due to bandwidth limitations, audio quality is in conflict with the number of transmitted radio programs. This paper analyzes whether the quality of real-time digital...
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Highlighting interlanguage phoneme differences based on similarity matrices and convolutional neural network
PublikacjaThe goal of this research is to find a way of highlighting the acoustic differences between consonant phonemes of the Polish and Lithuanian languages. For this purpose, similarity matrices are employed based on speech acoustic parameters combined with a convolutional neural network (CNN). In the first experiment, we compare the effectiveness of the similarity matrices applied to discerning acoustic differences between consonant...