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Search results for: SPEECH
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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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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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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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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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Comparison of Language Models Trained on Written Texts and Speech Transcripts in the Context of Automatic Speech Recognition
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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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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 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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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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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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Integration of speech enhancement and coding techniques
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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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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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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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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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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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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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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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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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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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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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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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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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Comprehensive Evaluation of Statistical Speech Waveform Synthesis
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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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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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Influence of modulation detection threshold on speech intelligibility
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Improving Objective Speech Quality Indicators in Noise Conditions
PublicationThis work aims at modifying speech signal samples and test them with objective speech quality indicators after mixing the original signals with noise or with an interfering signal. Modifications that are applied to the signal are related to the Lombard speech characteristics, i.e., pitch shifting, utterance duration changes, vocal tract scaling, manipulation of formants. A set of words and sentences in Polish, recorded in silence,...
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Detecting Lombard Speech Using Deep Learning Approach
PublicationRobust Lombard speech-in-noise detecting is challenging. This study proposes a strategy to detect Lombard speech using a machine learning approach for applications such as public address systems that work in near real time. The paper starts with the background concerning the Lombard effect. Then, assumptions of the work performed for Lombard speech detection are outlined. The framework proposed combines convolutional neural networks...
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Acoustic Sensing Analytics Applied to Speech in Reverberation Conditions
PublicationThe 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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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.
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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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Examining Influence of Distance to Microphone on Accuracy of Speech Recognition
PublicationThe problem of controlling a machine by the distant-talking speaker without a necessity of handheld or body-worn equipment usage is considered. A laboratory setup is introduced for examination of performance of the developed automatic speech recognition system fed by direct and by distant speech acquired by microphones placed at three different distances from the speaker (0.5 m to 1.5 m). For feature extraction from the voice signal...
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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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An audio-visual corpus for multimodal automatic speech recognition
Publicationreview 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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Ranking Speech Features for Their Usage in Singing Emotion Classification
PublicationThis 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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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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Transfer learning in imagined speech EEG-based BCIs
PublicationThe 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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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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Evaluation and Irony in Text in the Light of Speech Act Theory
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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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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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