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Wyniki wyszukiwania dla: speech synthesis

Wyniki wyszukiwania dla: speech synthesis

  • Speech synthesis controlled by eye gazing

    Publikacja

    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...

  • 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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  • Virtual keyboard controlled by eye gaze employing speech synthesis

    Publikacja

    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...

  • An Attempt to Create Speech Synthesis Model That Retains Lombard Effect Characteristics

    Publikacja

    - Rok 2019

    The speech with the Lombard effect has been extensively studied in the context of speech recognition or speech enhancement. However, few studies have investigated the Lombard effect in the context of speech synthesis. The aim of this paper is to create a mathematical model that allows for retaining the Lombard effect. These models could be used as a basis of a formant speech synthesizer. The proposed models are based on dividing...

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  • Computer-assisted pronunciation training—Speech synthesis is almost all you need

    Publikacja

    - SPEECH COMMUNICATION - Rok 2022

    The research community has long studied computer-assisted pronunciation training (CAPT) methods in non-native speech. Researchers focused on studying various model architectures, such as Bayesian networks and deep learning methods, as well as on the analysis of different representations of the speech signal. Despite significant progress in recent years, existing CAPT methods are not able to detect pronunciation errors with high...

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  • Evaluation of Lombard Speech Models in the Context of Speech in Noise Enhancement

    Publikacja

    - IEEE Access - Rok 2020

    The 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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  • Time-domain prosodic modifications for text-to-speech synthesizer

    Publikacja

    - Rok 2010

    An 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.

  • High quality speech codec employing sines+noise+transients model

    A method of high quality wideband speech signal representation employing sines+transients+noise model is presented. The need for a wideband speech coding approach as well as various methods for analysis and synthesis of sines, residual and transient states of speech signal is discussed. The perceptual criterion is applied in the proposed approach during encoding of sines amplitudes in order to reduce bandwidth requirements and...

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  • Speech Intelligibility Measurements in Auditorium

    Publikacja

    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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  • Estimation of the excitation variances of speech and noise AR-models for enhanced speech coding

    Publikacja

    - Rok 2001

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  • Transient detection for speech coding applications

    Signal 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

    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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  • Intelligent processing of stuttered speech.

    W 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.

  • Improving the quality of speech in the conditions of noise and interference

    Publikacja

    The 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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  • Constructing a Dataset of Speech Recordingswith Lombard Effect

    Publikacja

    - Rok 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...

  • Improved method for real-time speech stretching

    Publikacja

    n 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

    A 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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  • Methodology and technology for the polymodal allophonic speech transcription

    A 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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  • Real-time speech-rate modification experiments

    Publikacja

    An 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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  • Comparison of Language Models Trained on Written Texts and Speech Transcripts in the Context of Automatic Speech Recognition

    Publikacja
    • S. Dziadzio
    • A. Nabożny
    • A. Smywiński-Pohl
    • B. Ziółko

    - Rok 2015

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  • Multimodal English corpus for automatic speech recognition

    A 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...

  • Silence/noise detection for speech and music signals

    Publikacja

    - Rok 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.

  • 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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  • 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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  • Integration of speech enhancement and coding techniques

    Publikacja

    - Rok 1999

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  • Novel approaches to wideband speech coding

    Publikacja

    Dwie metoda kodowania szerokopasmowego mowy zostały zaprezentowane. W pierwszej metodzie wykorzystano algorytm kompresji i ekspansji czasowej sygnału mowy, pozwalający na kodowanie szerokopasmowe sygnału mowy z wykorzystaniem ustandaryzowanych kodeków. Metoda ta jest przewidziana do zastosowania w adaptacyjnych algorytmach kodowania mowy. Drugie z proponowanych rozwiazan dotyczy nowej metody estymacji obwiedni widma sygnalu mowy...

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  • Broadband interference in speech reinforcement systems

    Publikacja

    - Rok 2008

    Artykuł 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...

  • A system for multitask noisy speech enhancement.

    Publikacja

    - Rok 2004

    W 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...

  • Multitask Noisy Speech Enhancement System

    Publikacja

    - Rok 2005

    W 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...

  • Improving Objective Speech Quality Indicators in Noise Conditions

    Publikacja

    - Rok 2020

    This 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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  • Acoustic Sensing Analytics Applied to Speech in Reverberation Conditions

    Publikacja

    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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  • A Method of Real-Time Non-uniform Speech Stretching

    Publikacja

    Developed 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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  • Introduction to the special issue on machine learning in acoustics

    Publikacja
    • Z. Michalopoulou
    • P. Gerstoft
    • B. Kostek
    • M. A. Roch

    - Journal of the Acoustical Society of America - Rok 2021

    When 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...

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  • Examining Influence of Distance to Microphone on Accuracy of Speech Recognition

    The 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

    The 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...

  • 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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  • Automatic prosodic modification in a Text-To-Speech synthesizer of Polish language

    Przedstawiono system syntezy mowy polskiej z funkcją automatycznej modyfikacji prozodii wypowiedzi. Opisane zostały metody automatycznego wyznaczania akcentu i intonacji wypowiedzi. Przedstawiono zastosowanie algorytmów przetwarzania sygnału mowy w procesie kształtowania prozodii. Omówiono wpływ zastosowanych modyfikacji na naturalność brzmienia syntezowanego sygnału. Zastosowana metoda oparta jest na algorytmie TD-PSOLA. Opracowany...

  • Ranking Speech Features for Their Usage in Singing Emotion Classification

    Publikacja

    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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  • WYKORZYSTANIE SIECI NEURONOWYCH DO SYNTEZY MOWY WYRAŻAJĄCEJ EMOCJE

    Publikacja

    W niniejszym artykule przedstawiono analizę rozwiązań do rozpoznawania emocji opartych na mowie i możliwości ich wykorzystania w syntezie mowy z emocjami, wykorzystując do tego celu sieci neuronowe. Przedstawiono aktualne rozwiązania dotyczące rozpoznawania emocji w mowie i metod syntezy mowy za pomocą sieci neuronowych. Obecnie obserwuje się znaczny wzrost zainteresowania i wykorzystania uczenia głębokiego w aplikacjach związanych...

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

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

    - Rok 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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  • Visual Lip Contour Detection for the Purpose of Speech Recognition

    Publikacja

    A 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...

  • Speech codec enhancements utilizing time compression and perceptual coding

    Publikacja

    A method for encoding wideband speech signal employing standardized narrowband speech codecs is presented as well as experimental results concerning detection of tonal spectral components. The speech signal sampled with a higher sampling rate than it is suitable for narrowband coding algorithm is compressed in order to decrease the amount of samples. Next, the time-compressed representation of a signal is encoded using a narrowband...

  • Methods of Improving Speech Intelligibility for Listeners with Hearing Resolution Deficit

    Methods developed for real-time time scale modification (TSM) of speech signal are presented. They are based onthe non-uniform, speech rate depended SOLA algorithm (Synchronous Overlap and Add). Influence of theproposed method on the intelligibility of speech was investigated for two separate groups of listeners, i.e. hearingimpaired children and elderly listeners. It was shown that for the speech with average rate equal to or...

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

    Publikacja

    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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  • Transient detection algorithms for speech coding applications

    Publikacja

    - Journal of the Acoustical Society of America - Rok 2006

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  • New generation speech aid for stuttering people

    Publikacja

    - Rok 2008

    Współczesne 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

    Publikacja

    Współczesne 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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  • Influence of modulation detection threshold on speech intelligibility

    Publikacja

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  • Speech recognition system for hearing impaired people.

    Publikacja

    - Rok 2005

    Praca 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...

  • System Supporting Speech Perception in Special Educational Needs Schoolchildren

    Publikacja

    - Rok 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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