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Search results for: AUDIO SIGNALS
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Detection of impulsive disturbances in archive audio signals
PublicationIn this paper the problem of detection of impulsive disturbances in archive audio signals is considered. It is shown that semi-causal/noncausal solutions based on joint evaluation of signal prediction errors and leave-one-out signal interpolation errors, allow one to noticeably improve detection results compared to the prediction-only based solutions. The proposed approaches are evaluated on a set of clean audio signals contaminated...
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Adaptive filter for reconstruction of stereo audio signals.
PublicationArtykuł poświęcony jest omówieniu metody rekonstrukcji zakłóconych impulsowo sygnałów stereofonicznych. W pracy zdefiniowano model sygnału stereofonicznego i przedstawiono zaprojektowany dla tego modelu filtr Kalmana. Przedstawiono modyfikacje filtru, w wyniku których algorytm dokonuje rekonstrukcji zakłóconego impulsowo sygnału w jednym kanale z wykorzystaniem dodatkowej informacji zawartej w niezakłóconych próbkach sygnału pochodzącego...
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Localization of impulsive disturbances in audio signals using template matching
PublicationIn this paper, a new solution to the problem of elimination of impulsive disturbances from audio signals, based on the matched filtering technique, is proposed. The new approach stems from the observation that a large proportion of noise pulses corrupting audio recordings have highly repetitive shapes that match several typical “patterns”. In many cases a representative set of exemplary pulse waveforms can be extracted from the...
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Two-stage method of impulsive noise detection for audio signals
PublicationPrzedstawiono nowa dwuetapową metodę detekcji zakłóceń impulsowych opartą na analizie funkcji gęstości rozkładu prawdopodobieństwa zakłóconego sygnału. Opisano algorytm określania poziomu wyzwalania detektora progowego.
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Localization of impulsive disturbances in archive audio signals using predictive matched filtering
PublicationThe problem of elimination of impulsive disturbances from archive audio signals is considered and its new solution, called predictive matched filtering, is proposed. The new approach is based on the observation that a large percentage of noise pulses corrupting archive audio recordings have highly repetitive shapes that match several typical “patterns”, called click templates. To localize noise pulses, click templates can be correlated...
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Intelligent acquisition of audio signals, employing neutral networks and rough set algorithms
PublicationAlgorytmy oparte na sztucznych sieciach neuronowych i metodzie zbiorówprzybliżonych zostały zastosowane do lokalizacji sygnałów fonicznych obar-czonych pasożytniczym szumem i rewerberacjami. Informacja o kierunku napły-wania dźwięku była uzyskiwana na wyjściach tych algorytmów na podstawie re-prezentacji parametrycznej. Przedstawiono wyniki eksperymentalne i przepro-wadzono ich dyskusję.
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Elimination of Impulsive Disturbances From Archive Audio Signals Using Bidirectional Processing
PublicationIn 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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Parametric impulsive noise detector for corrupted audio signals based on hidden Markow model
PublicationThe paper addresses the problem of impulsive noise detection for audio signals. A structure of threshold parameter detectors using modelingof signals was introduced. the algorithm of the noise detection, based on discrete-time hidden Markow model (HMM)of whitened audio signal is elaborated
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Sparse vector autoregressive modeling of audio signals and its application to the elimination of impulsive disturbances
PublicationArchive audio files are often corrupted by impulsive disturbances, such as clicks, pops and record scratches. This paper presents a new method for elimination of impulsive disturbances from stereo audio signals. The proposed approach is based on a sparse vector autoregressive signal model, made up of two components: one taking care of short-term signal correlations, and the other one taking care of long-term correlations. The method...
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Using concentrated spectrogram for analysis of audio acoustic signals
PublicationThe paper presents results of time-frequency analysis of audio acoustic signals using the method of Concentrated Spectrograph also known as ''Cross-spectral method'' or ''Reassignment method''. Presented algorithm involves signal's local group delay and channelized instantaneous frequency to relevantly redistribute all Short-time Fourier transform lines in time-frequency plain. The main intention of the paper is to compare various...
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New semi-causal and noncausal techniques for detection of impulsive disturbances in multivariate signals with audio applications
PublicationThis paper deals with the problem of localization of impulsive disturbances in nonstationary multivariate signals. Both unidirectional and bidirectional (noncausal) detection schemes are proposed. It is shown that the strengthened pulse detection rule, which combines analysis of one-step-ahead signal prediction errors with critical evaluation of leave-one-out signal interpolation errors, allows one to noticeably improve detection results...
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Piotr Odya dr inż.
PeoplePiotr Odya was born in Gdansk in 1974. He received his M.Sc. in 1999 from the Faculty of Electronics, Telecommunications and Informatics, Gdansk University of Technology, Poland. His thesis was related to the problem of sound quality improvement in the contemporary broadcasting studio. He is interested in video editing and multichannel sound systems. The goal of Mr. Odya Ph.D. thesis concerned methods and algorithms for correcting...
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Elimination of impulsive disturbances from stereo audio recordings
PublicationThis paper presents a new approach to elimination of impulsive disturbances from stereo audio recordings. The proposed solution is based on vector autoregressive modeling of audio signals. On-line tracking of signal model parameters is performed using the stability-preserving Whittle-Wiggins-Robinson algorithm with exponential data weighting. Detection of noise pulses and model-based interpolation of the irrevocably distorted samples...
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Using Physiological Signals for Emotion Recognition
PublicationRecognizing user’s emotions is the promising area of research in a field of human-computer interaction. It is possible to recognize emotions using facial expression, audio signals, body poses, gestures etc. but physiological signals are very useful in this field because they are spontaneous and not controllable. In this paper a problem of using physiological signals for emotion recognition is presented. The kinds of physiological...
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Elimination of Impulsive Disturbances From Stereo Audio Recordings Using Vector Autoregressive Modeling and Variable-order Kalman Filtering
PublicationThis paper presents a new approach to elimination of impulsive disturbances from stereo audio recordings. The proposed solution is based on vector autoregressive modeling of audio signals. Online tracking of signal model parameters is performed using the exponential ly weighted least squares algo- rithm. Detection of noise pulses an d model-based interpolation of the irrevocably distorted sampl es is realized using an adaptive, variable-order...
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EXAMINING INFLUENCE OF VIDEO FRAMERATE AND AUDIO/VIDEO SYNCHRONIZATION ON AUDIO-VISUAL SPEECH RECOGNITION ACCURACY
PublicationThe problem of video framerate and audio/video synchronization in audio-visual speech recogni-tion is considered. The visual features are added to the acoustic parameters in order to improve the accuracy of speech recognition in noisy conditions. The Mel-Frequency Cepstral Coefficients are used on the acoustic side whereas Active Appearance Model features are extracted from the image. The feature fusion approach is employed. The...
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EXAMINING INFLUENCE OF VIDEO FRAMERATE AND AUDIO/VIDEO SYNCHRONIZATION ON AUDIO-VISUAL SPEECH RECOGNITION ACCURACY
PublicationThe problem of video framerate and audio/video synchronization in audio-visual speech recognition is considered. The visual features are added to the acoustic parameters in order to improve the accuracy of speech recognition in noisy conditions. The Mel-Frequency Cepstral Coefficients are used on the acoustic side whereas Active Appearance Model features are extracted from the image. The feature fusion approach is employed. The...
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Testing Watermark Robustness against Application of Audio Restoration Algorithms
PublicationThe purpose of this study was to test to what extent watermarks embedded in distorted audio signals are immune to audio restoration algorithm performing. Several restoration routines such as noise reduction, spectrum expansion, clipping or clicks reduction were applied in the online website system. The online service was extended with some copyright protection mechanisms proposed by the authors. They contain low-level music features...
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An new method of audio-visual correlation analysis
PublicationThis paper presents a new methodology of conducting the audio-visual correlation analysis employing the gaze tracking system. Interaction between two perceptual modalities, seeing and hearing, their interaction and mutual reinforcement in a complex relationship was a subject of many research studies. Earlier stage of the carried out experiments at the Multimedia Systems Department (MSD) showed that there exists a relationship between...
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Audio Content and Crowdsourcing: A Subjective Quality Evaluation of Radio Programs Streamed Online
PublicationRadio broadcasting has been present in our lives for over 100 years. The transmission of speech and music signals accompanies us from an early age. Broadcasts provide the latest information from home and abroad. They also shape musical tastes and allow many artists to share their creativity. Modern distribution involves transmission over a number of terrestrial systems. The most popular are analog FM (Frequency Modulation) and...
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A Study on Audio Signal Processed by "Instant Mastering"
PublicationAn increasing amount of music produced in home- and project-studios results in development and growth of "automatic mastering services". The presented investigation explores changes introduced to audio signal by various online mastering platforms. A music set consisting of 10 songs produced in small facilities was processed by eight on-line automatic mastering services. Additionally, some laboratory-constructed signals were tested....
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Bimodal deep learning model for subjectively enhanced emotion classification in films
PublicationThis research delves into the concept of color grading in film, focusing on how color influences the emotional response of the audience. The study commenced by recalling state-of-the-art works that process audio-video signals and associated emotions by machine learning. Then, assumptions of subjective tests for refining and validating an emotion model for assigning specific emotional labels to selected film excerpts were presented....
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Determining Pronunciation Differences in English Allophones Utilizing Audio Signal Parameterization
PublicationAn 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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Towards Audio Signal Equalization Based on Spectral Characteristics of a Listening Room and Music Content Reproduced
PublicationThis study presents investigations of the influence of the room acoustics on the frequency characteristic of the audio signal playback. First, the concept of a novel spectral equalization method of the room acoustic conditions is introduced. On the basis of the room spectral response, a system for room acoustics compensation based on an equalizer designed is proposed. The system settings depend on music genre recognized automatically....
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Objectivization of phonological evaluation of speech elements by means of audio parametrization
PublicationThis study addresses two issues related to both machine- and subjective-based speech evaluation by investigating five phonological phenomena related to allophone production. Its aim is to use objective parametrization and phonological classification of the recorded allophones. These allophones were selected as specifically difficult for Polish speakers of English: aspiration, final obstruent devoicing, dark lateral /l/, velar nasal...
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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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Intelligent equalizer solution employing music genre and the room characteristics analysis
PublicationThe paper presents an intelligent equalizer solution based on room acoustic conditions and music genre analysis. A series of acoustic characteristic measurements are performed for checking the concept proposed. White noise (reference signal) and audio excerpts belonging to six music genres are utilized as excitation signals in measurements. This results in registration of frequency responses of rooms and reverberation times. Signals...
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In uence of Low-Level Features Extracted from Rhythmic and Harmonic Sections on Music Genre Classi cation
PublicationWe present a comprehensive evaluation of the infuence of 'harmonic' and rhythmic sections contained in an audio file on automatic music genre classi cation. The study is performed using the ISMIS database composed of music files, which are represented by vectors of acoustic parameters describing low-level music features. Non-negative Matrix Factorization serves for blind separation of instrument components. Rhythmic components...
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A concept of Signal Equalization Method Based on Music Genre and the Listener's Room Characteristics
PublicationA research study that investigates the influence of the room acoustics environment on the frequency characteristic of the audio signal playback is presented. First, a novel spectral equalization method of the room acoustic conditions is introduced. On the basis of the frequency response of the room, a system for room acoustics compensation based on eight-band equalizer is proposed. The system settings depend on music genre. In...
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Sparse autoregressive modeling
PublicationIn the paper the comparison of the popular pitch determination (PD) algorithms for thepurpose of elimination of clicks from archive audio signals using sparse autoregressive (SAR)modeling is presented. The SAR signal representation has been widely used in code-excitedlinear prediction (CELP) systems. The appropriate construction of the SAR model is requiredto guarantee model stability. For this reason the signal representation...
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Implementation Of The Innovative Radiolocalization System VCS-MLAT (Voice Communication System Multilateration)
PublicationIn the article the concept of the radiolocalization subsystem of the VHF communication for aviation VCS-MLAT (Voice Communication System – Multilateration) is presented. The distributed localization system can estimate the position of the aircraft using the audio signals from aircraft transmitters in the VHF band (118-136 MHz). This paper shows initial verification of the possibility to use voice airband communication to estimate...
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Digital Transformation of Terrestrial Radio: An Analysis of Simulcasted Broadcasts in FM and DAB+ for a Smart and Successful Switchover
PublicationThe process of digitizing radio is far from over. It is an important interdisciplinary aspect, involving Big Data and AI (Artificial Intelligence) when it comes to classifying and handling content, and an organizational challenge in the Industry 4.0 concept. There exist several methods for delivering audio signals, including terrestrial broadcasting and internet streaming. Among them, the DAB+ (Digital Audio Broadcasting plus)...
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Subjective and Objective Comparative Study of DAB+ Broadcast System
PublicationBroadcasting 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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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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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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Evaluation of Sound Enhancement in Mobile Device Using Virtual Bass Synthesiss Algorithm
PublicationAn experiment conducted to validate possibility of use virtual bass synthesis (VBS) algorithm in a portable computer is presented. The subjective listening tests based on the procedure of pairwise comparison between VBS, based on the so-called missing fundamental phenomenon, and standard bass boost technique are employed. The evaluation was carried out in two types of conditions: in a professional listening room and employing an...
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Comparing traffic intensity estimates employing passive acoustic radar and microwave Doppler radar sensor
PublicationThe purpose of our applied research project is to develop an autonomous road sign with built-in radar devices of our design. In this paper, we show that it is possible to calibrate the acoustic vector sensor so that it can be used to measure traffic volume and count the vehicles involved in the traffic through the analysis of the noise emitted by them. Signals obtained from a Doppler radar are used as a reference source. Although...
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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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MODALITY corpus - SPEAKER 35 - COMMANDS C1
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 21 - SEQUENCE S6
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 21 - COMMANDS C5
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 21 - SEQUENCE S4
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 10 - SEQUENCE S1
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 01 - SEQUENCE S2
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 39 - COMMANDS C1
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 01 - SEQUENCE S3
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 01 - COMMANDS C3
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 21 - SEQUENCE S2
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 33 - SEQUENCE S1
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...
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MODALITY corpus - SPEAKER 01 - COMMANDS C2
Open Research DataThe MODALITY corpus is one of the multimodal database of word recordings in English. It consists of over 30 hours of multimodal recordings. The database contains high-resolution, high-framerate stereoscopic video streams and audio signals obtained from a microphone array and a laptop microphone. The corpus can be employed to develop an AVSR system,...