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Wyniki wyszukiwania dla: audio processing objects
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Wow detection and compensation employing spectral processing of audio.
PublikacjaPraca zawiera opis opracowanych algorytmów detekcji i kompensacji pasożytniczych modulacji częstotliwości wynikających z nierównomiernego przesuwu nośnika dźwięku. Proponowane metody opracowano ze szczególnym uwzględnieniem przypadkowych zniekształceń drżenia obecnych w archiwalnych filmowych ścieżkach dźwiękowych. Dodatkowo algorytmy badają wpływ zniekształceń na strukturę formantową sygnałów. Analiza zmian położenia formantów...
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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING
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IEEE Transactions on Audio Speech and Language Processing
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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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RENOVATION OF ARCHIVE AUDIO RECORDINGS USING SPARSE AUTOREGRESSIVE MODELING AND BIDIRECTIONAL PROCESSING
PublikacjaThe paper presents a new approach to elimination of broadband noise and impulsive disturbances from archive audio recordings. The proposed adaptive Kalman-like algorithm, based on a sparse autoregressive model of the audio signal, simultaneously detects noise pulses, interpolates the irrevocably distorted samples and performs signal smoothing. It is shown that bidirectional (forward-backward) processing of the archive signal improves...
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Intelligent Audio Signal Processing − Do We Still Need Annotated Datasets?
PublikacjaIn this paper, intelligent audio signal processing examples are shortly described. The focus is, however, on the machine learning approach and datasets needed, especially for deep learning models. Years of intense research produced many important results in this area; however, the goal of fully intelligent signal processing, characterized by its autonomous acting, is not yet achieved. Therefore, a review of state-of-the-art concerning...
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IEEE-ACM Transactions on Audio Speech and Language Processing
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Adaptive system for recognition of sounds indicating threats to security of people and property employing parallel processing of audio data streams
PublikacjaA system for recognition of threatening acoustic events employing parallel processing on a supercomputing cluster is featured. The methods for detection, parameterization and classication of acoustic events are introduced. The recognition engine is based onthreshold-based detection with adaptive threshold and Support Vector Machine classifcation. Spectral, temporal and mel-frequency descriptors are used as signal features. The...
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EURASIP Journal on Audio Speech and Music Processing
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International Symposium on Audio, Video, Image Processing and Intelligent Applications
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Elimination of Impulsive Disturbances From Stereo Audio Recordings Using Vector Autoregressive Modeling and Variable-order Kalman Filtering
PublikacjaThis 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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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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Dynamic Bayesian Networks for Symbolic Polyphonic Pitch Modeling
PublikacjaSymbolic pitch modeling is a way of incorporating knowledge about relations between pitches into the process of an- alyzing musical information or signals. In this paper, we propose a family of probabilistic symbolic polyphonic pitch models, which account for both the “horizontal” and the “vertical” pitch struc- ture. These models are formulated as linear or log-linear interpo- lations of up to fi ve sub-models, each of which is...
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Estimation of the short-term predictor parameters of speech under noisy conditions
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Genre-Based Music Language Modeling with Latent Hierarchical Pitman-Yor Process Allocation
PublikacjaIn this work we present a new Bayesian topic model: latent hierarchical Pitman-Yor process allocation (LHPYA), which uses hierarchical Pitman-Yor pr ocess priors for both word and topic distributions, and generalizes a few of the existing topic models, including the latent Dirichlet allocation (LDA), the bi- gram topic model and the hierarchical Pitman-Yor topic model. Using such priors allows for integration of -grams with a topic model,...
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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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Piotr Szczuko dr hab. inż.
OsobyDr hab. inż. Piotr Szczuko w 2002 roku ukończył studia na Wydziale Elektroniki, Telekomunikacji i Informatyki Politechniki Gdańskiej zdobywając tytuł magistra inżyniera. Tematem pracy dyplomowej było badanie zjawisk jednoczesnej percepcji obrazu cyfrowego i dźwięku dookólnego. W roku 2008 obronił rozprawę doktorską zatytułowaną "Zastosowanie reguł rozmytych w komputerowej animacji postaci", za którą otrzymał nagrodę Prezesa Rady...
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Personal adaptive tuning of mobile computer audio
PublikacjaAn integrated methodology for enhancing audio quality in mobile computers is presented. The key features are adaptation of the characteristics of the acoustic track to the changing conditions and to the user's individual preferences. Original signal processing algorithms are introduced, which concern: linearization of frequency response, dialogue intelligibility enhancement and dynamics processing tuned up to the user's preferences....
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Marek Blok dr hab. inż.
OsobyMarek Blok w 1994 roku ukończył studia na kierunku Telekomunikacja wydziału Elektroniki Politechniki Gdańskiej i uzyskał tytuł mgra inżyniera. Doktorat w zakresie telekomunikacji uzyskał w 2003 roku na Wydziale Elektroniki, Telekomunikacji i Informatyki Politechniki Gdańskiej. W 2017 roku uzyskał stopień naukowy dra habilitowanego w dyscyplinie telekomunikacja. Jego zainteresowania badawcze ukierunkowane są na telekomunikacyjne...
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Adaptive Personal Tuning of Sound in Mobile Computers
PublikacjaAn integrated methodology for enhancing audio quality in mobile computers is presented. The key features are adaptation of the characteristics of their acoustic track to changing acoustic conditions of the environment and to users’ individual preferences. Signal processing algorithms are introduced that concern: linearization of frequency response, dialogue intelligibility enhancement, and dynamics processing tuned up to the users’...