Search results for: AUDIO SIGNAL RESTORATION - Bridge of Knowledge

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Search results for: AUDIO SIGNAL RESTORATION

Search results for: AUDIO SIGNAL RESTORATION

  • Intelligent algorithms for optical track audio restoration

    W referacie przedstawiono dwa algorytmy dedykowane redukcji pasożytniczych zniekształceń dźwięku spotykanych w optycznych ścieżkach dźwiękowych. Pierwszy algorytm umożliwia redukcję szerokopasmowego szumu w nagraniach fonicznych. Wykorzystano w nim psycho-akustyczny model słuchu oparty o miarę nieprzewidywalność sygnału (ang. Unpredictability Measure). Ocena jakości redukcji szumu została wykonana z wykorzystaniem metod inteligentnych....

  • Applications of neural networks and perceptual masking to audio restoration

    Publication

    Omówiono zastosowania algorytmów uczących się w dziedzinie rekonstruowania nagrań fonicznych. Szczególną uwagę zwrócono na zastosowanie sztucznych sieci neuronowych do usuwania zakłócających impulsów. Ponadto opisano zastosowanie inteligentnego algorytmu decyzyjnego do sterowania maskowaniem perceptualnym w celu redukowania szumu.

  • Testing Watermark Robustness against Application of Audio Restoration Algorithms

    Publication

    The 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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  • A Study on Audio Signal Processed by "Instant Mastering"

    Publication

    - Year 2018

    An 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....

  • Analysis of allophones based on audio signal recordings and parameterization

    The aim of this study is to develop an allophonic description of English plosive consonants based on recordings of 600 specially selected words. Allophonic variations addressed in the study may have two sources: positional and contextual. The former one depends on the syllabic or prosodic position in which a particular phoneme occurs. Contextual allophony is conditioned by the local phonetic environment. Co-articulation overlapping...

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  • Determining Pronunciation Differences in English Allophones Utilizing Audio Signal Parameterization

    Publication

    - Year 2017

    An 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...

  • Classifying type of vehicles on the basis of data extracted from audio signal characteristics

    The aim of this study is to find and optimize a feature vector for an automatic recognition of the type of vehicles, extracted form an audio signal. First, the influence of weather-based conditions of road surface on spectral characteristic of the audio signal recorded from a passing vehicle in close proximity to the road is discussed. Next, parameterization of the recorded audio signal is performed. For that purpose, the MIRtoolbox,...

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  • Intelligent Audio Signal Processing − Do We Still Need Annotated Datasets?

    Publication

    - Year 2022

    In 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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  • AUDIO SIGNAL EQUALIZATION BASED ON IMPULSE RESPONSE OF A LISTENING ROOM AND MUSIC CONTENT REPRODUCED

    A research study presents investigations of the influence of the room acoustics on the frequency characteristic of the audio signal playback. First, a 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....

  • Automatic audio signal mixing system based on one-dimensional Wave-U-Net autoencoders

    Publication

    - Year 2023

    The purpose of this dissertation is to develop an automatic song mixing system that is capable of automatically mixing a song with good quality in any music genre. This work recalls first the audio signal processing methods used in audio mixing, and it describes selected methods for automatic audio mixing. Then, a novel architecture built based on one-dimensional Wave-U-Net autoencoders is proposed for automatic music mixing. Models...

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  • Towards Audio Signal Equalization Based on Spectral Characteristics of a Listening Room and Music Content Reproduced

    Publication

    - Year 2018

    This 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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  • Online Sound Restoration for Digital Library Applications

    Publication

    A system for sound restoration was conceived and engineered having the following features: no special sound restoration software is needed to perform audio restoration by the user, the process of restoration employs automatic reduction of noise, wow and impulse distortions performed in the online mode, no skills in digital signal processing from the user are needed. The principles of the created system and its features as well...

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  • Online sound restoration system for digital library applications

    Audio signal processing algorithms were introduced to the new online non-commercial service for audio restoration intended to enhance the content of digitized audio repositories. Missing or distorted audio samples are predicted using neural networks and a specific implementation of the Jannsen interpolation method based on the autoregressive model (AR) combined with the iterative restoring of missing signal samples. Since the distortion...

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  • Online sound restoration system for digital library applications.

    Audio signal processing algorithms were introduced to the new online non-commercial service for audio restoration intended to enhance the content of digitized audio repositories. Missing or distorted audio samples are predicted using neural networks and a specific implementation of the Jannsen interpolation method based on the autoregressive model (AR) combined with the iterative restoring of missing signal samples. Since the distortion...

  • Further Developments of the Online Sound Restoration System for Digital Library Applications

    Publication

    New signal processing algorithms were introduced to the online service for audio restoration available at the web address: www.youarchive.net. Missing or distorted audio samples are estimated using a specific implementation of the Jannsen interpolation method. The algorithm is based on the autoregressive model (AR) combined with the iterative complementation of signal samples. Since the interpolation algorithm is computationally...

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  • Localization of impulsive disturbances in audio signals using template matching

    In 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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  • Marek Blok dr hab. inż.

    People

    Marek Blok in 1994 graduated from the Faculty of Electronics at Gdansk University of Technology receiving his MSc in telecommunications. In 2003 received Ph.D.  and in 2017 D.Sc. in telecommunications from the Faculty of Electronics, Telecommunications and Informatics of Gdańsk University of Technology. His research interests are focused on application of digital signal processing in telecommunications. He provides lectures, laboratory...

  • Localization of impulsive disturbances in archive audio signals using predictive matched filtering

    Publication

    The 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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  • Michał Lech dr inż.

    People

    Michał Lech was born in Gdynia in 1983. In 2007 he graduated from the faculty of Electronics, Telecommunications and Informatics of Gdansk University of Technology. In June 2013, he received his Ph.D. degree. The subject of the dissertation was: “A Method and Algorithms for Controlling the Sound Mixing Processes by Hand Gestures Recognized Using Computer Vision”. The main focus of the thesis was the bias of audio perception caused...

  • Grzegorz Szwoch dr hab. inż.

    Grzegorz Szwoch was born in 1972 in Gdansk. In 1991-1996 he studied at the Technical University of Gdansk. In 1996 he graduated as a student from the Sound Engineering Department. His thesis was related to physical modeling of musical instruments. Since that time he has been a member of the research staff at the Multimedia Systems Department as a PhD student (1996-2001), Assistant (2001-2004), Assistant professor (2004-2020) and...