Peter Balazs | |
---|---|
Born | |
Nationality | Austrian |
Alma mater | University of Vienna |
Awards | Start-Preis |
Scientific career | |
Fields | Mathematics, Signal processing, Acoustics |
Institutions | Acoustics Research Institute, CNRS, Université catholique de Louvain |
Doctoral advisor | Hans Georg Feichtinger |
Peter Balazs (born 11 December 1970 in Tulln an der Donau) is an Austrian mathematician working at the Acoustics Research Institute Vienna of the Austrian Academy of Sciences.
Peter Balazs studied mathematics and physics at the University of Vienna. In 2001, he graduated with honors in mathematics and an MSc thesis on "Polynomials over Groups" ("Polynome über Gruppen"). He successfully defended his PhD thesis and graduated (with distinction) in June 2005. His PhD thesis is titled, "Regular and Irregular Gabor Multiplier with Application to Psychoacoustic Masking".
Peter Balazs has been part of the Acoustics Research Institute since 1999. His PhD thesis was written at NuHaG [1] (Numerical Harmonic Analysis Group), Faculty of Mathematics, University of Vienna. The cooperation formed during his thesis also resulted in him becoming a fellow of the HASSIP (Harmonic Analysis and Statistics for Signal and Image Processing) EU network. He joined the LATP (Laboratoire d'Analyse, Topologie, Probabilités), CMI and LMA, CNRS Marseille from November 2003 to April 2004 and in March, May and June 2006. He also worked with the FYMA, UCL, Louvain-La-Neuve in August 2005.
For the project FLAME (Frames and Linear Operators for Acoustical Modeling and Parameter Estimation) Peter Balazs 2011 was honored with the high reputed Start-Preis. [2] [3] He is director of the Acoustics Research Institute since 2012.
Peter Balazs has published 27 journal and 25 conference papers, [4] a selection of which is presented below (in chronological order):
A Fast Fourier Transform (FFT) is an algorithm that computes the Discrete Fourier Transform (DFT) of a sequence, or its inverse (IDFT). Fourier analysis converts a signal from its original domain to a representation in the frequency domain and vice versa. The DFT is obtained by decomposing a sequence of values into components of different frequencies. This operation is useful in many fields, but computing it directly from the definition is often too slow to be practical. An FFT rapidly computes such transformations by factorizing the DFT matrix into a product of sparse factors. As a result, it manages to reduce the complexity of computing the DFT from , which arises if one simply applies the definition of DFT, to , where n is the data size. The difference in speed can be enormous, especially for long data sets where n may be in the thousands or millions. In the presence of round-off error, many FFT algorithms are much more accurate than evaluating the DFT definition directly or indirectly. There are many different FFT algorithms based on a wide range of published theories, from simple complex-number arithmetic to group theory and number theory.
Christopher "Monty" Montgomery is an American programmer and engineer. He is the original creator of the Ogg Free Software container format and the Vorbis audio codec and others, and the founder of The Xiph.Org Foundation, which promotes public domain multimedia codecs. He uses xiphmont as an online pseudonym.
Babak Hassibi is an Iranian-American electrical engineer, computer scientist, and applied mathematician who is the inaugural Mose and Lillian S. Bohn Professor of Electrical Engineering and Computing and Mathematical Sciences at the California Institute of Technology (Caltech). From 2011 to 2016 he was the Gordon M Binder/Amgen Professor of Electrical Engineering. During 2008-2015 he was the Executive Officer of Electrical Engineering and Associate Director of Information Science and Technology.
Andreas J. Winter is a German mathematician and mathematical physicist at the Universitat Autònoma de Barcelona (UAB) in Spain. He received his Ph.D. in 1999 under Rudolf Ahlswede and Friedrich Götze at the Universität Bielefeld in Germany before moving to the University of Bristol and then to the Centre for Quantum Technologies (CQT) at the National University of Singapore. In 2013 he was appointed ICREA Research Professor at UAB.
Speaker diarisation is the process of partitioning an audio stream containing human speech into homogeneous segments according to the identity of each speaker. It can enhance the readability of an automatic speech transcription by structuring the audio stream into speaker turns and, when used together with speaker recognition systems, by providing the speaker’s true identity. It is used to answer the question "who spoke when?" Speaker diarisation is a combination of speaker segmentation and speaker clustering. The first aims at finding speaker change points in an audio stream. The second aims at grouping together speech segments on the basis of speaker characteristics.
Hans Georg Feichtinger is an Austrian mathematician. He is Professor in the mathematical faculty of the University of Vienna. He is editor-in-chief of the Journal of Fourier Analysis and Applications (JFAA) and associate editor to several other journals. He is one of the founders and head of the Numerical Harmonic Analysis Group (NuHAG) at University of Vienna. Today Feichtinger's main field of research is harmonic analysis with a focus on time-frequency analysis.
Yasuo Matsuyama is a Japanese researcher in machine learning and human-aware information processing.
Peter (Petre) Stoica is a researcher and educator in the field of signal processing and its applications to radar/sonar, communications and bio-medicine. He is a professor of Signals and Systems Modeling at Uppsala University in Sweden, and a Member of the Royal Swedish Academy of Engineering Sciences, the United States National Academy of Engineering (International Member), the Romanian Academy, the European Academy of Sciences, and the Royal Society of Sciences. He is also a Fellow of IEEE, EURASIP, IETI, and the Royal Statistical Society.
An audio coding format is a content representation format for storage or transmission of digital audio. Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.
Peter John Olver is a British-American mathematician working in differential geometry.
In Western music, the term chroma feature or chromagram closely relates to twelve different pitch classes. Chroma-based features, which are also referred to as "pitch class profiles", are a powerful tool for analyzing music whose pitches can be meaningfully categorized and whose tuning approximates to the equal-tempered scale. One main property of chroma features is that they capture harmonic and melodic characteristics of music, while being robust to changes in timbre and instrumentation.
René Vidal is a Chilean electrical engineer and computer scientist who is known for his research in machine learning, computer vision, medical image computing, robotics, and control theory. He is the Herschel L. Seder Professor of the Johns Hopkins Department of Biomedical Engineering, and the founding director of the Mathematical Institute for Data Science (MINDS).
Sergio Barbarossa is an Italian professor, engineer and inventor. He is a professor at Sapienza University of Rome, Italy.
An event camera, also known as a neuromorphic camera, silicon retina or dynamic vision sensor, is an imaging sensor that responds to local changes in brightness. Event cameras do not capture images using a shutter as conventional (frame) cameras do. Instead, each pixel inside an event camera operates independently and asynchronously, reporting changes in brightness as they occur, and staying silent otherwise.
Video super-resolution (VSR) is the process of generating high-resolution video frames from the given low-resolution video frames. Unlike single-image super-resolution (SISR), the main goal is not only to restore more fine details while saving coarse ones, but also to preserve motion consistency.
Mark McMahon Wilde is an American quantum information scientist. He is an Associate Professor in the School of Electrical and Computer Engineering at Cornell University, and he is also a Fields Member in the School of Applied and Engineering Physics and the Department of Computer Science at Cornell.
Daniel Kressner is a German numerical analyst. He has a Chair of Numerical Algorithms and High Performance Computing in the Institute of Mathematics at EPF Lausanne.
Mário A. T. Figueiredo is a Portuguese engineer, academic, and researcher. He is an IST Distinguished Professor and holds the Feedzai chair of machine learning at IST, University of Lisbon.
John Cronan Kieffer is an American mathematician best known for his work in information theory, ergodic theory, and stationary process theory.
Audio inpainting is an audio restoration task which deals with the reconstruction of missing or corrupted portions of a digital audio signal. Inpainting techniques are employed when parts of the audio have been lost due to various factors such as transmission errors, data corruption or errors during recording.