Demo Data from EDBT 2026 / reviewers in the wild / expert
Dani Cherkassky
dblp:16/10875
· DBLP profile ↗
5ranked-venue papers
5as first author
0since 2021 · last 2020
0000-0001-8745-3843ORCID · corroborated
Domains — the database's venue-derived domains; a paper can count in several
Artificial intelligence and machine learning · 2 · 2 first-authorGraphics, computer vision, multimedia, augmented reality and games · 2 · 2 first-authorApplied, interdisciplinary, general and emerging computing · 1 · 1 first-author
Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.
| Computer networks
1 paper |
Physical-layer communications · 67% Internet of things and sensor networks · 33% | |
| Computer graphics and multimedia
2 papers |
Audio and music processing · 100% |
Topics — the 6 heaviest of 7, each with the papers that count most for it
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Audio and music processing
speech enhancement |
0.4 | 1 | 2020 | Successive Relative Transfer Function Identification Using Blind Oblique Projection · IEEE ACM Trans. Audio Speech Lang. Process. 2020 |
Internet of things and sensor networks › wireless sensor network › wireless multimedia sensor networks
acoustic sensor network |
0.3 | 1 | 2017 | Blind Synchronization in Wireless Acoustic Sensor Networks · IEEE ACM Trans. Audio Speech Lang. Process. 2017 |
Physical-layer communications › synchronization › synchronization parameter estimation
blind synchronization |
0.3 | 1 | 2017 | Blind Synchronization in Wireless Acoustic Sensor Networks · IEEE ACM Trans. Audio Speech Lang. Process. 2017 |
Physical-layer communications › synchronization › synchronization parameter estimation
sampling rate offset estimation |
0.3 | 1 | 2017 | Blind Synchronization in Wireless Acoustic Sensor Networks · IEEE ACM Trans. Audio Speech Lang. Process. 2017 |
Audio and music processing
beamforming |
0.2 | 2 | 2020 | Successive Relative Transfer Function Identification Using Blind Oblique Projection · IEEE ACM Trans. Audio Speech Lang. Process. 2020 Blind Synchronization in Wireless Acoustic Sensor Networks · IEEE ACM Trans. Audio Speech Lang. Process. 2017 |
Audio and music processing › acoustic signal processing › acoustic sensing › acoustic sensor network
wireless acoustic sensor network |
0.1 | 1 | 2017 | Blind Synchronization in Wireless Acoustic Sensor Networks · IEEE ACM Trans. Audio Speech Lang. Process. 2017 |
Methods — techniques the papers use, named apart from their topics
wideband correlation · 0.6recursive band-limited interpolation · 0.6ambiguity function · 0.6oblique projection · 0.4null steering · 0.4
| Year | Publication | Venue | Position |
|---|---|---|---|
| 2020 | Successive Relative Transfer Function Identification Using Blind Oblique ProjectionabstractDistortionless speech extraction in a reverberant environment can be achieved by applying a beamforming algorithm, provided that the relative transfer functions (RTFs) of the sources and the covariance matrix of the noise are known. In this paper, the challenge of RTF identification in a multi-speaker scenario is addressed. We propose a successive RTF identification (SRI) technique, based on the sole assumption that sources do not become simultaneously active. That is, we address the challenge of estimating the RTF of a specific speech source while assuming that the RTFs of all other active sources in the environment were previously estimated in an earlier stage. The RTF of interest is identified by applying the blind oblique projection (BOP)-SRI technique. When a new speech source is identified, the BOP algorithm is applied. BOP results in a null steering toward the RTF of interest, by means of applying an oblique projection to the microphone measurements. We prove that by artificially increasing the rank of the range of the projection matrix, the RTF of interest can be identified. An experimental study is carried out to evaluate the performance of the BOP-SRI algorithm in various signal to noise ratio (SNR) and signal to interference ratio (SIR) conditions and to demonstrate its effectiveness in speech extraction tasks. Dani Cherkassky, Sharon Gannot |
IEEE ACM Trans. Audio Speech Lang. Process. | 1 |
| 2017 | Blind Synchronization in Wireless Acoustic Sensor NetworksabstractThe challenge of blindly resynchronizing the data acquisition processes in a wireless acoustic sensor network (WASN) is addressed in this paper. The sampling rate offset (SRO) is precisely modeled as a time scaling. The applicability of a wideband correlation processor for estimating the SRO, even in a reverberant and multiple source environment, is presented. An explicit expression for the ambiguity function, which in our case involves time scaling of the received signals, is derived by applying truncated band-limited interpolation. We then propose the recursive band-limited interpolation (RBI) algorithm for recursive SRO estimation. A complete resynchronization scheme utilizing the RBI algorithm, in parallel with the SRO compensation module, is presented. The resulting resynchronization method operates in the time domain in a sequential manner and is, thus, capable of tracking a potentially time-varying SRO. We compared the performance of the proposed RBI algorithm to other available methods in a simulation study. The importance of resynchronization in a beamforming application is demonstrated by both a simulation study and experiments with a real WASN. Finally, we present an experimental study evaluating the expected SRO level between typical data acquisition devices. Dani Cherkassky, Sharon Gannot |
IEEE ACM Trans. Audio Speech Lang. Process. | 1 |
| 2016 | New Insights into the Kalman Filter Beamformer: Applications to Speech and RobustnessabstractStatistically optimal spatial processors (also referred to as data-dependent beamformers) are widely-used spatial focusing techniques for desired source extraction. The Kalman filter-based beamformer (KFB) [1] is a recursive Bayesian method for implementing the beamformer. This letter provides new insights into the KFB. Specifically, we adopt the KFB framework to the task of speech extraction. We formalize the KFB with a set of linear constraints and present its equivalence to the linearly constrained minimum power (LCMP) beamformer. We further show that the optimal output power, required for implementing the KFB, is merely controlling the white noise gain (WNG) of the beamformer. We also show, that in static scenarios, the adaptation rule of the KFB reduces to the simpler affine projection algorithm (APA). The analytically derived results are verified and exemplified by a simulation study. Dani Cherkassky, Sharon Gannot |
IEEE Signal Process. Lett. | 1 |
| 2014 | Multichannel Wiener filter performance analysis in presence of mismodelingabstractA randomly positioned microphone array is considered in this work. In many applications, the locations of the array elements are known up to a certain degree of random mismatch. We derive a novel statistical model for performance analysis of the multi-channel Wiener filter (MWF) beamformer under random mismatch in sensors location. We consider the scenario of one desired source and one interfering source arriving from the far-field and impinging on a linear array. A theoretical model for predicting the MWF mean squared error (MSE) for a given variation in sensors location is developed and verified by simulations. It is postulated that the probability density function (p.d.f) of the MSE of the MWF obeys Γ distribution. This claim is verified empirically by simulations. Dani Cherkassky, Sharon Gannot |
ICASSP | 1 |
| 2014 | Precipitation Classification Using Measurements From Commercial Microwave LinksabstractCommercial wireless microwave links have been recently proven to be an effective tool for precipitation monitoring, mainly for accurate rainfall estimation and high-resolution rainfall mapping. This paper focuses on the challenge of precipitation classification from the measurements of received signal level (RSL) in several commercial wireless microwave links, by suggesting a tree of classification based on the physical features that distinguish between different phenomena. Wet periods are first identified, followed by a classification of the wet periods into pure rain or sleet. The classification is based on the kernel Fisher discriminant analysis, followed by a decision-making process. The suggested procedure is tested on real data, and its performance is evaluated. It is shown that the proposed classification is in very good agreement (85%) with that of a special-purpose meteorological device called disdrometer. Dani Cherkassky, Jonatan Ostrometzky, Hagit Messer |
IEEE Trans. Geosci. Remote. Sens. | 1 |