Wykaz publikacji wybranego autora

Konrad Kowalczyk, dr hab. inż., prof. AGH

profesor nadzwyczajny

Wydział Informatyki, Elektroniki i Telekomunikacji
WIEiT-ke, Instytut Elektroniki

  • 2023

    [dyscyplina 1] dziedzina nauk inżynieryjno-technicznych / automatyka, elektronika, elektrotechnika i technologie kosmiczne

    [dyscyplina 2] dziedzina nauk inżynieryjno-technicznych / informatyka techniczna i telekomunikacja (50%)

  • 2020

    [dyscyplina 1] dziedzina nauk inżynieryjno-technicznych / automatyka, elektronika i elektrotechnika

    [dyscyplina 2] dziedzina nauk inżynieryjno-technicznych / informatyka techniczna i telekomunikacja (50%)

  • 2018

    [dyscyplina 1] dziedzina nauk inżynieryjno-technicznych / automatyka, elektronika i elektrotechnika

[poprzednia klasyfikacja] obszar nauk technicznych / dziedzina nauk technicznych / elektronika

Identyfikatory Autora Informacje o Autorze w systemach zewnętrznych

ORCID: 0000-0002-7834-6920 orcid iD

ResearcherID: R-2001-2019

Scopus: 18042288600

PBN: 5e709442878c28a0473b923e

OPI Nauka Polska

System Informacyjny AGH (SkOs)

1
  • Adversarial domain adaptation with paired examples for acoustic scene classification on different recording devices
2
  • Audio replay attack detection using high-frequency features
3
  • Causal signal-based DCCRN with overlapped-frame prediction for online speech enhancement
4
  • Combating reverberation in NTF-based speech separation using a sub-source weighted multichannel Wiener filter and linear prediction
5
  • Comparison of convolution types in CNN-based feature extraction for sound source localization
6
  • Convolutional weighted minimum mean square error filter for joint source separation and dereverberation
7
  • Convolutive NTF for ambisonic source separation under reverberant conditions
8
  • Convolutive weighted multichannel Wiener filter front-end for distant automatic speech recognition in reverberant multispeaker scenarios
9
  • Data diversity for improving DNN-based localization of concurrent sound events
10
  • Distributed multiarray noise reduction with online estimation of masks and spatial filters
11
  • Embedded system for acquisition and enhancement of audio signals
12
  • End-to-end neural speaker diarization with an iterative refinement of non-autoregressive attention-based attractors
13
  • Exploiting rays in blind localization of distributed sensor arrays
14
  • Feature overview for joint modeling of sound event detection and localization using a microphone array
15
  • Frame-based maximum a posteriori estimation of second-order statistics for multichannel speech enhancement in presence of noise
16
  • Game audio: from virtual acoustics to natural spoken dialogues
17
  • Incorporation of localization information for sound source separation in spherical harmonic domain
18
  • Joint blind source separation and dereverberation for automatic speech recognition using delayed-subsource MNMF with localization prior
19
  • Maximum a posteriori estimator for convolutive sound source separation with sub-source based NTF model and the localization probabilistic prior on the mixing matrix
20
  • Multichannel fusion and audio-based features for acoustic event classification
21
  • Multichannel Wiener filter with early reflection raking for automatic speech recognition in presence of reverberation
22
  • NTF of spectral and spatial features for tracking and separation of moving sound sources in spherical harmonic domain
23
  • On parameter adaptation in softmax-based cross-entropy loss for improved convergence speed and accuracy in DNN-based speaker recognition
24
  • On the extraction of early reflection signals for automatic speech recognition
25
  • Refining DNN-based mask estimation using CGMM-based EM algorithm for multi-channel noise reduction