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Gerald Enzner
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2020 – today
- 2024
- [c67]Aleksej Chinaev, Till Spitz, Stefan Thaleiser, Gerald Enzner:
Matrix Study of Feature Compression Types and Instrumental Speech Quality Metrics in Ultra-Light DNN-Based Spectral Speech Enhancement. IWAENC 2024: 11-15 - [c66]Svantje Voit, Gerald Enzner:
Tiny Neural-Network Control of Frequency-Domain Adaptive Filtering for Linear System Identification in Acoustic Echo Cancellation. IWAENC 2024: 75-79 - 2023
- [j25]Aleksej Chinaev
, Niklas Knaepper
Online distributed waveform-synchronization for acoustic sensor networks with dynamic topology. EURASIP J. Audio Speech Music. Process. 2023(1): 55 (2023) - [j24]Gerald Enzner
Hybrid-Frequency-Resolution Adaptive Kalman Filter for Online Identification of Long Acoustic Responses With Low Input-Output Latency. IEEE ACM Trans. Audio Speech Lang. Process. 31: 3550-3563 (2023) - [j23]Stefan Thaleiser
Binaural-Projection Multichannel Wiener Filter for Cue-Preserving Binaural Speech Enhancement. IEEE ACM Trans. Audio Speech Lang. Process. 31: 3730-3745 (2023) - [c65]Aleksej Chinaev, Niklas Knaepper, Gerald Enzner:
Long-Term Synchronization of Wireless Acoustic Sensor Networks with Nonpersistent Acoustic Activity Using Coherence State. ICASSP 2023: 1-5 - [c64]Svantje Voit, Gerald Enzner:
Neural Network Models with Integrated Training and Adaptation For Nonlinear Acoustic System Identification. ICASSP 2023: 1-5 - 2022
- [c63]Stefan Thaleiser, Gerald Enzner:
Binaural Wind-Noise Tracking with Steering Preset. EUSIPCO 2022: 70-74 - [c62]Gerald Enzner, Christoph Urbanietz, Rainer Martin:
Optimized Learning of Spatial-Fourier Representations from Fast HRIR Recordings. EUSIPCO 2022: 304-308 - [c61]Aleksej Chinaev, Gerald Enzner:
Distributed Synchronization For Ad-Hoc Acoustic Sensor Networks Using Closed-Loop Double-Cross-Correlation Processing. IWAENC 2022: 1-5 - [c60]Christian Gruber, Gerald Enzner, Rainer Martin:
A State-Space Recurrent Neural Network Model for Dynamical Loudspeaker System Identification. IWAENC 2022: 1-5 - [c59]Stefan Thaleiser, Aleksej Chinaev, Rainer Martin, Gerald Enzner:
Dual-Compression Neural Network with Optimized Output Weighting For Improved Single-Channel Speech Enhancement. IWAENC 2022: 1-5 - 2021
- [j22]Feiran Yang
New Insights into Convergence Theory of Constrained Frequency-Domain Adaptive Filters. Circuits Syst. Signal Process. 40(4): 2076-2090 (2021) - [j21]Aleksej Chinaev
Double-Cross-Correlation Processing for Blind Sampling-Rate and Time-Offset Estimation. IEEE ACM Trans. Audio Speech Lang. Process. 29: 1881-1896 (2021) - [j20]Feiran Yang
On the Convergence Behavior of Partitioned-Block Frequency-Domain Adaptive Filters. IEEE Trans. Signal Process. 69: 4906-4920 (2021) - [c58]Aleksej Chinaev, Gerald Enzner, Tobias Gburrek, Joerg Schmalenstroeer:
Online Estimation of Sampling Rate Offsets in Wireless Acoustic Sensor Networks with Packet Loss. EUSIPCO 2021: 1110-1114 - [c57]Aleksej Chinaev
Control Architecture of the Double-Cross-Correlation Processor for Sampling-Rate-Offset Estimation in Acoustic Sensor Networks. ICASSP 2021: 801-805 - [c56]Stefan Thaleiser
Cue-Preserving MMSE Filter with Bayesian SNR Marginalization for Binaural Speech Enhancement. ICASSP 2021: 6124-6128 - [i2]Aleksej Chinaev
Control Architecture of the Double-Cross-Correlation Processor for Sampling-Rate-Offset Estimation in Acoustic Sensor Networks. CoRR abs/2105.13743 (2021) - 2020
- [j19]Christoph Urbanietz
Direct Spatial-Fourier Regression of HRIRs from Multi-Elevation Continuous-Azimuth Recordings. IEEE ACM Trans. Audio Speech Lang. Process. 28: 1129-1142 (2020) - [c55]Stefan Thaleiser
A Computationally Light Algorithm for Bayesian Speech Enhancement with SNR Marginalization. ICASSP 2020: 6209-6213 - [c54]Mario Wolf
Implementing Continuous-Azimuth Binaural Sound in Unity 3D. VR Workshops 2020: 384-389 - [c53]Mario Wolf
Experiencing and Navigating Virtual Reality without Sight (The all-seeING ears). VR Workshops 2020: 525-526
2010 – 2019
- 2019
- [j18]Feiran Yang
A Unified Approach to the Statistical Convergence Analysis of Frequency-Domain Adaptive Filters. IEEE Trans. Signal Process. 67(7): 1785-1796 (2019) - [j17]Hendrik Vogt
State-Space Adaptive Nonlinear Self-Interference Cancellation for Full-Duplex Communication. IEEE Trans. Signal Process. 67(11): 2810-2825 (2019) - [c52]Philipp Thüne, Gerald Enzner
Tracking Theory of Adaptive Filters with Input-Output Sampling Rate Offset. EUSIPCO 2019: 1-5 - [c51]Feiran Yang, Gerald Enzner
Transient Analysis of Partitioned-Block Frequency-Domain Adaptive Filters. EUSIPCO 2019: 1-5 - [c50]Aleksej Chinaev
A Double-cross-correlation Processor for Blind Sampling Rate Offset Estimation in Acoustic Sensor Networks. ICASSP 2019: 641-645 - [c49]Christoph Urbanietz
Spatial-fourier Retrieval of Head-related Impulse Responses from Fast Continuous-azimuth Recordings in the Time-domain. ICASSP 2019: 950-954 - 2018
- [j16]Mehdi Zohourian
Binaural Speaker Localization Integrated Into an Adaptive Beamformer for Hearing Aids. IEEE ACM Trans. Audio Speech Lang. Process. 26(3): 515-528 (2018) - [j15]Gerald Enzner
Bayesian MMSE Filtering of Noisy Speech by SNR Marginalization With Global PSD Priors. IEEE ACM Trans. Audio Speech Lang. Process. 26(12): 2289-2304 (2018) - [c48]Aleksej Chinaev, Gerald Enzner, Joerg Schmalenstroeer:
Fast and Accurate Audio Resampling for Acoustic Sensor Networks by Polyphase-Farrow Filters with FFT Realization. ITG Symposium on Speech Communication 2018: 1-5 - [c47]Philipp Thuene, Gerald Enzner:
Maximum-Likelihood and Maximum-A-Posteriori Perspectives for Blind Channel Identification on Acoustic Sensor Network Data. ITG Symposium on Speech Communication 2018: 1-5 - [c46]Aleksej Chinaev
Low-Rate Farrow Structure with Discrete-Lowpass and Polynomial Support for Audio Resampling. EUSIPCO 2018: 475-479 - [c45]Christoph Urbanietz
Binaural Rendering of Dynamic Head and Sound Source Orientation Using High-Resolution HRTF and Retarded Time. ICASSP 2018: 566-570 - [c44]Jianfeng Guo, Feiran Yang, Gerald Enzner, Jun Yang:
Tracking Analysis and Improvement of Broadband Kalman Filter Using the Two-Echo-Path Model as a Rapid Tunnel. IWAENC 2018: 341-345 - [i1]Hendrik Vogt, Gerald Enzner, Aydin Sezgin:
State-Space Adaptive Nonlinear Self-Interference Cancellation for Full-Duplex Communication. CoRR abs/1806.01004 (2018) - 2017
- [j14]Masoumeh Azarpour, Gerald Enzner
Binaural noise reduction via cue-preserving MMSE filter and adaptive-blocking-based noise PSD estimation. EURASIP J. Adv. Signal Process. 2017: 49 (2017) - [j13]Feiran Yang, Gerald Enzner
Frequency-Domain Adaptive Kalman Filter With Fast Recovery of Abrupt Echo-Path Changes. IEEE Signal Process. Lett. 24(12): 1778-1782 (2017) - [j12]Feiran Yang, Gerald Enzner
Statistical Convergence Analysis for Optimal Control of DFT-Domain Adaptive Echo Canceler. IEEE ACM Trans. Audio Speech Lang. Process. 25(5): 1095-1106 (2017) - [j11]Philipp Thuene
Maximum-Likelihood Approach With Bayesian Refinement for Multichannel-Wiener Postfiltering. IEEE Trans. Signal Process. 65(13): 3399-3413 (2017) - [c43]Gerald Enzner
Robust MMSE filtering for single-microphone speech enhancement. ICASSP 2017: 4009-4013 - [c42]Masoumeh Azarpour, Jan Siska, Gerald Enzner
Realtime binaural speech enhancement demo on raspberry Pi. ICASSP 2017: 6572-6573 - 2016
- [c41]Philipp Thuene, Gerald Enzner:
Maximum-Likelihood Approach to Adaptive Multichannel-Wiener Postfiltering for Wind-Noise Reduction. ITG Symposium on Speech Communication 2016: 1-5 - [c40]Mehdi Zohourian, Gerald Enzner, Rainer Martin:
On the Use of Beamforming Approaches for Binaural Speaker Localization. ITG Symposium on Speech Communication 2016: 1-5 - [c39]Gerald Enzner
Evaluation of estimated hammerstein models via normalized projection misalignment of linear and nonlinear subsystems. ICASSP 2016: 4234-4238 - [c38]Gerald Enzner
Cue-preserving MMSE filter for binaural speech enhancement. IWAENC 2016: 1-5 - 2015
- [c37]Rainer Martin
Binaural speech enhancement with instantaneous coherence smoothing using the cepstral correlation coefficient. ICASSP 2015: 111-115 - [c36]Philipp Thuene
Multichannel Wiener filtering via multichannel decorrelation. ICASSP 2015: 3611-3615 - 2014
- [j10]Dominic Schmid, Gerald Enzner
Variational Bayesian Inference for Multichannel Dereverberation and Noise Reduction. IEEE ACM Trans. Audio Speech Lang. Process. 22(8): 1320-1335 (2014) - [c35]Gerald Enzner
Uncertainty modeling in acoustic echo control. ACSSC 2014: 1633-1638 - [c34]Fabian Kuech, Edwin Mabande, Gerald Enzner
State-space architecture of the partitioned-block-based acoustic echo controller. ICASSP 2014: 1295-1299 - [c33]Masoumeh Azarpour, Gerald Enzner
Binaural noise PSD estimation for binaural speech enhancement. ICASSP 2014: 7068-7072 - [c32]Gerald Enzner
On the statistics and the detection of multichannel common zeros. IWAENC 2014: 21-25 - [c31]Masoumeh Azarpour, Gerald Enzner
Fast noise PSD estimation based on blind channel identification. IWAENC 2014: 223-227 - 2013
- [j9]Sarmad Malik, Gerald Enzner
A Variational Bayesian Learning Approach for Nonlinear Acoustic Echo Control. IEEE Trans. Signal Process. 61(23): 5853-5867 (2013) - [c30]Gerald Enzner, Dominic Schmid, Reinhold Haeb-Umbach:
On acoustic channel identification in multi-microphone systems via adaptive blind signal enhancement techniques. EUSIPCO 2013: 1-5 - [c29]Gerald Enzner
Advanced system options for binaural rendering of Ambisonic format. ICASSP 2013: 251-255 - [c28]Philipp Thuene
Improved online identification of acoustic MISO systems based on separated input signal components. ICASSP 2013: 413-417 - [c27]Ivo Merks, Gerald Enzner
Sound source localization with binaural hearing aids using adaptive blind channel identification. ICASSP 2013: 438-442 - [c26]Masoumeh Azarpour, Gerald Enzner
Adaptive binaural noise reduction based on matched-filter equalization and post-filtering. ICASSP 2013: 7259-7263 - [c25]Philipp Thüne
Trends in adaptive MISO system identification for multichannel audio reproduction and speech communication. ISPA 2013: 767-772 - 2012
- [j8]Sarmad Malik, Dominic Schmid, Gerald Enzner
A State-Space Cross-Relation Approach to Adaptive Blind SIMO System Identification. IEEE Signal Process. Lett. 19(8): 511-514 (2012) - [j7]Sarmad Malik, Gerald Enzner
State-Space Frequency-Domain Adaptive Filtering for Nonlinear Acoustic Echo Cancellation. IEEE Trans. Speech Audio Process. 20(7): 2065-2079 (2012) - [j6]Dominic Schmid, Gerald Enzner
Cross-Relation-Based Blind SIMO Identifiability in the Presence of Near-Common Zeros and Noise. IEEE Trans. Signal Process. 60(1): 60-72 (2012) - [c24]Gerald Enzner:
From Acoustic Nonlinearity to Adaptive Nonlinear System Identification. ITG Conference on Speech Communication 2012: 1-4 - [c23]Dominic Schmid, Philipp Thuene, Dorothea Kolossa, Gerald Enzner:
Dereverberation Preprocessing and Training Data Adjustments for Robust Speech Recognition in Reverberant Environments. ITG Conference on Speech Communication 2012: 1-4 - [c22]Gerald Enzner, Ivo Merks, Tao Zhang:
Adaptive filter algorithms and misalignment criteria for blind binaural channel identification in hearing-aids. EUSIPCO 2012: 315-319 - [c21]Dominic Schmid, Sarmad Malik, Gerald Enzner
An expectation-maximization algorithm for multichannel adaptive speech dereverberation in the frequency-domain. ICASSP 2012: 17-20 - [c20]Sarmad Malik, Gerald Enzner
Variational Bayesian inference for nonlinear acoustic echo cancellation using adaptive cascade modeling. ICASSP 2012: 37-40 - [c19]Christiane Antweiler, Aulis Telle, Peter Vary, Gerald Enzner
Perfect-sweep NLMS for time-variant acoustic system identification. ICASSP 2012: 517-520 - [c18]Masoumeh Azarpour, Gerald Enzner, Rainer Martin:
Distortionless-Response vs. Matched-Filter-Array Processing for Adaptive Binaural Noise Reduction. IWAENC 2012 - [c17]Dominic Schmid, Sarmad Malik, Gerald Enzner:
A Maximum A Posteriori Approach to Multichannel Speech Dereverberation and Denoising. IWAENC 2012 - [c16]Philipp Thüne, Gerald Enzner:
Assessment of Multichannel Acoustic System Identification Using a Spectral-Importance Weighted Misalignment. IWAENC 2012 - 2011
- [j5]Sarmad Malik, Gerald Enzner
Recursive Bayesian Control of Multichannel Acoustic Echo Cancellation. IEEE Signal Process. Lett. 18(11): 619-622 (2011) - [c15]Sarmad Malik, Gerald Enzner
Fourier expansion of hammerstein models for nonlinear acoustic system identification. ICASSP 2011: 85-88 - [c14]Dominic Schmid, Gerald Enzner
Evaluation of adaptive blind SIMO identification in terms of a normalized filter-projection misalignment. ICASSP 2011: 4140-4143 - [c13]Gerald Enzner
3D reconstruction of HRTF-fields from 1D continuous measurements. WASPAA 2011: 157-160 - 2010
- [j4]Matthias Pawig, Gerald Enzner
Adaptive sampling rate correction for acoustic echo control in voice-over-IP. IEEE Trans. Signal Process. 58(1): 189-199 (2010) - [c12]Sarmad Malik, Jan Fligge, Gerald Enzner:
Continuous HRTF Acquisition vs. HRTF Interpolation for Binaural Rendering of Dynamical Auditory Virtual Environments. Sprachkommunikation 2010: 1-4 - [c11]Dominic Schmid, Philipp Thüne, Gerald Enzner:
A Real-Time Speech Dereverberation Environment Based on Multichannel Parametric Room Equalization. Sprachkommunikation 2010: 1-4 - [c10]Sarmad Malik, Gerald Enzner:
Iterative learning of DFT-domain dynamical models subject to parameter variations. EUSIPCO 2010: 845-849 - [c9]Gerald Enzner:
Bayesian inference model for applications of time-varying acoustic system identification. EUSIPCO 2010: 2126-2130 - [c8]Sarmad Malik, Gerald Enzner
Online maximum-likelihood learning of time-varying dynamical models in block-frequency-domain. ICASSP 2010: 3822-3825
2000 – 2009
- 2009
- [c7]Christiane Antweiler, Gerald Enzner
Perfect sequence lms for rapid acquisition of continuous-azimuth head related impulse responses. WASPAA 2009: 281-284 - [c6]Gerald Enzner
3D-continuous-azimuth acquisition of head-related impulse responses using multi-channel adaptive filtering. WASPAA 2009: 325-328 - 2008
- [c5]Gerald Enzner:
Analysis and optimal control of LMS-type adaptive filtering for continuous-azimuth acquisition of head related impulse responses. ICASSP 2008: 393-396 - 2006
- [j3]Gerald Enzner
Frequency-domain adaptive Kalman filter for acoustic echo control in hands-free telephones. Signal Process. 86(6): 1140-1156 (2006) - 2005
- [c4]Gerald Enzner, Peter Vary:
New insights into the statistical signal model and the performance bounds of acoustic echo control. EUSIPCO 2005: 1-4 - 2003
- [c3]Gerald Enzner
A soft-partitioned frequency-domain adaptive filter for acoustic echo cancellation. ICASSP (5) 2003: 393-396 - 2002
- [j2]Gerald Enzner
Partitioned residual echo power estimation for frequency-domain acoustic echo cancellation and postfiltering. Eur. Trans. Telecommun. 13(2): 103-114 (2002) - [j1]Lutz H.-J. Lampe, Robert Schober, Gerald Enzner
Coded continuous phase modulation with low-complexity noncoherent reception. IEEE Trans. Commun. 50(4): 517-520 (2002) - [c2]Gerald Enzner
Unbiased residual echo power estimation for hands-free telephony. ICASSP 2002: 1893-1896 - 2001
- [c1]Lutz H.-J. Lampe, Robert Schober, Gerald Enzner
Noncoherent coded continuous phase modulation. ICC 2001: 2216-2220
Coauthor Index
Philipp Thuene
aka: Philipp Thüne
aka: Philipp Thüne
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