Thomas Esch – Veröffentlichungen
Dipl.-Ing. Thomas Esch
Wissenschaftlicher Mitarbeiter
- Raum:
- 314
- Telefon:
- +49 241 80-26961
- Fax:
- +49 241 80-22254
- E-Mail:
- esch@ind.rwth-aachen.de
- An Information Theoretic View on Artificial Bandwidth Extension in Noisy Environments,
in: Proceedings of IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), (Kyoto, Japan, 25.-30.3.2012), Piscataway, NJ, USA, März 2012, S. 4073–4076. - Dual Channel Reduction of Rapidly Varying Harmonic and Random Noise Using a Spot Microphone,
in: Konferenz Elektronische Sprachsignalverarbeitung (ESSV), (Aachen, Germany, 28.-30.9.2011), Sep. 2011, S. 23–30, ISBN: 978-3-94271-037-4. - Model-Based Speech Enhancement Using SNR Dependent MMSE Estimation,
in: Proceedings of IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), (Prague, Czech Republic, 22.-27.5.2011), Piscataway, NJ, USA, Mai 2011, S. 4652–4655, ISBN: 978-1-45770-539-7. - Combined Reduction of Time Varying Harmonic and Stationary Noise Using Frequency Warping,
in: Conference Record of Asilomar Conference on Signals, Systems, and Computers (ACSSC), (Pacific Grove, CA, USA, 7.-10.11.2010), Piscataway, NJ, USA, Nov. 2010, S. 533–537, ISBN: 978-1-42449-720-1. - A Modified Minimum Statistics Algorithm for Reducing Time Varying Harmonic Noise,
in: ITG-Fachtagung Sprachkommunikation, (Bochum, Germany, 6.-8.10.2010), Berlin, Germany: VDE Verlag GmbH, Okt. 2010, ISBN: 978-3-80073-300-2. - Noise Reduction for Wideband Speech Exploiting Spectral Dependencies Based on Conditional Estimation,
in: ITG-Fachtagung Sprachkommunikation, (Bochum, Germany, 6.-8.10.2011), Berlin: VDE Verlag GmbH, Okt. 2010, ISBN: 978-3-80073-300-2. - Model-Based Dereverberation Preserving Binaural Cues,
in: IEEE Transactions on Audio, Speech, and Language Processing, Band 18, Nr. 7, Sep. 2010, S. 1732–1745, ISSN: 1558-7916, DOI: 10.1109/TASL.2010.2052156.
– Special Issue on Processing Reverberant Speech –. - Wideband Noise Suppression Supported by Artificial Bandwidth Extension Techniques,
in: Proceedings of IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), (Dallas, TX, USA), März 2010, S. 4790–4793. - Matlab and the GNU Debugger: A Powerful Duo for Implementing DSP Algorithms?,
in: IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), (Dallas, TX, USA), März 2010.
– Show and Tell Demonstration –. - RTPROC: Rapid Real-Time Prototyping for Audio Signal Processing,
in: Konferenz Elektronische Sprachsignalverarbeitung (ESSV), (Dresden, Germany, 21.-23.9.2009), Band 1, TUDpress Verlag der WIssenschaften, Sep. 2009, S. 158–166, ISBN: 978-3-94129-831-6. - Efficient Musical Noise Suppression for Speech Enhancement Systems,
in: Proceedings of IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), (Taipei, Taiwan), Apr. 2009, S. 4409–4412, ISBN: 978-1-42442-354-5. - Exploiting Temporal Correlation of Speech and Noise Magnitudes Using a Modified Kalman Filter for Speech Enhancement,
in: ITG-Fachtagung Sprachkommunikation, (Aachen, Germany), VDE Verlag GmbH, Okt. 2008, ISBN: 978-3-80073-120-6. - Modified Kalman Filter Exploiting Interframe Correlation of Speech and Noise Magnitudes,
in: Proceedings of International Workshop on Acoustic Echo and Noise Control (IWAENC), (Seattle, Washington, USA), Sep. 2008. - Speech Enhancement Using a Modified Kalman Filter Based on Complex Linear Prediction and Supergaussian Priors,
in: Proceedings of IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP), (Las Vegas, NV, USA), März 2008, S. 4877–4880, ISBN: 978-1-42441-483-3. - Wideband Coding of Speech and Audio Signals using Bandwidth Extension Techniques,
in: Proceedings of International Student Conference on Electrical Engineering (POSTER), (Prague, Czech Republic), Mai 2006.