[{"type":"conference","publication":"IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)","abstract":[{"text":"For human-machine interfaces in distant-talking environments multichannel signal processing is often employed to obtain an enhanced signal for subsequent processing. In this paper we propose a novel adaptation algorithm for a filter-and-sum beamformer to adjust the coefficients of FIR filters to changing acoustic room impulses, e.g. due to speaker movement. A deterministic and a stochastic gradient ascent algorithm are derived from a constrained optimization problem, which iteratively estimates the eigenvector corresponding to the largest eigenvalue of the cross power spectral density of the microphone signals. The method does not require an explicit estimation of the speaker location. The experimental results show fast adaptation and excellent robustness of the proposed algorithm.","lang":"eng"}],"status":"public","_id":"11930","user_id":"44006","department":[{"_id":"54"}],"keyword":["acoustic filter-and-sum beamforming","acoustic room impulses","acoustic signal processing","adaptive principal component analysis","adaptive signal processing","architectural acoustics","constrained optimization problem","cross power spectral density","deterministic algorithm","deterministic algorithms","distant-talking environments","eigenvalues and eigenfunctions","eigenvector","enhanced signal","filter-and-sum beamformer","FIR filter coefficients","FIR filter coefficients","FIR filters","gradient methods","human-machine interfaces","iterative estimation","iterative methods","largest eigenvalue","microphone signals","multichannel signal processing","optimisation","principal component analysis","spectral analysis","stochastic gradient ascent algorithm","stochastic processes"],"language":[{"iso":"eng"}],"year":"2005","citation":{"bibtex":"@inproceedings{Warsitz_Haeb-Umbach_2005, title={Acoustic filter-and-sum beamforming by adaptive principal component analysis}, volume={4}, DOI={<a href=\"https://doi.org/10.1109/ICASSP.2005.1416129\">10.1109/ICASSP.2005.1416129</a>}, booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)}, author={Warsitz, Ernst and Haeb-Umbach, Reinhold}, year={2005}, pages={iv/797-iv/800 Vol. 4} }","mla":"Warsitz, Ernst, and Reinhold Haeb-Umbach. “Acoustic Filter-and-Sum Beamforming by Adaptive Principal Component Analysis.” <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)</i>, vol. 4, 2005, p. iv/797-iv/800 Vol. 4, doi:<a href=\"https://doi.org/10.1109/ICASSP.2005.1416129\">10.1109/ICASSP.2005.1416129</a>.","short":"E. Warsitz, R. Haeb-Umbach, in: IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005), 2005, p. iv/797-iv/800 Vol. 4.","apa":"Warsitz, E., &#38; Haeb-Umbach, R. (2005). Acoustic filter-and-sum beamforming by adaptive principal component analysis. In <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)</i> (Vol. 4, p. iv/797-iv/800 Vol. 4). <a href=\"https://doi.org/10.1109/ICASSP.2005.1416129\">https://doi.org/10.1109/ICASSP.2005.1416129</a>","ama":"Warsitz E, Haeb-Umbach R. Acoustic filter-and-sum beamforming by adaptive principal component analysis. In: <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)</i>. Vol 4. ; 2005:iv/797-iv/800 Vol. 4. doi:<a href=\"https://doi.org/10.1109/ICASSP.2005.1416129\">10.1109/ICASSP.2005.1416129</a>","chicago":"Warsitz, Ernst, and Reinhold Haeb-Umbach. “Acoustic Filter-and-Sum Beamforming by Adaptive Principal Component Analysis.” In <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)</i>, 4:iv/797-iv/800 Vol. 4, 2005. <a href=\"https://doi.org/10.1109/ICASSP.2005.1416129\">https://doi.org/10.1109/ICASSP.2005.1416129</a>.","ieee":"E. Warsitz and R. Haeb-Umbach, “Acoustic filter-and-sum beamforming by adaptive principal component analysis,” in <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)</i>, 2005, vol. 4, p. iv/797-iv/800 Vol. 4."},"page":"iv/797-iv/800 Vol. 4","intvolume":"         4","oa":"1","date_updated":"2022-01-06T06:51:12Z","author":[{"full_name":"Warsitz, Ernst","last_name":"Warsitz","first_name":"Ernst"},{"last_name":"Haeb-Umbach","id":"242","full_name":"Haeb-Umbach, Reinhold","first_name":"Reinhold"}],"date_created":"2019-07-12T05:31:00Z","volume":4,"title":"Acoustic filter-and-sum beamforming by adaptive principal component analysis","main_file_link":[{"open_access":"1","url":"https://groups.uni-paderborn.de/nt/pubs/2005/WaHa05.pdf"}],"doi":"10.1109/ICASSP.2005.1416129"},{"language":[{"iso":"eng"}],"keyword":["acoustic echo cancellation algorithms","adverse environmental conditions","automatic speech recognition","cepstral analysis","cepstral features","cepstral mean normalization","command word task","delta-delta features","delta features","echo suppression","error rate reductions","feature vector components","FIR filters","LDA derived cepstral trajectory filters","linear discriminant analysis","long-range feature filters","phone accuracy","real-life room impulse responses","reverberant data","spectral parameters","speech recognition","standard TIMIT phone recognition task"],"department":[{"_id":"54"}],"user_id":"44006","_id":"11869","status":"public","abstract":[{"lang":"eng","text":"Amongst several data driven approaches for designing filters for the time sequence of spectral parameters, the linear discriminant analysis (LDA) based method has been proposed for automatic speech recognition. Here we apply LDA-based filter design to cepstral features, which better match the inherent assumption of this method that feature vector components are uncorrelated. Extensive recognition experiments have been conducted both on the standard TIMIT phone recognition task and on a proprietary 130-words command word task under various adverse environmental conditions, including reverberant data with real-life room impulse responses and data processed by acoustic echo cancellation algorithms. Significant error rate reductions have been achieved when applying the novel long-range feature filters compared to standard approaches employing cepstral mean normalization and delta and delta-delta features, in particular when facing acoustic echo cancellation scenarios and room reverberation. For example, the phone accuracy on reverberated TIMIT data could be increased from 50.7\\% to 56.0\\%"}],"publication":"IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP 2000)","type":"conference","doi":"10.1109/ICASSP.2000.859157","main_file_link":[{"open_access":"1","url":"https://groups.uni-paderborn.de/nt/pubs/2000/LiHa00.pdf"}],"title":"LDA derived cepstral trajectory filters in adverse environmental conditions","volume":2,"author":[{"first_name":"M.","last_name":"Lieb","full_name":"Lieb, M."},{"first_name":"Reinhold","id":"242","full_name":"Haeb-Umbach, Reinhold","last_name":"Haeb-Umbach"}],"date_created":"2019-07-12T05:29:50Z","date_updated":"2022-01-06T06:51:11Z","oa":"1","page":"II1105-II1108 vol.2","intvolume":"         2","citation":{"apa":"Lieb, M., &#38; Haeb-Umbach, R. (2000). LDA derived cepstral trajectory filters in adverse environmental conditions. In <i>IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP 2000)</i> (Vol. 2, pp. II1105-II1108 vol.2). <a href=\"https://doi.org/10.1109/ICASSP.2000.859157\">https://doi.org/10.1109/ICASSP.2000.859157</a>","mla":"Lieb, M., and Reinhold Haeb-Umbach. “LDA Derived Cepstral Trajectory Filters in Adverse Environmental Conditions.” <i>IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP 2000)</i>, vol. 2, 2000, pp. II1105-II1108 vol.2, doi:<a href=\"https://doi.org/10.1109/ICASSP.2000.859157\">10.1109/ICASSP.2000.859157</a>.","short":"M. Lieb, R. Haeb-Umbach, in: IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP 2000), 2000, pp. II1105-II1108 vol.2.","bibtex":"@inproceedings{Lieb_Haeb-Umbach_2000, title={LDA derived cepstral trajectory filters in adverse environmental conditions}, volume={2}, DOI={<a href=\"https://doi.org/10.1109/ICASSP.2000.859157\">10.1109/ICASSP.2000.859157</a>}, booktitle={IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP 2000)}, author={Lieb, M. and Haeb-Umbach, Reinhold}, year={2000}, pages={II1105-II1108 vol.2} }","ama":"Lieb M, Haeb-Umbach R. LDA derived cepstral trajectory filters in adverse environmental conditions. In: <i>IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP 2000)</i>. Vol 2. ; 2000:II1105-II1108 vol.2. doi:<a href=\"https://doi.org/10.1109/ICASSP.2000.859157\">10.1109/ICASSP.2000.859157</a>","ieee":"M. Lieb and R. Haeb-Umbach, “LDA derived cepstral trajectory filters in adverse environmental conditions,” in <i>IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP 2000)</i>, 2000, vol. 2, pp. II1105-II1108 vol.2.","chicago":"Lieb, M., and Reinhold Haeb-Umbach. “LDA Derived Cepstral Trajectory Filters in Adverse Environmental Conditions.” In <i>IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP 2000)</i>, 2:II1105-II1108 vol.2, 2000. <a href=\"https://doi.org/10.1109/ICASSP.2000.859157\">https://doi.org/10.1109/ICASSP.2000.859157</a>."},"year":"2000"}]