[{"date_updated":"2022-01-06T06:51:11Z","intvolume":"        19","title":"Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation","year":"2011","author":[{"first_name":"Alexander","last_name":"Krueger","full_name":"Krueger, Alexander"},{"first_name":"Ernst","last_name":"Warsitz","full_name":"Warsitz, Ernst"},{"full_name":"Haeb-Umbach, Reinhold","first_name":"Reinhold","last_name":"Haeb-Umbach","id":"242"}],"doi":"10.1109/TASL.2010.2047324","main_file_link":[{"open_access":"1","url":"https://groups.uni-paderborn.de/nt/pubs/2011/KrWaHa11.pdf"}],"language":[{"iso":"eng"}],"abstract":[{"text":"In this paper, we present a novel blocking matrix and fixed beamformer design for a generalized sidelobe canceler for speech enhancement in a reverberant enclosure. They are based on a new method for estimating the acoustical transfer function ratios in the presence of stationary noise. The estimation method relies on solving a generalized eigenvalue problem in each frequency bin. An adaptive eigenvector tracking utilizing the power iteration method is employed and shown to achieve a high convergence speed. Simulation results demonstrate that the proposed beamformer leads to better noise and interference reduction and reduced speech distortions compared to other blocking matrix designs from the literature.","lang":"eng"}],"publication":"IEEE Transactions on Audio, Speech, and Language Processing","issue":"1","keyword":["acoustical transfer function ratio","adaptive eigenvector tracking","array signal processing","beamformer design","blocking matrix","eigenvalues and eigenfunctions","eigenvector-based transfer function ratios estimation","generalized sidelobe canceler","interference reduction","iterative methods","power iteration method","reduced speech distortions","reverberant enclosure","reverberation","speech enhancement","stationary noise"],"type":"journal_article","department":[{"_id":"54"}],"date_created":"2019-07-12T05:29:28Z","status":"public","user_id":"44006","volume":19,"page":"206-219","_id":"11850","citation":{"bibtex":"@article{Krueger_Warsitz_Haeb-Umbach_2011, title={Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation}, volume={19}, DOI={<a href=\"https://doi.org/10.1109/TASL.2010.2047324\">10.1109/TASL.2010.2047324</a>}, number={1}, journal={IEEE Transactions on Audio, Speech, and Language Processing}, author={Krueger, Alexander and Warsitz, Ernst and Haeb-Umbach, Reinhold}, year={2011}, pages={206–219} }","ama":"Krueger A, Warsitz E, Haeb-Umbach R. Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation. <i>IEEE Transactions on Audio, Speech, and Language Processing</i>. 2011;19(1):206-219. doi:<a href=\"https://doi.org/10.1109/TASL.2010.2047324\">10.1109/TASL.2010.2047324</a>","mla":"Krueger, Alexander, et al. “Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation.” <i>IEEE Transactions on Audio, Speech, and Language Processing</i>, vol. 19, no. 1, 2011, pp. 206–19, doi:<a href=\"https://doi.org/10.1109/TASL.2010.2047324\">10.1109/TASL.2010.2047324</a>.","chicago":"Krueger, Alexander, Ernst Warsitz, and Reinhold Haeb-Umbach. “Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation.” <i>IEEE Transactions on Audio, Speech, and Language Processing</i> 19, no. 1 (2011): 206–19. <a href=\"https://doi.org/10.1109/TASL.2010.2047324\">https://doi.org/10.1109/TASL.2010.2047324</a>.","short":"A. Krueger, E. Warsitz, R. Haeb-Umbach, IEEE Transactions on Audio, Speech, and Language Processing 19 (2011) 206–219.","ieee":"A. Krueger, E. Warsitz, and R. Haeb-Umbach, “Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation,” <i>IEEE Transactions on Audio, Speech, and Language Processing</i>, vol. 19, no. 1, pp. 206–219, 2011.","apa":"Krueger, A., Warsitz, E., &#38; Haeb-Umbach, R. (2011). Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation. <i>IEEE Transactions on Audio, Speech, and Language Processing</i>, <i>19</i>(1), 206–219. <a href=\"https://doi.org/10.1109/TASL.2010.2047324\">https://doi.org/10.1109/TASL.2010.2047324</a>"},"oa":"1"},{"date_created":"2019-07-12T05:31:06Z","department":[{"_id":"54"}],"oa":"1","type":"conference","keyword":["adaptive interference canceller","adaptive signal processing","array signal processing","beamforming method","eigenvalues and eigenfunctions","generalized eigenvector blocking matrix","generalized sidelobe canceller","interference suppression","matrix algebra","noise suppression","speech enhancement","transfer function estimation","transfer functions"],"citation":{"mla":"Warsitz, Ernst, et al. “Speech Enhancement with a New Generalized Eigenvector Blocking Matrix for Application in a Generalized Sidelobe Canceller.” <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008)</i>, 2008, pp. 73–76, doi:<a href=\"https://doi.org/10.1109/ICASSP.2008.4517549\">10.1109/ICASSP.2008.4517549</a>.","bibtex":"@inproceedings{Warsitz_Krueger_Haeb-Umbach_2008, title={Speech enhancement with a new generalized eigenvector blocking matrix for application in a generalized sidelobe canceller}, DOI={<a href=\"https://doi.org/10.1109/ICASSP.2008.4517549\">10.1109/ICASSP.2008.4517549</a>}, booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008)}, author={Warsitz, Ernst and Krueger, Alexander and Haeb-Umbach, Reinhold}, year={2008}, pages={73–76} }","ama":"Warsitz E, Krueger A, Haeb-Umbach R. Speech enhancement with a new generalized eigenvector blocking matrix for application in a generalized sidelobe canceller. In: <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008)</i>. ; 2008:73-76. doi:<a href=\"https://doi.org/10.1109/ICASSP.2008.4517549\">10.1109/ICASSP.2008.4517549</a>","ieee":"E. Warsitz, A. Krueger, and R. Haeb-Umbach, “Speech enhancement with a new generalized eigenvector blocking matrix for application in a generalized sidelobe canceller,” in <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008)</i>, 2008, pp. 73–76.","apa":"Warsitz, E., Krueger, A., &#38; Haeb-Umbach, R. (2008). Speech enhancement with a new generalized eigenvector blocking matrix for application in a generalized sidelobe canceller. In <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008)</i> (pp. 73–76). <a href=\"https://doi.org/10.1109/ICASSP.2008.4517549\">https://doi.org/10.1109/ICASSP.2008.4517549</a>","short":"E. Warsitz, A. Krueger, R. Haeb-Umbach, in: IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008), 2008, pp. 73–76.","chicago":"Warsitz, Ernst, Alexander Krueger, and Reinhold Haeb-Umbach. “Speech Enhancement with a New Generalized Eigenvector Blocking Matrix for Application in a Generalized Sidelobe Canceller.” In <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008)</i>, 73–76, 2008. <a href=\"https://doi.org/10.1109/ICASSP.2008.4517549\">https://doi.org/10.1109/ICASSP.2008.4517549</a>."},"publication":"IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008)","abstract":[{"lang":"eng","text":"The generalized sidelobe canceller by Griffith and Jim is a robust beamforming method to enhance a desired (speech) signal in the presence of stationary noise. Its performance depends to a high degree on the construction of the blocking matrix which produces noise reference signals for the subsequent adaptive interference canceller. Especially in reverberated environments the beamformer may suffer from signal leakage and reduced noise suppression. In this paper a new blocking matrix is proposed. It is based on a generalized eigenvalue problem whose solution provides an indirect estimation of the transfer functions from the source to the sensors. The quality of the new generalized eigenvector blocking matrix is studied in simulated rooms with different reverberation times and is compared to alternatives proposed in the literature."}],"language":[{"iso":"eng"}],"_id":"11935","page":"73-76","main_file_link":[{"open_access":"1","url":"https://groups.uni-paderborn.de/nt/pubs/2008/WaKrHa08.pdf"}],"user_id":"44006","doi":"10.1109/ICASSP.2008.4517549","author":[{"first_name":"Ernst","last_name":"Warsitz","full_name":"Warsitz, Ernst"},{"full_name":"Krueger, Alexander","last_name":"Krueger","first_name":"Alexander"},{"id":"242","full_name":"Haeb-Umbach, Reinhold","first_name":"Reinhold","last_name":"Haeb-Umbach"}],"year":"2008","title":"Speech enhancement with a new generalized eigenvector blocking matrix for application in a generalized sidelobe canceller","status":"public","date_updated":"2022-01-06T06:51:12Z"},{"status":"public","_id":"11927","page":"1529-1539","volume":15,"user_id":"44006","citation":{"bibtex":"@article{Warsitz_Haeb-Umbach_2007, title={Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition}, volume={15}, DOI={<a href=\"https://doi.org/10.1109/TASL.2007.898454\">10.1109/TASL.2007.898454</a>}, number={5}, journal={IEEE Transactions on Audio, Speech, and Language Processing}, author={Warsitz, Ernst and Haeb-Umbach, Reinhold}, year={2007}, pages={1529–1539} }","ama":"Warsitz E, Haeb-Umbach R. Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition. <i>IEEE Transactions on Audio, Speech, and Language Processing</i>. 2007;15(5):1529-1539. doi:<a href=\"https://doi.org/10.1109/TASL.2007.898454\">10.1109/TASL.2007.898454</a>","mla":"Warsitz, Ernst, and Reinhold Haeb-Umbach. “Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition.” <i>IEEE Transactions on Audio, Speech, and Language Processing</i>, vol. 15, no. 5, 2007, pp. 1529–39, doi:<a href=\"https://doi.org/10.1109/TASL.2007.898454\">10.1109/TASL.2007.898454</a>.","chicago":"Warsitz, Ernst, and Reinhold Haeb-Umbach. “Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition.” <i>IEEE Transactions on Audio, Speech, and Language Processing</i> 15, no. 5 (2007): 1529–39. <a href=\"https://doi.org/10.1109/TASL.2007.898454\">https://doi.org/10.1109/TASL.2007.898454</a>.","short":"E. Warsitz, R. Haeb-Umbach, IEEE Transactions on Audio, Speech, and Language Processing 15 (2007) 1529–1539.","ieee":"E. Warsitz and R. Haeb-Umbach, “Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition,” <i>IEEE Transactions on Audio, Speech, and Language Processing</i>, vol. 15, no. 5, pp. 1529–1539, 2007.","apa":"Warsitz, E., &#38; Haeb-Umbach, R. (2007). Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition. <i>IEEE Transactions on Audio, Speech, and Language Processing</i>, <i>15</i>(5), 1529–1539. <a href=\"https://doi.org/10.1109/TASL.2007.898454\">https://doi.org/10.1109/TASL.2007.898454</a>"},"oa":"1","author":[{"full_name":"Warsitz, Ernst","first_name":"Ernst","last_name":"Warsitz"},{"id":"242","first_name":"Reinhold","last_name":"Haeb-Umbach","full_name":"Haeb-Umbach, Reinhold"}],"year":"2007","title":"Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition","intvolume":"        15","date_updated":"2022-01-06T06:51:12Z","language":[{"iso":"eng"}],"main_file_link":[{"url":"https://groups.uni-paderborn.de/nt/pubs/2007/WaHa07.pdf","open_access":"1"}],"doi":"10.1109/TASL.2007.898454","publication":"IEEE Transactions on Audio, Speech, and Language Processing","issue":"5","abstract":[{"text":"Maximizing the output signal-to-noise ratio (SNR) of a sensor array in the presence of spatially colored noise leads to a generalized eigenvalue problem. While this approach has extensively been employed in narrowband (antenna) array beamforming, it is typically not used for broadband (microphone) array beamforming due to the uncontrolled amount of speech distortion introduced by a narrowband SNR criterion. In this paper, we show how the distortion of the desired signal can be controlled by a single-channel post-filter, resulting in a performance comparable to the generalized minimum variance distortionless response beamformer, where arbitrary transfer functions relate the source and the microphones. Results are given both for directional and diffuse noise. A novel gradient ascent adaptation algorithm is presented, and its good convergence properties are experimentally revealed by comparison with alternatives from the literature. A key feature of the proposed beamformer is that it operates blindly, i.e., it neither requires knowledge about the array geometry nor an explicit estimation of the transfer functions from source to sensors or the direction-of-arrival.","lang":"eng"}],"date_created":"2019-07-12T05:30:57Z","department":[{"_id":"54"}],"type":"journal_article","keyword":["acoustic signal processing","arbitrary transfer function","array signal processing","blind acoustic beamforming","direction-of-arrival","direction-of-arrival estimation","eigenvalues and eigenfunctions","generalized eigenvalue decomposition","gradient ascent adaptation algorithm","microphone arrays","microphones","narrowband array beamforming","sensor array","single-channel post-filter","spatially colored noise","transfer functions"]},{"volume":4,"user_id":"44006","_id":"11930","page":"iv/797-iv/800 Vol. 4","status":"public","oa":"1","citation":{"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>.","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>","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} }","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>","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.","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.","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>."},"doi":"10.1109/ICASSP.2005.1416129","language":[{"iso":"eng"}],"main_file_link":[{"open_access":"1","url":"https://groups.uni-paderborn.de/nt/pubs/2005/WaHa05.pdf"}],"intvolume":"         4","date_updated":"2022-01-06T06:51:12Z","author":[{"first_name":"Ernst","last_name":"Warsitz","full_name":"Warsitz, Ernst"},{"id":"242","full_name":"Haeb-Umbach, Reinhold","first_name":"Reinhold","last_name":"Haeb-Umbach"}],"title":"Acoustic filter-and-sum beamforming by adaptive principal component analysis","year":"2005","department":[{"_id":"54"}],"type":"conference","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"],"date_created":"2019-07-12T05:31:00Z","abstract":[{"lang":"eng","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."}],"publication":"IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)"}]