--- _id: '11850' abstract: - lang: eng 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. author: - first_name: Alexander full_name: Krueger, Alexander last_name: Krueger - first_name: Ernst full_name: Warsitz, Ernst last_name: Warsitz - first_name: Reinhold full_name: Haeb-Umbach, Reinhold id: '242' last_name: Haeb-Umbach citation: ama: Krueger A, Warsitz E, Haeb-Umbach R. Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation. IEEE Transactions on Audio, Speech, and Language Processing. 2011;19(1):206-219. doi:10.1109/TASL.2010.2047324 apa: Krueger, A., Warsitz, E., & Haeb-Umbach, R. (2011). Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation. IEEE Transactions on Audio, Speech, and Language Processing, 19(1), 206–219. https://doi.org/10.1109/TASL.2010.2047324 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={10.1109/TASL.2010.2047324}, 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} }' chicago: 'Krueger, Alexander, Ernst Warsitz, and Reinhold Haeb-Umbach. “Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation.” IEEE Transactions on Audio, Speech, and Language Processing 19, no. 1 (2011): 206–19. https://doi.org/10.1109/TASL.2010.2047324.' ieee: A. Krueger, E. Warsitz, and R. Haeb-Umbach, “Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation,” IEEE Transactions on Audio, Speech, and Language Processing, vol. 19, no. 1, pp. 206–219, 2011. mla: Krueger, Alexander, et al. “Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation.” IEEE Transactions on Audio, Speech, and Language Processing, vol. 19, no. 1, 2011, pp. 206–19, doi:10.1109/TASL.2010.2047324. short: A. Krueger, E. Warsitz, R. Haeb-Umbach, IEEE Transactions on Audio, Speech, and Language Processing 19 (2011) 206–219. date_created: 2019-07-12T05:29:28Z date_updated: 2022-01-06T06:51:11Z department: - _id: '54' doi: 10.1109/TASL.2010.2047324 intvolume: ' 19' 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 language: - iso: eng main_file_link: - open_access: '1' url: https://groups.uni-paderborn.de/nt/pubs/2011/KrWaHa11.pdf oa: '1' page: 206-219 publication: IEEE Transactions on Audio, Speech, and Language Processing status: public title: Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation type: journal_article user_id: '44006' volume: 19 year: '2011' ... --- _id: '11935' 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. author: - first_name: Ernst full_name: Warsitz, Ernst last_name: Warsitz - first_name: Alexander full_name: Krueger, Alexander last_name: Krueger - first_name: Reinhold full_name: Haeb-Umbach, Reinhold id: '242' last_name: Haeb-Umbach citation: 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: IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008). ; 2008:73-76. doi:10.1109/ICASSP.2008.4517549' apa: Warsitz, E., Krueger, A., & Haeb-Umbach, R. (2008). Speech enhancement with a new generalized eigenvector blocking matrix for application in a generalized sidelobe canceller. In IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008) (pp. 73–76). https://doi.org/10.1109/ICASSP.2008.4517549 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={10.1109/ICASSP.2008.4517549}, 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} }' 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 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008), 73–76, 2008. https://doi.org/10.1109/ICASSP.2008.4517549. 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 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008), 2008, pp. 73–76. mla: Warsitz, Ernst, et al. “Speech Enhancement with a New Generalized Eigenvector Blocking Matrix for Application in a Generalized Sidelobe Canceller.” IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008), 2008, pp. 73–76, doi:10.1109/ICASSP.2008.4517549. short: 'E. Warsitz, A. Krueger, R. Haeb-Umbach, in: IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008), 2008, pp. 73–76.' date_created: 2019-07-12T05:31:06Z date_updated: 2022-01-06T06:51:12Z department: - _id: '54' doi: 10.1109/ICASSP.2008.4517549 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 language: - iso: eng main_file_link: - open_access: '1' url: https://groups.uni-paderborn.de/nt/pubs/2008/WaKrHa08.pdf oa: '1' page: 73-76 publication: IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008) status: public title: Speech enhancement with a new generalized eigenvector blocking matrix for application in a generalized sidelobe canceller type: conference user_id: '44006' year: '2008' ... --- _id: '11930' 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. author: - first_name: Ernst full_name: Warsitz, Ernst last_name: Warsitz - first_name: Reinhold full_name: Haeb-Umbach, Reinhold id: '242' last_name: Haeb-Umbach citation: ama: 'Warsitz E, Haeb-Umbach R. Acoustic filter-and-sum beamforming by adaptive principal component analysis. In: IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005). Vol 4. ; 2005:iv/797-iv/800 Vol. 4. doi:10.1109/ICASSP.2005.1416129' apa: Warsitz, E., & Haeb-Umbach, R. (2005). Acoustic filter-and-sum beamforming by adaptive principal component analysis. In IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005) (Vol. 4, p. iv/797-iv/800 Vol. 4). https://doi.org/10.1109/ICASSP.2005.1416129 bibtex: '@inproceedings{Warsitz_Haeb-Umbach_2005, title={Acoustic filter-and-sum beamforming by adaptive principal component analysis}, volume={4}, DOI={10.1109/ICASSP.2005.1416129}, 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} }' chicago: Warsitz, Ernst, and Reinhold Haeb-Umbach. “Acoustic Filter-and-Sum Beamforming by Adaptive Principal Component Analysis.” In IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005), 4:iv/797-iv/800 Vol. 4, 2005. https://doi.org/10.1109/ICASSP.2005.1416129. ieee: E. Warsitz and R. Haeb-Umbach, “Acoustic filter-and-sum beamforming by adaptive principal component analysis,” in IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005), 2005, vol. 4, p. iv/797-iv/800 Vol. 4. mla: Warsitz, Ernst, and Reinhold Haeb-Umbach. “Acoustic Filter-and-Sum Beamforming by Adaptive Principal Component Analysis.” IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005), vol. 4, 2005, p. iv/797-iv/800 Vol. 4, doi:10.1109/ICASSP.2005.1416129. 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.' date_created: 2019-07-12T05:31:00Z date_updated: 2022-01-06T06:51:12Z department: - _id: '54' doi: 10.1109/ICASSP.2005.1416129 intvolume: ' 4' 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 main_file_link: - open_access: '1' url: https://groups.uni-paderborn.de/nt/pubs/2005/WaHa05.pdf oa: '1' page: iv/797-iv/800 Vol. 4 publication: IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005) status: public title: Acoustic filter-and-sum beamforming by adaptive principal component analysis type: conference user_id: '44006' volume: 4 year: '2005' ...