---
_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. <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>
  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>
  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} }'
  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>.'
  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.
  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>.
  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: <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>'
  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>
  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} }'
  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>.
  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.
  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>.
  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: <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>'
  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>
  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} }'
  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.
  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.'
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'
...