---
_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: '11913'
abstract:
- lang: eng
  text: In this paper we propose to employ directional statistics in a complex vector
    space to approach the problem of blind speech separation in the presence of spatially
    correlated noise. We interpret the values of the short time Fourier transform
    of the microphone signals to be draws from a mixture of complex Watson distributions,
    a probabilistic model which naturally accounts for spatial aliasing. The parameters
    of the density are related to the a priori source probabilities, the power of
    the sources and the transfer function ratios from sources to sensors. Estimation
    formulas are derived for these parameters by employing the Expectation Maximization
    (EM) algorithm. The E-step corresponds to the estimation of the source presence
    probabilities for each time-frequency bin, while the M-step leads to a maximum
    signal-to-noise ratio (MaxSNR) beamformer in the presence of uncertainty about
    the source activity. Experimental results are reported for an implementation in
    a generalized sidelobe canceller (GSC) like spatial beamforming configuration
    for 3 speech sources with significant coherent noise in reverberant environments,
    demonstrating the usefulness of the novel modeling framework.
author:
- first_name: Dang Hai
  full_name: Tran Vu, Dang Hai
  last_name: Tran Vu
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Tran Vu DH, Haeb-Umbach R. Blind speech separation employing directional statistics
    in an Expectation Maximization framework. In: <i>IEEE International Conference
    on Acoustics, Speech and Signal Processing (ICASSP 2010)</i>. ; 2010:241-244.
    doi:<a href="https://doi.org/10.1109/ICASSP.2010.5495994">10.1109/ICASSP.2010.5495994</a>'
  apa: Tran Vu, D. H., &#38; Haeb-Umbach, R. (2010). Blind speech separation employing
    directional statistics in an Expectation Maximization framework. In <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2010)</i> (pp. 241–244).
    <a href="https://doi.org/10.1109/ICASSP.2010.5495994">https://doi.org/10.1109/ICASSP.2010.5495994</a>
  bibtex: '@inproceedings{Tran Vu_Haeb-Umbach_2010, title={Blind speech separation
    employing directional statistics in an Expectation Maximization framework}, DOI={<a
    href="https://doi.org/10.1109/ICASSP.2010.5495994">10.1109/ICASSP.2010.5495994</a>},
    booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
    (ICASSP 2010)}, author={Tran Vu, Dang Hai and Haeb-Umbach, Reinhold}, year={2010},
    pages={241–244} }'
  chicago: Tran Vu, Dang Hai, and Reinhold Haeb-Umbach. “Blind Speech Separation Employing
    Directional Statistics in an Expectation Maximization Framework.” In <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2010)</i>, 241–44,
    2010. <a href="https://doi.org/10.1109/ICASSP.2010.5495994">https://doi.org/10.1109/ICASSP.2010.5495994</a>.
  ieee: D. H. Tran Vu and R. Haeb-Umbach, “Blind speech separation employing directional
    statistics in an Expectation Maximization framework,” in <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2010)</i>, 2010,
    pp. 241–244.
  mla: Tran Vu, Dang Hai, and Reinhold Haeb-Umbach. “Blind Speech Separation Employing
    Directional Statistics in an Expectation Maximization Framework.” <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2010)</i>, 2010,
    pp. 241–44, doi:<a href="https://doi.org/10.1109/ICASSP.2010.5495994">10.1109/ICASSP.2010.5495994</a>.
  short: 'D.H. Tran Vu, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
    Speech and Signal Processing (ICASSP 2010), 2010, pp. 241–244.'
date_created: 2019-07-12T05:30:40Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2010.5495994
keyword:
- array signal processing
- blind source separation
- blind speech separation
- complex vector space
- complex Watson distribution
- directional statistics
- expectation-maximisation algorithm
- expectation maximization algorithm
- Fourier transform
- Fourier transforms
- generalized sidelobe canceller
- interference suppression
- maximum signal-to-noise ratio beamformer
- microphone signal
- probabilistic model
- spatial aliasing
- spatial beamforming configuration
- speech enhancement
- statistical distributions
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2010/DaHa10-2.pdf
oa: '1'
page: 241-244
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
  (ICASSP 2010)
status: public
title: Blind speech separation employing directional statistics in an Expectation
  Maximization framework
type: conference
user_id: '44006'
year: '2010'
...
---
_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: '11927'
abstract:
- lang: eng
  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.
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. 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>
  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>
  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}
    }'
  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>.'
  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.
  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>.
  short: E. Warsitz, R. Haeb-Umbach, IEEE Transactions on Audio, Speech, and Language
    Processing 15 (2007) 1529–1539.
date_created: 2019-07-12T05:30:57Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/TASL.2007.898454
intvolume: '        15'
issue: '5'
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
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/WaHa07.pdf
oa: '1'
page: 1529-1539
publication: IEEE Transactions on Audio, Speech, and Language Processing
status: public
title: Blind Acoustic Beamforming Based on Generalized Eigenvalue Decomposition
type: journal_article
user_id: '44006'
volume: 15
year: '2007'
...