---
_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'
...
---
_id: '11869'
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\%
author:
- first_name: M.
  full_name: Lieb, M.
  last_name: Lieb
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  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>'
  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>
  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} }'
  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>.
  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.
  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.'
date_created: 2019-07-12T05:29:50Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2000.859157
intvolume: '         2'
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
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2000/LiHa00.pdf
oa: '1'
page: II1105-II1108 vol.2
publication: IEEE International Conference on Acoustics, Speech, and Signal Processing
  (ICASSP 2000)
status: public
title: LDA derived cepstral trajectory filters in adverse environmental conditions
type: conference
user_id: '44006'
volume: 2
year: '2000'
...