---
_id: '11934'
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
- first_name: Dang Hai
  full_name: Tran Vu, Dang Hai
  last_name: Tran Vu
citation:
  ama: 'Warsitz E, Haeb-Umbach R, Tran Vu DH. Blind Adaptive Principal Eigenvector
    Beamforming for Acoustical Source Separation. In: <i>Interspeech 2007</i>. ; 2007.'
  apa: Warsitz, E., Haeb-Umbach, R., &#38; Tran Vu, D. H. (2007). Blind Adaptive Principal
    Eigenvector Beamforming for Acoustical Source Separation. In <i>Interspeech 2007</i>.
  bibtex: '@inproceedings{Warsitz_Haeb-Umbach_Tran Vu_2007, title={Blind Adaptive
    Principal Eigenvector Beamforming for Acoustical Source Separation}, booktitle={Interspeech
    2007}, author={Warsitz, Ernst and Haeb-Umbach, Reinhold and Tran Vu, Dang Hai},
    year={2007} }'
  chicago: Warsitz, Ernst, Reinhold Haeb-Umbach, and Dang Hai Tran Vu. “Blind Adaptive
    Principal Eigenvector Beamforming for Acoustical Source Separation.” In <i>Interspeech
    2007</i>, 2007.
  ieee: E. Warsitz, R. Haeb-Umbach, and D. H. Tran Vu, “Blind Adaptive Principal Eigenvector
    Beamforming for Acoustical Source Separation,” in <i>Interspeech 2007</i>, 2007.
  mla: Warsitz, Ernst, et al. “Blind Adaptive Principal Eigenvector Beamforming for
    Acoustical Source Separation.” <i>Interspeech 2007</i>, 2007.
  short: 'E. Warsitz, R. Haeb-Umbach, D.H. Tran Vu, in: Interspeech 2007, 2007.'
date_created: 2019-07-12T05:31:05Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/WaHaDa07.pdf
oa: '1'
publication: Interspeech 2007
status: public
title: Blind Adaptive Principal Eigenvector Beamforming for Acoustical Source Separation
type: conference
user_id: '44006'
year: '2007'
...
---
_id: '11941'
author:
- first_name: Stefan
  full_name: Windmann, Stefan
  last_name: Windmann
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Windmann S, Haeb-Umbach R. An Approach to Iterative Speech Feature Enhancement
    and Recognition. In: <i>Interspeech 2007</i>. ; 2007.'
  apa: Windmann, S., &#38; Haeb-Umbach, R. (2007). An Approach to Iterative Speech
    Feature Enhancement and Recognition. In <i>Interspeech 2007</i>.
  bibtex: '@inproceedings{Windmann_Haeb-Umbach_2007, title={An Approach to Iterative
    Speech Feature Enhancement and Recognition}, booktitle={Interspeech 2007}, author={Windmann,
    Stefan and Haeb-Umbach, Reinhold}, year={2007} }'
  chicago: Windmann, Stefan, and Reinhold Haeb-Umbach. “An Approach to Iterative Speech
    Feature Enhancement and Recognition.” In <i>Interspeech 2007</i>, 2007.
  ieee: S. Windmann and R. Haeb-Umbach, “An Approach to Iterative Speech Feature Enhancement
    and Recognition,” in <i>Interspeech 2007</i>, 2007.
  mla: Windmann, Stefan, and Reinhold Haeb-Umbach. “An Approach to Iterative Speech
    Feature Enhancement and Recognition.” <i>Interspeech 2007</i>, 2007.
  short: 'S. Windmann, R. Haeb-Umbach, in: Interspeech 2007, 2007.'
date_created: 2019-07-12T05:31:13Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/WiHa07.pdf
oa: '1'
publication: Interspeech 2007
status: public
title: An Approach to Iterative Speech Feature Enhancement and Recognition
type: conference
user_id: '44006'
year: '2007'
...
---
_id: '11893'
author:
- first_name: Joerg
  full_name: Schmalenstroeer, Joerg
  id: '460'
  last_name: Schmalenstroeer
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Schmalenstroeer J, Haeb-Umbach R. Joint Speaker Segmentation, Localization
    and Identification for Streaming Audio. In: <i>Interspeech 2007</i>. ; 2007.'
  apa: Schmalenstroeer, J., &#38; Haeb-Umbach, R. (2007). Joint Speaker Segmentation,
    Localization and Identification for Streaming Audio. <i>Interspeech 2007</i>.
  bibtex: '@inproceedings{Schmalenstroeer_Haeb-Umbach_2007, title={Joint Speaker Segmentation,
    Localization and Identification for Streaming Audio}, booktitle={Interspeech 2007},
    author={Schmalenstroeer, Joerg and Haeb-Umbach, Reinhold}, year={2007} }'
  chicago: Schmalenstroeer, Joerg, and Reinhold Haeb-Umbach. “Joint Speaker Segmentation,
    Localization and Identification for Streaming Audio.” In <i>Interspeech 2007</i>,
    2007.
  ieee: J. Schmalenstroeer and R. Haeb-Umbach, “Joint Speaker Segmentation, Localization
    and Identification for Streaming Audio,” 2007.
  mla: Schmalenstroeer, Joerg, and Reinhold Haeb-Umbach. “Joint Speaker Segmentation,
    Localization and Identification for Streaming Audio.” <i>Interspeech 2007</i>,
    2007.
  short: 'J. Schmalenstroeer, R. Haeb-Umbach, in: Interspeech 2007, 2007.'
date_created: 2019-07-12T05:30:17Z
date_updated: 2023-10-26T08:10:02Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/ScHa07.pdf
oa: '1'
publication: Interspeech 2007
quality_controlled: '1'
status: public
title: Joint Speaker Segmentation, Localization and Identification for Streaming Audio
type: conference
user_id: '460'
year: '2007'
...
---
_id: '11901'
author:
- first_name: Joerg
  full_name: Schmalenstroeer, Joerg
  id: '460'
  last_name: Schmalenstroeer
- first_name: Volker
  full_name: Leutnant, Volker
  last_name: Leutnant
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Schmalenstroeer J, Leutnant V, Haeb-Umbach R. Amigo Context Management Service
    with Applications in Ambient Communication Scenarios. In: <i>AMI-07 - European
    Conference on Ambient Intelligence</i>. ; 2007.'
  apa: Schmalenstroeer, J., Leutnant, V., &#38; Haeb-Umbach, R. (2007). Amigo Context
    Management Service with Applications in Ambient Communication Scenarios. <i>AMI-07
    - European Conference on Ambient Intelligence</i>.
  bibtex: '@inproceedings{Schmalenstroeer_Leutnant_Haeb-Umbach_2007, title={Amigo
    Context Management Service with Applications in Ambient Communication Scenarios},
    booktitle={AMI-07 - European Conference on Ambient Intelligence}, author={Schmalenstroeer,
    Joerg and Leutnant, Volker and Haeb-Umbach, Reinhold}, year={2007} }'
  chicago: Schmalenstroeer, Joerg, Volker Leutnant, and Reinhold Haeb-Umbach. “Amigo
    Context Management Service with Applications in Ambient Communication Scenarios.”
    In <i>AMI-07 - European Conference on Ambient Intelligence</i>, 2007.
  ieee: J. Schmalenstroeer, V. Leutnant, and R. Haeb-Umbach, “Amigo Context Management
    Service with Applications in Ambient Communication Scenarios,” 2007.
  mla: Schmalenstroeer, Joerg, et al. “Amigo Context Management Service with Applications
    in Ambient Communication Scenarios.” <i>AMI-07 - European Conference on Ambient
    Intelligence</i>, 2007.
  short: 'J. Schmalenstroeer, V. Leutnant, R. Haeb-Umbach, in: AMI-07 - European Conference
    on Ambient Intelligence, 2007.'
date_created: 2019-07-12T05:30:27Z
date_updated: 2023-10-26T08:13:09Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/ScLeHa07.pdf
oa: '1'
publication: AMI-07 - European Conference on Ambient Intelligence
quality_controlled: '1'
status: public
title: Amigo Context Management Service with Applications in Ambient Communication
  Scenarios
type: conference
user_id: '460'
year: '2007'
...
---
_id: '11933'
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
- first_name: Joerg
  full_name: Schmalenstroeer, Joerg
  id: '460'
  last_name: Schmalenstroeer
citation:
  ama: 'Warsitz E, Haeb-Umbach R, Schmalenstroeer J. Zweistufige Sprache/Pause-Detektion
    in stark gestoerter Umgebung. In: <i>33. Deutsche Jahrestagung Fuer Akustik (DAGA
    2007)</i>. ; 2007.'
  apa: Warsitz, E., Haeb-Umbach, R., &#38; Schmalenstroeer, J. (2007). Zweistufige
    Sprache/Pause-Detektion in stark gestoerter Umgebung. <i>33. Deutsche Jahrestagung
    Fuer Akustik (DAGA 2007)</i>.
  bibtex: '@inproceedings{Warsitz_Haeb-Umbach_Schmalenstroeer_2007, title={Zweistufige
    Sprache/Pause-Detektion in stark gestoerter Umgebung}, booktitle={33. Deutsche
    Jahrestagung fuer Akustik (DAGA 2007)}, author={Warsitz, Ernst and Haeb-Umbach,
    Reinhold and Schmalenstroeer, Joerg}, year={2007} }'
  chicago: Warsitz, Ernst, Reinhold Haeb-Umbach, and Joerg Schmalenstroeer. “Zweistufige
    Sprache/Pause-Detektion in Stark Gestoerter Umgebung.” In <i>33. Deutsche Jahrestagung
    Fuer Akustik (DAGA 2007)</i>, 2007.
  ieee: E. Warsitz, R. Haeb-Umbach, and J. Schmalenstroeer, “Zweistufige Sprache/Pause-Detektion
    in stark gestoerter Umgebung,” 2007.
  mla: Warsitz, Ernst, et al. “Zweistufige Sprache/Pause-Detektion in Stark Gestoerter
    Umgebung.” <i>33. Deutsche Jahrestagung Fuer Akustik (DAGA 2007)</i>, 2007.
  short: 'E. Warsitz, R. Haeb-Umbach, J. Schmalenstroeer, in: 33. Deutsche Jahrestagung
    Fuer Akustik (DAGA 2007), 2007.'
date_created: 2019-07-12T05:31:04Z
date_updated: 2023-10-26T08:12:49Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/WaHaSc07.pdf
oa: '1'
publication: 33. Deutsche Jahrestagung fuer Akustik (DAGA 2007)
quality_controlled: '1'
status: public
title: Zweistufige Sprache/Pause-Detektion in stark gestoerter Umgebung
type: conference
user_id: '460'
year: '2007'
...
---
_id: '11902'
author:
- first_name: Joerg
  full_name: Schmalenstroeer, Joerg
  id: '460'
  last_name: Schmalenstroeer
- 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: 'Schmalenstroeer J, Warsitz E, Haeb-Umbach R. Projekt Amigo - Sprachsignalverarbeitung
    im vernetzten Haus. In: <i>33. Deutsche Jahrestagung Fuer Akustik (DAGA 2007)</i>.
    ; 2007.'
  apa: Schmalenstroeer, J., Warsitz, E., &#38; Haeb-Umbach, R. (2007). Projekt Amigo
    - Sprachsignalverarbeitung im vernetzten Haus. <i>33. Deutsche Jahrestagung Fuer
    Akustik (DAGA 2007)</i>.
  bibtex: '@inproceedings{Schmalenstroeer_Warsitz_Haeb-Umbach_2007, title={Projekt
    Amigo - Sprachsignalverarbeitung im vernetzten Haus}, booktitle={33. Deutsche
    Jahrestagung fuer Akustik (DAGA 2007)}, author={Schmalenstroeer, Joerg and Warsitz,
    Ernst and Haeb-Umbach, Reinhold}, year={2007} }'
  chicago: Schmalenstroeer, Joerg, Ernst Warsitz, and Reinhold Haeb-Umbach. “Projekt
    Amigo - Sprachsignalverarbeitung Im Vernetzten Haus.” In <i>33. Deutsche Jahrestagung
    Fuer Akustik (DAGA 2007)</i>, 2007.
  ieee: J. Schmalenstroeer, E. Warsitz, and R. Haeb-Umbach, “Projekt Amigo - Sprachsignalverarbeitung
    im vernetzten Haus,” 2007.
  mla: Schmalenstroeer, Joerg, et al. “Projekt Amigo - Sprachsignalverarbeitung Im
    Vernetzten Haus.” <i>33. Deutsche Jahrestagung Fuer Akustik (DAGA 2007)</i>, 2007.
  short: 'J. Schmalenstroeer, E. Warsitz, R. Haeb-Umbach, in: 33. Deutsche Jahrestagung
    Fuer Akustik (DAGA 2007), 2007.'
date_created: 2019-07-12T05:30:28Z
date_updated: 2023-10-26T08:13:01Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/ScWaHa07.pdf
oa: '1'
publication: 33. Deutsche Jahrestagung fuer Akustik (DAGA 2007)
quality_controlled: '1'
status: public
title: Projekt Amigo - Sprachsignalverarbeitung im vernetzten Haus
type: conference
user_id: '460'
year: '2007'
...
---
_id: '11823'
abstract:
- lang: eng
  text: In this study we evaluate transmission error compensation techniques for distributed
    speech recognition systems based on modification of the speech decoder. The candidates
    are marginalization, weighted Viterbi and our recently proposed soft-feature uncertainty
    decoding. For the latter, it is shown how the Bayesian speech recognition approach
    must be reformulated for recognition at the server side. The resulting predictive
    classifier is able to take account of the transmission errors by changing the
    contribution of the affected speech features to the acoustic score. The comparison
    of the experimental results has proven the superiority of our approach.
author:
- first_name: Valentin
  full_name: Ion, Valentin
  last_name: Ion
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Ion V, Haeb-Umbach R. Comparison of Decoder-based Transmission Error Compensation
    Techniques for Distributed Speech Recognition. In: <i>7. ITG-Fachtagung Sprachkommunikation</i>.
    ; 2006.'
  apa: Ion, V., &#38; Haeb-Umbach, R. (2006). Comparison of Decoder-based Transmission
    Error Compensation Techniques for Distributed Speech Recognition. In <i>7. ITG-Fachtagung
    Sprachkommunikation</i>.
  bibtex: '@inproceedings{Ion_Haeb-Umbach_2006, title={Comparison of Decoder-based
    Transmission Error Compensation Techniques for Distributed Speech Recognition},
    booktitle={7. ITG-Fachtagung Sprachkommunikation}, author={Ion, Valentin and Haeb-Umbach,
    Reinhold}, year={2006} }'
  chicago: Ion, Valentin, and Reinhold Haeb-Umbach. “Comparison of Decoder-Based Transmission
    Error Compensation Techniques for Distributed Speech Recognition.” In <i>7. ITG-Fachtagung
    Sprachkommunikation</i>, 2006.
  ieee: V. Ion and R. Haeb-Umbach, “Comparison of Decoder-based Transmission Error
    Compensation Techniques for Distributed Speech Recognition,” in <i>7. ITG-Fachtagung
    Sprachkommunikation</i>, 2006.
  mla: Ion, Valentin, and Reinhold Haeb-Umbach. “Comparison of Decoder-Based Transmission
    Error Compensation Techniques for Distributed Speech Recognition.” <i>7. ITG-Fachtagung
    Sprachkommunikation</i>, 2006.
  short: 'V. Ion, R. Haeb-Umbach, in: 7. ITG-Fachtagung Sprachkommunikation, 2006.'
date_created: 2019-07-12T05:28:56Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/IoHa06-1.pdf
oa: '1'
publication: 7. ITG-Fachtagung Sprachkommunikation
status: public
title: Comparison of Decoder-based Transmission Error Compensation Techniques for
  Distributed Speech Recognition
type: conference
user_id: '44006'
year: '2006'
...
---
_id: '11824'
abstract:
- lang: eng
  text: Soft-feature based speech recognition, which is an example of uncertainty
    decoding, has been proven to be a robust error mitigation method for distributed
    speech recognition over wireless channels exhibiting bit errors. In this paper
    we extend this concept to packet-oriented transmissions. The a posteriori probability
    density function of the lost feature vector, given the closest received neighbours,
    is computed. In the experiments, the nearest frame repetition, which is shown
    to be equivalent to the MAP estimate, outperforms the MMSE estimate for long bursts.
    Taking the variance into account at the speech recognition stage results in superior
    performance compared to classical schemes using point estimates. A computationally
    and memory efficient implementation of the proposed packet loss compensation scheme
    based on table lookup is presented
author:
- first_name: Valentin
  full_name: Ion, Valentin
  last_name: Ion
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Ion V, Haeb-Umbach R. An Inexpensive Packet Loss Compensation Scheme for Distributed
    Speech Recognition Based on Soft-Features. In: <i>IEEE International Conference
    on Acoustics, Speech and Signal Processing (ICASSP 2006)</i>. Vol 1. ; 2006:I.
    doi:<a href="https://doi.org/10.1109/ICASSP.2006.1659984">10.1109/ICASSP.2006.1659984</a>'
  apa: Ion, V., &#38; Haeb-Umbach, R. (2006). An Inexpensive Packet Loss Compensation
    Scheme for Distributed Speech Recognition Based on Soft-Features. In <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2006)</i> (Vol.
    1, p. I). <a href="https://doi.org/10.1109/ICASSP.2006.1659984">https://doi.org/10.1109/ICASSP.2006.1659984</a>
  bibtex: '@inproceedings{Ion_Haeb-Umbach_2006, title={An Inexpensive Packet Loss
    Compensation Scheme for Distributed Speech Recognition Based on Soft-Features},
    volume={1}, DOI={<a href="https://doi.org/10.1109/ICASSP.2006.1659984">10.1109/ICASSP.2006.1659984</a>},
    booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
    (ICASSP 2006)}, author={Ion, Valentin and Haeb-Umbach, Reinhold}, year={2006},
    pages={I} }'
  chicago: Ion, Valentin, and Reinhold Haeb-Umbach. “An Inexpensive Packet Loss Compensation
    Scheme for Distributed Speech Recognition Based on Soft-Features.” In <i>IEEE
    International Conference on Acoustics, Speech and Signal Processing (ICASSP 2006)</i>,
    1:I, 2006. <a href="https://doi.org/10.1109/ICASSP.2006.1659984">https://doi.org/10.1109/ICASSP.2006.1659984</a>.
  ieee: V. Ion and R. Haeb-Umbach, “An Inexpensive Packet Loss Compensation Scheme
    for Distributed Speech Recognition Based on Soft-Features,” in <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2006)</i>, 2006,
    vol. 1, p. I.
  mla: Ion, Valentin, and Reinhold Haeb-Umbach. “An Inexpensive Packet Loss Compensation
    Scheme for Distributed Speech Recognition Based on Soft-Features.” <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2006)</i>, vol.
    1, 2006, p. I, doi:<a href="https://doi.org/10.1109/ICASSP.2006.1659984">10.1109/ICASSP.2006.1659984</a>.
  short: 'V. Ion, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
    Speech and Signal Processing (ICASSP 2006), 2006, p. I.'
date_created: 2019-07-12T05:28:58Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2006.1659984
intvolume: '         1'
keyword:
- distributed speech recognition
- least mean squares methods
- MAP estimate
- maximum likelihood estimation
- MMSE estimate
- packet loss compensation scheme
- packet switched communication
- posteriori probability density function
- robust error mitigation method
- soft-features
- speech recognition
- table lookup
- voice communication
- wireless channels
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/IoHa06-2.pdf
oa: '1'
page: I
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
  (ICASSP 2006)
status: public
title: An Inexpensive Packet Loss Compensation Scheme for Distributed Speech Recognition
  Based on Soft-Features
type: conference
user_id: '44006'
volume: 1
year: '2006'
...
---
_id: '11825'
abstract:
- lang: eng
  text: In this paper, we propose an enhanced error concealment strategy at the server
    side of a distributed speech recognition (DSR) system, which is fully compatible
    with the existing DSR standard. It is based on a Bayesian approach, where the
    a posteriori probability density of the error-free feature vector is computed,
    given all received feature vectors which are possibly corrupted by transmission
    errors. Rather than computing a point estimate, such as the MMSE estimate, and
    plugging it into the Bayesian decision rule, we employ uncertainty decoding, which
    results in an integration over the uncertainty in the feature domain. In a typical
    scenario the communication between the thin client, often a mobile device, and
    the recognition server spreads across heterogeneous networks. Both bit errors
    on circuit-switched links and lost data packets on IP connections are mitigated
    by our approach in a unified manner. The experiments reveal improved robustness
    both for small- and large-vocabulary recognition tasks.
author:
- first_name: Valentin
  full_name: Ion, Valentin
  last_name: Ion
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: Ion V, Haeb-Umbach R. Uncertainty decoding for distributed speech recognition
    over error-prone networks. <i>Speech Communication</i>. 2006;48(11):1435-1446.
    doi:<a href="https://doi.org/10.1016/j.specom.2006.03.007">10.1016/j.specom.2006.03.007</a>
  apa: Ion, V., &#38; Haeb-Umbach, R. (2006). Uncertainty decoding for distributed
    speech recognition over error-prone networks. <i>Speech Communication</i>, <i>48</i>(11),
    1435–1446. <a href="https://doi.org/10.1016/j.specom.2006.03.007">https://doi.org/10.1016/j.specom.2006.03.007</a>
  bibtex: '@article{Ion_Haeb-Umbach_2006, title={Uncertainty decoding for distributed
    speech recognition over error-prone networks}, volume={48}, DOI={<a href="https://doi.org/10.1016/j.specom.2006.03.007">10.1016/j.specom.2006.03.007</a>},
    number={11}, journal={Speech Communication}, author={Ion, Valentin and Haeb-Umbach,
    Reinhold}, year={2006}, pages={1435–1446} }'
  chicago: 'Ion, Valentin, and Reinhold Haeb-Umbach. “Uncertainty Decoding for Distributed
    Speech Recognition over Error-Prone Networks.” <i>Speech Communication</i> 48,
    no. 11 (2006): 1435–46. <a href="https://doi.org/10.1016/j.specom.2006.03.007">https://doi.org/10.1016/j.specom.2006.03.007</a>.'
  ieee: V. Ion and R. Haeb-Umbach, “Uncertainty decoding for distributed speech recognition
    over error-prone networks,” <i>Speech Communication</i>, vol. 48, no. 11, pp.
    1435–1446, 2006.
  mla: Ion, Valentin, and Reinhold Haeb-Umbach. “Uncertainty Decoding for Distributed
    Speech Recognition over Error-Prone Networks.” <i>Speech Communication</i>, vol.
    48, no. 11, 2006, pp. 1435–46, doi:<a href="https://doi.org/10.1016/j.specom.2006.03.007">10.1016/j.specom.2006.03.007</a>.
  short: V. Ion, R. Haeb-Umbach, Speech Communication 48 (2006) 1435–1446.
date_created: 2019-07-12T05:28:59Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
doi: 10.1016/j.specom.2006.03.007
intvolume: '        48'
issue: '11'
keyword:
- Channel error robustness
- Distributed speech recognition
- Soft features
- Uncertainty decoding
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/IoHa06-3.pdf
oa: '1'
page: 1435-1446
publication: Speech Communication
status: public
title: Uncertainty decoding for distributed speech recognition over error-prone networks
type: journal_article
user_id: '44006'
volume: 48
year: '2006'
...
---
_id: '11826'
abstract:
- lang: eng
  text: The accuracy of distributed speech recognition has been shown to be very sensitive
    to errors occurring during transmission. One reason for this is that the classifier,
    usually trained under error free conditions, is unable to cope with the mismatch
    between an error free and error prone channel. In this paper we present a novel
    decision rule for classification which is able to account for channel errors.
    To achieve this, the classical Bayesian speech recognition approach has been reformulated
    for the server side, where the observation is known only to the extent, as is
    given by its a posteriori density function. We present a method to estimate the
    a posteriori density which is based on a Markov model of the source, which captures
    correlations of both static and dynamic features. A practical implementation is
    given, accompanied by experimental results for distributed speech recognition
    over an IP-network.
author:
- first_name: Valentin
  full_name: Ion, Valentin
  last_name: Ion
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Ion V, Haeb-Umbach R. Improved Source Modeling and Predictive Classification
    for Channel Robust Speech Recognition. In: <i>Interspeech 2006</i>. ; 2006.'
  apa: Ion, V., &#38; Haeb-Umbach, R. (2006). Improved Source Modeling and Predictive
    Classification for Channel Robust Speech Recognition. In <i>Interspeech 2006</i>.
  bibtex: '@inproceedings{Ion_Haeb-Umbach_2006, title={Improved Source Modeling and
    Predictive Classification for Channel Robust Speech Recognition}, booktitle={Interspeech
    2006}, author={Ion, Valentin and Haeb-Umbach, Reinhold}, year={2006} }'
  chicago: Ion, Valentin, and Reinhold Haeb-Umbach. “Improved Source Modeling and
    Predictive Classification for Channel Robust Speech Recognition.” In <i>Interspeech
    2006</i>, 2006.
  ieee: V. Ion and R. Haeb-Umbach, “Improved Source Modeling and Predictive Classification
    for Channel Robust Speech Recognition,” in <i>Interspeech 2006</i>, 2006.
  mla: Ion, Valentin, and Reinhold Haeb-Umbach. “Improved Source Modeling and Predictive
    Classification for Channel Robust Speech Recognition.” <i>Interspeech 2006</i>,
    2006.
  short: 'V. Ion, R. Haeb-Umbach, in: Interspeech 2006, 2006.'
date_created: 2019-07-12T05:29:00Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/IoSc06.pdf
oa: '1'
publication: Interspeech 2006
status: public
title: Improved Source Modeling and Predictive Classification for Channel Robust Speech
  Recognition
type: conference
user_id: '44006'
year: '2006'
...
---
_id: '11884'
abstract:
- lang: eng
  text: In this paper we present the design of a particle filter for post filtering
    instantaneous positioning estimates of GSM mobile terminals. The instantaneous
    estimates are obtained by comparing signal power levels, which are reported by
    the mobile terminal to the base station, with a database of predictions using
    a novel statistically motivated similarity measure. Unlike a simple Euclidian
    distance measure, the proposed scheme incorporates inherent information about
    signal power level measurements requested by the serving base station but not
    reported by the mobile terminal. Furthermore, we show how the Monte Carlo method
    of particle filtering helps to obtain better position estimates and, surprisingly,
    also helps to reduce the computational complexity. Results are presented for real
    field data.
author:
- first_name: Sven
  full_name: Peschke, Sven
  last_name: Peschke
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Peschke S, Haeb-Umbach R. A Probabilistic Similarity Measure and a Non-Linear
    Post-Filter for Mobile Phone Positioning using GSM Signal Power Measurements.
    In: <i>European Navigation Conference \&#38; Exhibition (ENC 2006)</i>. ; 2006.'
  apa: Peschke, S., &#38; Haeb-Umbach, R. (2006). A Probabilistic Similarity Measure
    and a Non-Linear Post-Filter for Mobile Phone Positioning using GSM Signal Power
    Measurements. In <i>European Navigation Conference \&#38; Exhibition (ENC 2006)</i>.
  bibtex: '@inproceedings{Peschke_Haeb-Umbach_2006, title={A Probabilistic Similarity
    Measure and a Non-Linear Post-Filter for Mobile Phone Positioning using GSM Signal
    Power Measurements}, booktitle={European Navigation Conference \&#38; Exhibition
    (ENC 2006)}, author={Peschke, Sven and Haeb-Umbach, Reinhold}, year={2006} }'
  chicago: Peschke, Sven, and Reinhold Haeb-Umbach. “A Probabilistic Similarity Measure
    and a Non-Linear Post-Filter for Mobile Phone Positioning Using GSM Signal Power
    Measurements.” In <i>European Navigation Conference \&#38; Exhibition (ENC 2006)</i>,
    2006.
  ieee: S. Peschke and R. Haeb-Umbach, “A Probabilistic Similarity Measure and a Non-Linear
    Post-Filter for Mobile Phone Positioning using GSM Signal Power Measurements,”
    in <i>European Navigation Conference \&#38; Exhibition (ENC 2006)</i>, 2006.
  mla: Peschke, Sven, and Reinhold Haeb-Umbach. “A Probabilistic Similarity Measure
    and a Non-Linear Post-Filter for Mobile Phone Positioning Using GSM Signal Power
    Measurements.” <i>European Navigation Conference \&#38; Exhibition (ENC 2006)</i>,
    2006.
  short: 'S. Peschke, R. Haeb-Umbach, in: European Navigation Conference \&#38; Exhibition
    (ENC 2006), 2006.'
date_created: 2019-07-12T05:30:07Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/PeHa06.pdf
oa: '1'
publication: European Navigation Conference \& Exhibition (ENC 2006)
status: public
title: A Probabilistic Similarity Measure and a Non-Linear Post-Filter for Mobile
  Phone Positioning using GSM Signal Power Measurements
type: conference
user_id: '44006'
year: '2006'
...
---
_id: '11885'
abstract:
- lang: eng
  text: In this paper we present a novel and statistically motivated similarity measure
    for database assisted positioning of GSM mobile terminals by evaluating signal
    power level reports which are transmitted regulary. Unlike a simple Euclidian
    distance measure, the proposed scheme incorporates inherent information about
    signal power level measurements requested by the serving base station but not
    reported by the mobile terminal. Furthermore we show how the Monte Carlo method
    of nonlinear post filtering using particle filtering helps to obtain better position
    estimates and surprisingly also helps to reduce the computational complexity.
    Results are presented for real field data.
author:
- first_name: Sven
  full_name: Peschke, Sven
  last_name: Peschke
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Peschke S, Haeb-Umbach R. Particle Filtering of Database assisted Positioning
    Estimates using a novel Similarity Measure for GSM Signal Power Level Measurements.
    In: <i>3rd Workshop on Positioning Navigation and Communication (WPNC 2006)</i>.
    ; 2006.'
  apa: Peschke, S., &#38; Haeb-Umbach, R. (2006). Particle Filtering of Database assisted
    Positioning Estimates using a novel Similarity Measure for GSM Signal Power Level
    Measurements. In <i>3rd Workshop on Positioning Navigation and Communication (WPNC
    2006)</i>.
  bibtex: '@inproceedings{Peschke_Haeb-Umbach_2006, title={Particle Filtering of Database
    assisted Positioning Estimates using a novel Similarity Measure for GSM Signal
    Power Level Measurements}, booktitle={3rd Workshop on Positioning Navigation and
    Communication (WPNC 2006)}, author={Peschke, Sven and Haeb-Umbach, Reinhold},
    year={2006} }'
  chicago: Peschke, Sven, and Reinhold Haeb-Umbach. “Particle Filtering of Database
    Assisted Positioning Estimates Using a Novel Similarity Measure for GSM Signal
    Power Level Measurements.” In <i>3rd Workshop on Positioning Navigation and Communication
    (WPNC 2006)</i>, 2006.
  ieee: S. Peschke and R. Haeb-Umbach, “Particle Filtering of Database assisted Positioning
    Estimates using a novel Similarity Measure for GSM Signal Power Level Measurements,”
    in <i>3rd Workshop on Positioning Navigation and Communication (WPNC 2006)</i>,
    2006.
  mla: Peschke, Sven, and Reinhold Haeb-Umbach. “Particle Filtering of Database Assisted
    Positioning Estimates Using a Novel Similarity Measure for GSM Signal Power Level
    Measurements.” <i>3rd Workshop on Positioning Navigation and Communication (WPNC
    2006)</i>, 2006.
  short: 'S. Peschke, R. Haeb-Umbach, in: 3rd Workshop on Positioning Navigation and
    Communication (WPNC 2006), 2006.'
date_created: 2019-07-12T05:30:08Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/PeHa06-1.pdf
oa: '1'
publication: 3rd Workshop on Positioning Navigation and Communication (WPNC 2006)
status: public
title: Particle Filtering of Database assisted Positioning Estimates using a novel
  Similarity Measure for GSM Signal Power Level Measurements
type: conference
user_id: '44006'
year: '2006'
...
---
_id: '11928'
abstract:
- lang: eng
  text: 'Broadband adaptive beamformers, which use a narrowband SNR-maximization optimization
    criterion for noise reduction, typically cause distortions of the desired speech
    signal at the beamformer output. In this paper two methods are investigated to
    control the speech distortion by comparing the eigenvector beamformer with a maximum
    likelihood beamformer: One is an analytic solution for the ideal case of absence
    of reverberation and the other one is a statistically motivated approach. We use
    the recently introduced gradient-ascent algorithm for adaptive principal eigenvector
    beamforming and then normalize the filter coefficients by the proposed distortion
    control methods. Experimental results in terms of the achievable SNR gain and
    a perceptual speech quality measure are given for the normalized eigenvector beamformer
    and are compared to standard beamforming methods.'
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. Mehrkanalige Sprachsignalverarbeitung durch adaptives
    Eigenbeamforming fuer Freisprecheinrichtungen im Kraftfahrzeug. In: <i>32. Deutsche
    Jahrestagung Fuer Akustik (DAGA 2006)</i>. ; 2006.'
  apa: Warsitz, E., &#38; Haeb-Umbach, R. (2006). Mehrkanalige Sprachsignalverarbeitung
    durch adaptives Eigenbeamforming fuer Freisprecheinrichtungen im Kraftfahrzeug.
    In <i>32. Deutsche Jahrestagung fuer Akustik (DAGA 2006)</i>.
  bibtex: '@inproceedings{Warsitz_Haeb-Umbach_2006, title={Mehrkanalige Sprachsignalverarbeitung
    durch adaptives Eigenbeamforming fuer Freisprecheinrichtungen im Kraftfahrzeug},
    booktitle={32. Deutsche Jahrestagung fuer Akustik (DAGA 2006)}, author={Warsitz,
    Ernst and Haeb-Umbach, Reinhold}, year={2006} }'
  chicago: Warsitz, Ernst, and Reinhold Haeb-Umbach. “Mehrkanalige Sprachsignalverarbeitung
    Durch Adaptives Eigenbeamforming Fuer Freisprecheinrichtungen Im Kraftfahrzeug.”
    In <i>32. Deutsche Jahrestagung Fuer Akustik (DAGA 2006)</i>, 2006.
  ieee: E. Warsitz and R. Haeb-Umbach, “Mehrkanalige Sprachsignalverarbeitung durch
    adaptives Eigenbeamforming fuer Freisprecheinrichtungen im Kraftfahrzeug,” in
    <i>32. Deutsche Jahrestagung fuer Akustik (DAGA 2006)</i>, 2006.
  mla: Warsitz, Ernst, and Reinhold Haeb-Umbach. “Mehrkanalige Sprachsignalverarbeitung
    Durch Adaptives Eigenbeamforming Fuer Freisprecheinrichtungen Im Kraftfahrzeug.”
    <i>32. Deutsche Jahrestagung Fuer Akustik (DAGA 2006)</i>, 2006.
  short: 'E. Warsitz, R. Haeb-Umbach, in: 32. Deutsche Jahrestagung Fuer Akustik (DAGA
    2006), 2006.'
date_created: 2019-07-12T05:30:58Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/WaHa06-1.pdf
oa: '1'
publication: 32. Deutsche Jahrestagung fuer Akustik (DAGA 2006)
status: public
title: Mehrkanalige Sprachsignalverarbeitung durch adaptives Eigenbeamforming fuer
  Freisprecheinrichtungen im Kraftfahrzeug
type: conference
user_id: '44006'
year: '2006'
...
---
_id: '11929'
abstract:
- lang: eng
  text: 'Broadband adaptive beamformers, which use a narrowband SNR-maximization optimization
    criterion for noise reduction, typically cause distortions of the desired speech
    signal at the beamformer output. In this paper two methodsare investigated to
    control the speech distortion by comparing the eigenvector beamformer with a maximum
    likelihood beamformer: One is an analytic solution for the ideal case of absence
    of reverberation and the other one is a statistically motivated approach. We use
    the recently introduced gradient-ascent algorithm for adaptive principal eigenvector
    beamforming and then normalize the filter coefficient s by the proposed distortion
    control methods. Experimental results in terms of the achievable SNR gain and
    a perceptual speech quality measure are given for the normalized eigenvector beamformer
    and are compared to standard beamforming methods.'
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. Controlling Speech Distortion in Adaptive Frequency-Domain
    Principal Eigenvector Beamforming. In: <i>International Workshop on Acoustic Echo
    and Noise Control (IWAENC 2006)</i>. ; 2006.'
  apa: Warsitz, E., &#38; Haeb-Umbach, R. (2006). Controlling Speech Distortion in
    Adaptive Frequency-Domain Principal Eigenvector Beamforming. In <i>International
    Workshop on Acoustic Echo and Noise Control (IWAENC 2006)</i>.
  bibtex: '@inproceedings{Warsitz_Haeb-Umbach_2006, title={Controlling Speech Distortion
    in Adaptive Frequency-Domain Principal Eigenvector Beamforming}, booktitle={International
    Workshop on Acoustic Echo and Noise Control (IWAENC 2006)}, author={Warsitz, Ernst
    and Haeb-Umbach, Reinhold}, year={2006} }'
  chicago: Warsitz, Ernst, and Reinhold Haeb-Umbach. “Controlling Speech Distortion
    in Adaptive Frequency-Domain Principal Eigenvector Beamforming.” In <i>International
    Workshop on Acoustic Echo and Noise Control (IWAENC 2006)</i>, 2006.
  ieee: E. Warsitz and R. Haeb-Umbach, “Controlling Speech Distortion in Adaptive
    Frequency-Domain Principal Eigenvector Beamforming,” in <i>International Workshop
    on Acoustic Echo and Noise Control (IWAENC 2006)</i>, 2006.
  mla: Warsitz, Ernst, and Reinhold Haeb-Umbach. “Controlling Speech Distortion in
    Adaptive Frequency-Domain Principal Eigenvector Beamforming.” <i>International
    Workshop on Acoustic Echo and Noise Control (IWAENC 2006)</i>, 2006.
  short: 'E. Warsitz, R. Haeb-Umbach, in: International Workshop on Acoustic Echo
    and Noise Control (IWAENC 2006), 2006.'
date_created: 2019-07-12T05:30:59Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/WaHa06-2.pdf
oa: '1'
publication: International Workshop on Acoustic Echo and Noise Control (IWAENC 2006)
status: public
title: Controlling Speech Distortion in Adaptive Frequency-Domain Principal Eigenvector
  Beamforming
type: conference
user_id: '44006'
year: '2006'
...
---
_id: '11942'
abstract:
- lang: eng
  text: Es wird ein marginalisiertes Partikelfilter beschrieben, das zur einkanaligen
    Sprachsignalverbesserung mit einem nichtlinearen dynamischen Zustandsmodell eingesetzt
    werden soll. Das System besteht aus einem Partikelfilter zum Tracking von LSP-Parametern
    und einem Kalman-Filter fuer jedes Partikel, das zur Sprachsignalverbesserung
    verwendet wird. In unserem Ansatz wird angenommen, dass die Parameter in kurzen
    Sprachsignalbloecken konstant sind, waehrend das Sprachsignal sich mit jedem Abtastwert
    aendert. Bei weissem Rauschen werden aehnliche SNR-Gewinne wie mit einem Kalman-EM-iterative
    Algorithmus erzielt, waehrend das Hintergrundrauschen und die Log-spektrale Distanz
    etwas geringer sind. Mit einem erweiterten Zustandsmodell wurden auch Untersuchungen
    fuer farbiges Rauschen durchgefuehrt.
author:
- first_name: Stefan
  full_name: Windmann, Stefan
  last_name: Windmann
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Windmann S, Haeb-Umbach R. Einkanalige Sprachsignalverbesserung mit Hilfe
    eines marginalisierten Partikelfilters. In: <i>7. ITG-Fachtagung Sprachkommunikation</i>.
    ; 2006.'
  apa: Windmann, S., &#38; Haeb-Umbach, R. (2006). Einkanalige Sprachsignalverbesserung
    mit Hilfe eines marginalisierten Partikelfilters. In <i>7. ITG-Fachtagung Sprachkommunikation</i>.
  bibtex: '@inproceedings{Windmann_Haeb-Umbach_2006, title={Einkanalige Sprachsignalverbesserung
    mit Hilfe eines marginalisierten Partikelfilters}, booktitle={7. ITG-Fachtagung
    Sprachkommunikation}, author={Windmann, Stefan and Haeb-Umbach, Reinhold}, year={2006}
    }'
  chicago: Windmann, Stefan, and Reinhold Haeb-Umbach. “Einkanalige Sprachsignalverbesserung
    Mit Hilfe Eines Marginalisierten Partikelfilters.” In <i>7. ITG-Fachtagung Sprachkommunikation</i>,
    2006.
  ieee: S. Windmann and R. Haeb-Umbach, “Einkanalige Sprachsignalverbesserung mit
    Hilfe eines marginalisierten Partikelfilters,” in <i>7. ITG-Fachtagung Sprachkommunikation</i>,
    2006.
  mla: Windmann, Stefan, and Reinhold Haeb-Umbach. “Einkanalige Sprachsignalverbesserung
    Mit Hilfe Eines Marginalisierten Partikelfilters.” <i>7. ITG-Fachtagung Sprachkommunikation</i>,
    2006.
  short: 'S. Windmann, R. Haeb-Umbach, in: 7. ITG-Fachtagung Sprachkommunikation,
    2006.'
date_created: 2019-07-12T05:31:14Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/WiHa06-1.pdf
oa: '1'
publication: 7. ITG-Fachtagung Sprachkommunikation
status: public
title: Einkanalige Sprachsignalverbesserung mit Hilfe eines marginalisierten Partikelfilters
type: conference
user_id: '44006'
year: '2006'
...
---
_id: '11943'
abstract:
- lang: eng
  text: A marginalized particle filter is proposed for performing single channel speech
    enhancement with a non-linear dynamic state model. The system consists of a particle
    filter for tracking line spectral pair (LSP) parameters and a Kalman filter per
    particle for speech enhancement. The state model for the LSPs has been learnt
    on clean speech training data. In our approach parameters and speech samples are
    processed at different time scales by assuming the parameters to be constant for
    small blocks of data. Further enhancement is obtained by an iteration which can
    be applied on these small blocks. The experiments show that similar SNR gains
    are obtained as with the Kalman-LM-iterative algorithm. However better values
    of the noise level and the log-spectral distance are achieved
author:
- first_name: Stefan
  full_name: Windmann, Stefan
  last_name: Windmann
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Windmann S, Haeb-Umbach R. Iterative Speech Enhancement using a Non-Linear
    Dynamic State Model of Speech and its Parameters. In: <i>IEEE International Conference
    on Acoustics, Speech and Signal Processing (ICASSP 2006)</i>. Vol 1. ; 2006:I.
    doi:<a href="https://doi.org/10.1109/ICASSP.2006.1660058">10.1109/ICASSP.2006.1660058</a>'
  apa: Windmann, S., &#38; Haeb-Umbach, R. (2006). Iterative Speech Enhancement using
    a Non-Linear Dynamic State Model of Speech and its Parameters. In <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2006)</i> (Vol.
    1, p. I). <a href="https://doi.org/10.1109/ICASSP.2006.1660058">https://doi.org/10.1109/ICASSP.2006.1660058</a>
  bibtex: '@inproceedings{Windmann_Haeb-Umbach_2006, title={Iterative Speech Enhancement
    using a Non-Linear Dynamic State Model of Speech and its Parameters}, volume={1},
    DOI={<a href="https://doi.org/10.1109/ICASSP.2006.1660058">10.1109/ICASSP.2006.1660058</a>},
    booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
    (ICASSP 2006)}, author={Windmann, Stefan and Haeb-Umbach, Reinhold}, year={2006},
    pages={I} }'
  chicago: Windmann, Stefan, and Reinhold Haeb-Umbach. “Iterative Speech Enhancement
    Using a Non-Linear Dynamic State Model of Speech and Its Parameters.” In <i>IEEE
    International Conference on Acoustics, Speech and Signal Processing (ICASSP 2006)</i>,
    1:I, 2006. <a href="https://doi.org/10.1109/ICASSP.2006.1660058">https://doi.org/10.1109/ICASSP.2006.1660058</a>.
  ieee: S. Windmann and R. Haeb-Umbach, “Iterative Speech Enhancement using a Non-Linear
    Dynamic State Model of Speech and its Parameters,” in <i>IEEE International Conference
    on Acoustics, Speech and Signal Processing (ICASSP 2006)</i>, 2006, vol. 1, p.
    I.
  mla: Windmann, Stefan, and Reinhold Haeb-Umbach. “Iterative Speech Enhancement Using
    a Non-Linear Dynamic State Model of Speech and Its Parameters.” <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2006)</i>, vol.
    1, 2006, p. I, doi:<a href="https://doi.org/10.1109/ICASSP.2006.1660058">10.1109/ICASSP.2006.1660058</a>.
  short: 'S. Windmann, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
    Speech and Signal Processing (ICASSP 2006), 2006, p. I.'
date_created: 2019-07-12T05:31:15Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2006.1660058
intvolume: '         1'
keyword:
- clean speech training data
- iterative methods
- iterative speech enhancement
- Kalman filter
- Kalman filters
- Kalman-LM-iterative algorithm
- line spectral pair parameters
- log-spectral distance
- marginalized particle filter
- noise level
- nonlinear dynamic state speech model
- particle filtering (numerical methods)
- single channel speech enhancement
- SNR gains
- speech enhancement
- speech samples
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/WiHa06-2.pdf
oa: '1'
page: I
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
  (ICASSP 2006)
status: public
title: Iterative Speech Enhancement using a Non-Linear Dynamic State Model of Speech
  and its Parameters
type: conference
user_id: '44006'
volume: 1
year: '2006'
...
---
_id: '11894'
abstract:
- lang: eng
  text: In this paper we consider the problem of detecting speaker changes in audio
    signals recorded by distant microphones. It is shown that the possibility to exploit
    the spatial separation of speakers more than makes up the degradation in detection
    accuracy due to the increased source-to-sensor distance compared to close-talking
    microphones. Speaker direction information is derived from the filter coefficients
    of an adaptive Filter-and-Sum Beamformer and is combined with BIC analysis. The
    experimental results reveal significant improvements compared to BIC-only change
    detection, be it with the distant or close-talking microphone.
author:
- first_name: Joerg
  full_name: Schmalenstroeer, Joerg
  id: '460'
  last_name: Schmalenstroeer
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Schmalenstroeer J, Haeb-Umbach R. Online Speaker Change Detection by Combining
    BIC with Microphone Array Beamforming. In: <i>Interspeech 2006</i>. ; 2006.'
  apa: Schmalenstroeer, J., &#38; Haeb-Umbach, R. (2006). Online Speaker Change Detection
    by Combining BIC with Microphone Array Beamforming. <i>Interspeech 2006</i>.
  bibtex: '@inproceedings{Schmalenstroeer_Haeb-Umbach_2006, title={Online Speaker
    Change Detection by Combining BIC with Microphone Array Beamforming}, booktitle={Interspeech
    2006}, author={Schmalenstroeer, Joerg and Haeb-Umbach, Reinhold}, year={2006}
    }'
  chicago: Schmalenstroeer, Joerg, and Reinhold Haeb-Umbach. “Online Speaker Change
    Detection by Combining BIC with Microphone Array Beamforming.” In <i>Interspeech
    2006</i>, 2006.
  ieee: J. Schmalenstroeer and R. Haeb-Umbach, “Online Speaker Change Detection by
    Combining BIC with Microphone Array Beamforming,” 2006.
  mla: Schmalenstroeer, Joerg, and Reinhold Haeb-Umbach. “Online Speaker Change Detection
    by Combining BIC with Microphone Array Beamforming.” <i>Interspeech 2006</i>,
    2006.
  short: 'J. Schmalenstroeer, R. Haeb-Umbach, in: Interspeech 2006, 2006.'
date_created: 2019-07-12T05:30:19Z
date_updated: 2023-10-26T08:09:54Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2006/ScHa06.pdf
oa: '1'
publication: Interspeech 2006
quality_controlled: '1'
status: public
title: Online Speaker Change Detection by Combining BIC with Microphone Array Beamforming
type: conference
user_id: '460'
year: '2006'
...
---
_id: '11803'
abstract:
- lang: eng
  text: In this paper we propose a novel adaptation algorithm for Filter-and-Sum beamforming
    in spatially correlated noise. Deterministic and stochastic gradient ascent algorithms
    are derived from a constrained optimization problem, which iteratively estimate
    the principal eigenvecto r of a generalized eigenvalue problem. The method does
    not require an explicit estimation of the speaker location. It is shown that the
    well-known Delay-and-Sum beamformer and the previously introduced Filter-and-Sum
    beamformer in spatially white noise are obtained as special cases. Further, bounds
    on the maximally achievable SNR gains are derived and it is shown that the proposed
    adaptation algorithm is able to approach these performance bounds.
author:
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
- first_name: Ernst
  full_name: Warsitz, Ernst
  last_name: Warsitz
citation:
  ama: 'Haeb-Umbach R, Warsitz E. Adaptive Filter-and-Sum Beamforming in Spatially
    Correlated Noise. In: <i>International Workshop on Acoustic Echo and Noise Control
    (IWAENC 2005)</i>. ; 2005.'
  apa: Haeb-Umbach, R., &#38; Warsitz, E. (2005). Adaptive Filter-and-Sum Beamforming
    in Spatially Correlated Noise. In <i>International Workshop on Acoustic Echo and
    Noise Control (IWAENC 2005)</i>.
  bibtex: '@inproceedings{Haeb-Umbach_Warsitz_2005, title={Adaptive Filter-and-Sum
    Beamforming in Spatially Correlated Noise}, booktitle={International Workshop
    on Acoustic Echo and Noise Control (IWAENC 2005)}, author={Haeb-Umbach, Reinhold
    and Warsitz, Ernst}, year={2005} }'
  chicago: Haeb-Umbach, Reinhold, and Ernst Warsitz. “Adaptive Filter-and-Sum Beamforming
    in Spatially Correlated Noise.” In <i>International Workshop on Acoustic Echo
    and Noise Control (IWAENC 2005)</i>, 2005.
  ieee: R. Haeb-Umbach and E. Warsitz, “Adaptive Filter-and-Sum Beamforming in Spatially
    Correlated Noise,” in <i>International Workshop on Acoustic Echo and Noise Control
    (IWAENC 2005)</i>, 2005.
  mla: Haeb-Umbach, Reinhold, and Ernst Warsitz. “Adaptive Filter-and-Sum Beamforming
    in Spatially Correlated Noise.” <i>International Workshop on Acoustic Echo and
    Noise Control (IWAENC 2005)</i>, 2005.
  short: 'R. Haeb-Umbach, E. Warsitz, in: International Workshop on Acoustic Echo
    and Noise Control (IWAENC 2005), 2005.'
date_created: 2019-07-12T05:28:33Z
date_updated: 2022-01-06T06:51:09Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2005/HaWa05.pdf
oa: '1'
publication: International Workshop on Acoustic Echo and Noise Control (IWAENC 2005)
status: public
title: Adaptive Filter-and-Sum Beamforming in Spatially Correlated Noise
type: conference
user_id: '44006'
year: '2005'
...
---
_id: '11827'
abstract:
- lang: eng
  text: The transmission errors in a wireless or packet oriented network may dramatically
    decrease the performance of a distributed speech recognition DSR) system. Error
    concealment has been shown to be an effective way to mantain an acceptable word
    error rate when dealing with error prone communication channels. In this paper
    we propose an extension of our previously introduced soft features approach for
    the case that the soft-output of the channel decoder is not available at the server
    side of the DSR system. We found a simple method to estimate bit reliability information
    which still gives good speech recognition results. It is shown that some other
    error concealment schemes turn out to be special cases of the method proposed
    here.
author:
- first_name: Valentin
  full_name: Ion, Valentin
  last_name: Ion
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Ion V, Haeb-Umbach R. A Unified Probabilistic Approach to Error Concealment
    for Distributed Speech Recognition. In: <i>Interspeech 2005</i>. ; 2005.'
  apa: Ion, V., &#38; Haeb-Umbach, R. (2005). A Unified Probabilistic Approach to
    Error Concealment for Distributed Speech Recognition. In <i>Interspeech 2005</i>.
  bibtex: '@inproceedings{Ion_Haeb-Umbach_2005, title={A Unified Probabilistic Approach
    to Error Concealment for Distributed Speech Recognition}, booktitle={Interspeech
    2005}, author={Ion, Valentin and Haeb-Umbach, Reinhold}, year={2005} }'
  chicago: Ion, Valentin, and Reinhold Haeb-Umbach. “A Unified Probabilistic Approach
    to Error Concealment for Distributed Speech Recognition.” In <i>Interspeech 2005</i>,
    2005.
  ieee: V. Ion and R. Haeb-Umbach, “A Unified Probabilistic Approach to Error Concealment
    for Distributed Speech Recognition,” in <i>Interspeech 2005</i>, 2005.
  mla: Ion, Valentin, and Reinhold Haeb-Umbach. “A Unified Probabilistic Approach
    to Error Concealment for Distributed Speech Recognition.” <i>Interspeech 2005</i>,
    2005.
  short: 'V. Ion, R. Haeb-Umbach, in: Interspeech 2005, 2005.'
date_created: 2019-07-12T05:29:01Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2005/IoHa05-1.pdf
oa: '1'
publication: Interspeech 2005
status: public
title: A Unified Probabilistic Approach to Error Concealment for Distributed Speech
  Recognition
type: conference
user_id: '44006'
year: '2005'
...
---
_id: '11828'
abstract:
- lang: eng
  text: 'In this paper we present a comparison of the recently proposed Soft-Feature
    Distributed Speech Recognition (SFDSR) with the two evaluated candidate codecs
    for Speech Enabled Services over wireless networks: Adaptive Multirate Codec (AMR)
    and the ETSI Extended Advanced Front-End for Distributed Speech Recognition (XAFE).
    It is shown that SFDSR achieves the best recognition performance on a simulated
    GSM transmission, followed by XAFE and AMR.We also present some new results concerning
    SFDSR which demonstrate the versatility of the approach. Further, a simple method
    is introduced which considerably reduces the computational effort.'
author:
- first_name: Valentin
  full_name: Ion, Valentin
  last_name: Ion
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Ion V, Haeb-Umbach R. A Comparison of Soft-Feature Distributed Speech Recognition
    with Candidate Codecs for Speech Enabled Mobile Services. In: <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)</i>. Vol 1.
    ; 2005:333-336. doi:<a href="https://doi.org/10.1109/ICASSP.2005.1415118">10.1109/ICASSP.2005.1415118</a>'
  apa: Ion, V., &#38; Haeb-Umbach, R. (2005). A Comparison of Soft-Feature Distributed
    Speech Recognition with Candidate Codecs for Speech Enabled Mobile Services. In
    <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP
    2005)</i> (Vol. 1, pp. 333–336). <a href="https://doi.org/10.1109/ICASSP.2005.1415118">https://doi.org/10.1109/ICASSP.2005.1415118</a>
  bibtex: '@inproceedings{Ion_Haeb-Umbach_2005, title={A Comparison of Soft-Feature
    Distributed Speech Recognition with Candidate Codecs for Speech Enabled Mobile
    Services}, volume={1}, DOI={<a href="https://doi.org/10.1109/ICASSP.2005.1415118">10.1109/ICASSP.2005.1415118</a>},
    booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
    (ICASSP 2005)}, author={Ion, Valentin and Haeb-Umbach, Reinhold}, year={2005},
    pages={333–336} }'
  chicago: Ion, Valentin, and Reinhold Haeb-Umbach. “A Comparison of Soft-Feature
    Distributed Speech Recognition with Candidate Codecs for Speech Enabled Mobile
    Services.” In <i>IEEE International Conference on Acoustics, Speech and Signal
    Processing (ICASSP 2005)</i>, 1:333–36, 2005. <a href="https://doi.org/10.1109/ICASSP.2005.1415118">https://doi.org/10.1109/ICASSP.2005.1415118</a>.
  ieee: V. Ion and R. Haeb-Umbach, “A Comparison of Soft-Feature Distributed Speech
    Recognition with Candidate Codecs for Speech Enabled Mobile Services,” in <i>IEEE
    International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005)</i>,
    2005, vol. 1, pp. 333–336.
  mla: Ion, Valentin, and Reinhold Haeb-Umbach. “A Comparison of Soft-Feature Distributed
    Speech Recognition with Candidate Codecs for Speech Enabled Mobile Services.”
    <i>IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP
    2005)</i>, vol. 1, 2005, pp. 333–36, doi:<a href="https://doi.org/10.1109/ICASSP.2005.1415118">10.1109/ICASSP.2005.1415118</a>.
  short: 'V. Ion, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
    Speech and Signal Processing (ICASSP 2005), 2005, pp. 333–336.'
date_created: 2019-07-12T05:29:02Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2005.1415118
intvolume: '         1'
keyword:
- adaptive codes
- adaptive multirate codec
- AMR
- distributed speech recognition
- ETSI
- extended advanced front-end
- recognition performance
- SFDSR
- simulated GSM transmission
- soft-feature distributed speech recognition
- speech codecs
- speech coding
- speech recognition
- variable rate codes
- XAFE
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2005/IoHa05-2.pdf
oa: '1'
page: 333-336
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
  (ICASSP 2005)
status: public
title: A Comparison of Soft-Feature Distributed Speech Recognition with Candidate
  Codecs for Speech Enabled Mobile Services
type: conference
user_id: '44006'
volume: 1
year: '2005'
...