---
_id: '11806'
abstract:
- lang: eng
  text: Microphone arrays represent the basis for many challenging acoustic sensing
    tasks. The accuracy of techniques like beamforming directly depends on a precise
    knowledge of the relative positions of the sensors used. Unfortunately, for certain
    use cases manually measuring the geometry of an array is not feasible due to practical
    constraints. In this paper we present an approach to unsupervised shape calibration
    of microphone array networks. We developed a hierarchical procedure that first
    performs local shape calibration based on coherence analysis and then employs
    SRP-PHAT in a network calibration method. Practical experiments demonstrate the
    effectiveness of our approach especially for highly reverberant acoustic environments.
author:
- first_name: Marius
  full_name: Hennecke, Marius
  last_name: Hennecke
- first_name: Thomas
  full_name: Ploetz, Thomas
  last_name: Ploetz
- first_name: Gernot A.
  full_name: Fink, Gernot A.
  last_name: Fink
- 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: 'Hennecke M, Ploetz T, Fink GA, Schmalenstroeer J, Haeb-Umbach R. A hierarchical
    approach to unsupervised shape calibration of microphone array networks. In: <i>IEEE/SP
    15th Workshop on Statistical Signal Processing (SSP 2009)</i>. ; 2009:257-260.
    doi:<a href="https://doi.org/10.1109/SSP.2009.5278589">10.1109/SSP.2009.5278589</a>'
  apa: Hennecke, M., Ploetz, T., Fink, G. A., Schmalenstroeer, J., &#38; Haeb-Umbach,
    R. (2009). A hierarchical approach to unsupervised shape calibration of microphone
    array networks. <i>IEEE/SP 15th Workshop on Statistical Signal Processing (SSP
    2009)</i>, 257–260. <a href="https://doi.org/10.1109/SSP.2009.5278589">https://doi.org/10.1109/SSP.2009.5278589</a>
  bibtex: '@inproceedings{Hennecke_Ploetz_Fink_Schmalenstroeer_Haeb-Umbach_2009, title={A
    hierarchical approach to unsupervised shape calibration of microphone array networks},
    DOI={<a href="https://doi.org/10.1109/SSP.2009.5278589">10.1109/SSP.2009.5278589</a>},
    booktitle={IEEE/SP 15th Workshop on Statistical Signal Processing (SSP 2009)},
    author={Hennecke, Marius and Ploetz, Thomas and Fink, Gernot A. and Schmalenstroeer,
    Joerg and Haeb-Umbach, Reinhold}, year={2009}, pages={257–260} }'
  chicago: Hennecke, Marius, Thomas Ploetz, Gernot A. Fink, Joerg Schmalenstroeer,
    and Reinhold Haeb-Umbach. “A Hierarchical Approach to Unsupervised Shape Calibration
    of Microphone Array Networks.” In <i>IEEE/SP 15th Workshop on Statistical Signal
    Processing (SSP 2009)</i>, 257–60, 2009. <a href="https://doi.org/10.1109/SSP.2009.5278589">https://doi.org/10.1109/SSP.2009.5278589</a>.
  ieee: 'M. Hennecke, T. Ploetz, G. A. Fink, J. Schmalenstroeer, and R. Haeb-Umbach,
    “A hierarchical approach to unsupervised shape calibration of microphone array
    networks,” in <i>IEEE/SP 15th Workshop on Statistical Signal Processing (SSP 2009)</i>,
    2009, pp. 257–260, doi: <a href="https://doi.org/10.1109/SSP.2009.5278589">10.1109/SSP.2009.5278589</a>.'
  mla: Hennecke, Marius, et al. “A Hierarchical Approach to Unsupervised Shape Calibration
    of Microphone Array Networks.” <i>IEEE/SP 15th Workshop on Statistical Signal
    Processing (SSP 2009)</i>, 2009, pp. 257–60, doi:<a href="https://doi.org/10.1109/SSP.2009.5278589">10.1109/SSP.2009.5278589</a>.
  short: 'M. Hennecke, T. Ploetz, G.A. Fink, J. Schmalenstroeer, R. Haeb-Umbach, in:
    IEEE/SP 15th Workshop on Statistical Signal Processing (SSP 2009), 2009, pp. 257–260.'
date_created: 2019-07-12T05:28:37Z
date_updated: 2023-10-26T08:09:22Z
department:
- _id: '54'
doi: 10.1109/SSP.2009.5278589
keyword:
- acoustic sensing tasks
- array geometry
- calibration
- coherence analysis
- hierarchical procedure
- local shape calibration
- microphone array networks
- microphone arrays
- network calibration method
- sensor arrays
- SRP-PHAT
- unsupervised shape calibration
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2009/HePlFiScHa09.pdf
oa: '1'
page: 257-260
publication: IEEE/SP 15th Workshop on Statistical Signal Processing (SSP 2009)
quality_controlled: '1'
status: public
title: A hierarchical approach to unsupervised shape calibration of microphone array
  networks
type: conference
user_id: '460'
year: '2009'
...
---
_id: '11899'
abstract:
- lang: eng
  text: In this paper we present a system for identifying and localizingspeakers using
    distant microphone arrays and a steerablepan-tilt-zoom camera. Audio and video
    streams are processedin real-time to obtain the diarization information {grqq}who
    speakswhen and where'' with low latency to be used in advanced videoconferencing
    systems or user-adaptive interfaces. A key featureof the proposed system is to
    first glean information about thespeaker{\rq}s location and identity from the
    audio and visual datastreams separately and then to fuse these data in a probabilisticframework
    employing the Viterbi algorithm. Here, visual evidenceof a person is utilized
    through a priori state probabilities,while location and speaker change information
    are employedvia time-variant transition probablities. Experiments show thatvideo
    information yields a substantial improvement comparedto pure audio-based diarization.
author:
- first_name: Joerg
  full_name: Schmalenstroeer, Joerg
  id: '460'
  last_name: Schmalenstroeer
- first_name: Martin
  full_name: Kelling, Martin
  last_name: Kelling
- 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, Kelling M, Leutnant V, Haeb-Umbach R. Fusing Audio and
    Video Information for Online Speaker Diarization. In: <i>Interspeech 2009</i>.
    ; 2009.'
  apa: Schmalenstroeer, J., Kelling, M., Leutnant, V., &#38; Haeb-Umbach, R. (2009).
    Fusing Audio and Video Information for Online Speaker Diarization. <i>Interspeech
    2009</i>.
  bibtex: '@inproceedings{Schmalenstroeer_Kelling_Leutnant_Haeb-Umbach_2009, title={Fusing
    Audio and Video Information for Online Speaker Diarization}, booktitle={Interspeech
    2009}, author={Schmalenstroeer, Joerg and Kelling, Martin and Leutnant, Volker
    and Haeb-Umbach, Reinhold}, year={2009} }'
  chicago: Schmalenstroeer, Joerg, Martin Kelling, Volker Leutnant, and Reinhold Haeb-Umbach.
    “Fusing Audio and Video Information for Online Speaker Diarization.” In <i>Interspeech
    2009</i>, 2009.
  ieee: J. Schmalenstroeer, M. Kelling, V. Leutnant, and R. Haeb-Umbach, “Fusing Audio
    and Video Information for Online Speaker Diarization,” 2009.
  mla: Schmalenstroeer, Joerg, et al. “Fusing Audio and Video Information for Online
    Speaker Diarization.” <i>Interspeech 2009</i>, 2009.
  short: 'J. Schmalenstroeer, M. Kelling, V. Leutnant, R. Haeb-Umbach, in: Interspeech
    2009, 2009.'
date_created: 2019-07-12T05:30:24Z
date_updated: 2023-10-26T08:10:10Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2009/ScKeLeHa09.pdf
oa: '1'
publication: Interspeech 2009
quality_controlled: '1'
status: public
title: Fusing Audio and Video Information for Online Speaker Diarization
type: conference
user_id: '460'
year: '2009'
...
---
_id: '11776'
abstract:
- lang: eng
  text: 'The term uncertainty decoding has been phrased for a class of robustness
    enhancing algorithms in automatic speech recognition that replace point estimates
    and plug-in rules by posterior densities and optimal decision rules. While uncertainty
    can be incorporated in the model domain, in the feature domain, or even in both,
    we concentrate here on feature domain approaches as they tend to be computationally
    less demanding. We derive optimal decision rules in the presence of uncertain
    observations and discuss simplifications which result in computationally efficient
    realizations. The usefulness of the presented statistical framework is then exemplified
    for two types of realworld problems: The first is improving the robustness of
    speech recognition towards incomplete or corrupted feature vectors due to a lossy
    communication link between the speech capturing front end and the backend recognition
    engine. And the second is the well-known and extensively studied issue of improving
    the robustness of the recognizer towards environmental noise.'
author:
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: Haeb-Umbach R. Uncertainty Decoding in Automatic Speech Recognition. <i>2008
    ITG Conference on Voice Communication (SprachKommunikation)</i>. 2008:1-7.
  apa: Haeb-Umbach, R. (2008). Uncertainty Decoding in Automatic Speech Recognition.
    <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>, 1–7.
  bibtex: '@article{Haeb-Umbach_2008, title={Uncertainty Decoding in Automatic Speech
    Recognition}, journal={2008 ITG Conference on Voice Communication (SprachKommunikation)},
    author={Haeb-Umbach, Reinhold}, year={2008}, pages={1–7} }'
  chicago: Haeb-Umbach, Reinhold. “Uncertainty Decoding in Automatic Speech Recognition.”
    <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>, 2008,
    1–7.
  ieee: R. Haeb-Umbach, “Uncertainty Decoding in Automatic Speech Recognition,” <i>2008
    ITG Conference on Voice Communication (SprachKommunikation)</i>, pp. 1–7, 2008.
  mla: Haeb-Umbach, Reinhold. “Uncertainty Decoding in Automatic Speech Recognition.”
    <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>, 2008,
    pp. 1–7.
  short: R. Haeb-Umbach, 2008 ITG Conference on Voice Communication (SprachKommunikation)
    (2008) 1–7.
date_created: 2019-07-12T05:28:02Z
date_updated: 2022-01-06T06:51:08Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2008/Ha08.pdf
oa: '1'
page: 1-7
publication: 2008 ITG Conference on Voice Communication (SprachKommunikation)
status: public
title: Uncertainty Decoding in Automatic Speech Recognition
type: journal_article
user_id: '44006'
year: '2008'
...
---
_id: '11789'
abstract:
- lang: eng
  text: In distributed and network speech recognition the actual recognition task
    is not carried out on the user{\rq}s terminal but rather on a remote server in
    the network. While there are good reasons for doing so, a disadvantage of this
    client-server architecture is clearly that the communication medium may introduce
    errors, which then impairs speech recognition accuracy. Even sophisticated channel
    coding cannot completely prevent the occurrence of residual bit errors in the
    case of temporarily adverse channel conditions, and in packet-oriented transmission
    packets of data may arrive too late for the given real-time constraints and have
    to be declared lost. The goal of error concealment is to reduce the detrimental
    effect that such errors may induce on the recipient of the transmitted speech
    signal by exploiting residual redundancy in the bit stream at the source coder
    output. In classical speech transmission a human is the recipient, and erroneous
    data are reconstructed so as to reduce the subjectively annoying effect of corrupted
    bits or lost packets. Here, however, a statistical classifier is at the receiving
    end, which can benefit from knowledge about the quality of the reconstruction.
    In this book chapter we show how the classical Bayesian decision rule needs to
    be modified to account for uncertain features, and illustrate how the required
    feature posterior density can be estimated in the case of distributed speech recognition.
    Some other techniques for error concealment can be related to this approach. Experimental
    results are given for both a small and a medium vocabulary recognition task and
    both for a channel exhibiting bit errors and a packet erasure channel.
author:
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
- first_name: Valentin
  full_name: Ion, Valentin
  last_name: Ion
citation:
  ama: 'Haeb-Umbach R, Ion V. Error Concealement. In: Lindenberg B, Tan Z-H, eds.
    <i>Automatic Speech Recognition on Mobile Devices and over Communication Networks</i>.
    Vol Advances in Computer Vision and Pattern Recognition. Advances in Pattern Recognition.
    Springer; 2008:187-210.'
  apa: Haeb-Umbach, R., &#38; Ion, V. (2008). Error Concealement. In B. Lindenberg
    &#38; Z.-H. Tan (Eds.), <i>Automatic Speech Recognition on Mobile Devices and
    over Communication Networks</i> (Vol. Advances in Computer Vision and Pattern
    Recognition, pp. 187–210). Springer.
  bibtex: '@inbook{Haeb-Umbach_Ion_2008, series={Advances in Pattern Recognition},
    title={Error Concealement}, volume={Advances in Computer Vision and Pattern Recognition},
    booktitle={Automatic Speech Recognition on Mobile Devices and over Communication
    Networks}, publisher={Springer}, author={Haeb-Umbach, Reinhold and Ion, Valentin},
    editor={Lindenberg, Borge and Tan, Zheng-HuaEditors}, year={2008}, pages={187–210},
    collection={Advances in Pattern Recognition} }'
  chicago: Haeb-Umbach, Reinhold, and Valentin Ion. “Error Concealement.” In <i>Automatic
    Speech Recognition on Mobile Devices and over Communication Networks</i>, edited
    by Borge Lindenberg and Zheng-Hua Tan, Advances in Computer Vision and Pattern
    Recognition:187–210. Advances in Pattern Recognition. Springer, 2008.
  ieee: R. Haeb-Umbach and V. Ion, “Error Concealement,” in <i>Automatic Speech Recognition
    on Mobile Devices and over Communication Networks</i>, vol. Advances in Computer
    Vision and Pattern Recognition, B. Lindenberg and Z.-H. Tan, Eds. Springer, 2008,
    pp. 187–210.
  mla: Haeb-Umbach, Reinhold, and Valentin Ion. “Error Concealement.” <i>Automatic
    Speech Recognition on Mobile Devices and over Communication Networks</i>, edited
    by Borge Lindenberg and Zheng-Hua Tan, vol. Advances in Computer Vision and Pattern
    Recognition, Springer, 2008, pp. 187–210.
  short: 'R. Haeb-Umbach, V. Ion, in: B. Lindenberg, Z.-H. Tan (Eds.), Automatic Speech
    Recognition on Mobile Devices and over Communication Networks, Springer, 2008,
    pp. 187–210.'
date_created: 2019-07-12T05:28:17Z
date_updated: 2022-01-06T06:51:08Z
department:
- _id: '54'
editor:
- first_name: Borge
  full_name: Lindenberg, Borge
  last_name: Lindenberg
- first_name: Zheng-Hua
  full_name: Tan, Zheng-Hua
  last_name: Tan
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2008/HaIo08.pdf
oa: '1'
page: 187-210
publication: Automatic Speech Recognition on Mobile Devices and over Communication
  Networks
publisher: Springer
series_title: Advances in Pattern Recognition
status: public
title: Error Concealement
type: book_chapter
user_id: '44006'
volume: Advances in Computer Vision and Pattern Recognition
year: '2008'
...
---
_id: '11820'
abstract:
- lang: eng
  text: In this paper, we derive an uncertainty decoding rule for automatic speech
    recognition (ASR), which accounts for both corrupted observations and inter-frame
    correlation. The conditional independence assumption, prevalent in hidden Markov
    model-based ASR, is relaxed to obtain a clean speech posterior that is conditioned
    on the complete observed feature vector sequence. This is a more informative posterior
    than one conditioned only on the current observation. The novel decoding is used
    to obtain a transmission-error robust remote ASR system, where the speech capturing
    unit is connected to the decoder via an error-prone communication network. We
    show how the clean speech posterior can be computed for communication links being
    characterized by either bit errors or packet loss. Recognition results are presented
    for both distributed and network speech recognition, where in the latter case
    common voice-over-IP codecs are employed.
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 Novel Uncertainty Decoding Rule With Applications to
    Transmission Error Robust Speech Recognition. <i>IEEE Transactions on Audio, Speech,
    and Language Processing</i>. 2008;16(5):1047-1060. doi:<a href="https://doi.org/10.1109/TASL.2008.925879">10.1109/TASL.2008.925879</a>
  apa: Ion, V., &#38; Haeb-Umbach, R. (2008). A Novel Uncertainty Decoding Rule With
    Applications to Transmission Error Robust Speech Recognition. <i>IEEE Transactions
    on Audio, Speech, and Language Processing</i>, <i>16</i>(5), 1047–1060. <a href="https://doi.org/10.1109/TASL.2008.925879">https://doi.org/10.1109/TASL.2008.925879</a>
  bibtex: '@article{Ion_Haeb-Umbach_2008, title={A Novel Uncertainty Decoding Rule
    With Applications to Transmission Error Robust Speech Recognition}, volume={16},
    DOI={<a href="https://doi.org/10.1109/TASL.2008.925879">10.1109/TASL.2008.925879</a>},
    number={5}, journal={IEEE Transactions on Audio, Speech, and Language Processing},
    author={Ion, Valentin and Haeb-Umbach, Reinhold}, year={2008}, pages={1047–1060}
    }'
  chicago: 'Ion, Valentin, and Reinhold Haeb-Umbach. “A Novel Uncertainty Decoding
    Rule With Applications to Transmission Error Robust Speech Recognition.” <i>IEEE
    Transactions on Audio, Speech, and Language Processing</i> 16, no. 5 (2008): 1047–60.
    <a href="https://doi.org/10.1109/TASL.2008.925879">https://doi.org/10.1109/TASL.2008.925879</a>.'
  ieee: V. Ion and R. Haeb-Umbach, “A Novel Uncertainty Decoding Rule With Applications
    to Transmission Error Robust Speech Recognition,” <i>IEEE Transactions on Audio,
    Speech, and Language Processing</i>, vol. 16, no. 5, pp. 1047–1060, 2008.
  mla: Ion, Valentin, and Reinhold Haeb-Umbach. “A Novel Uncertainty Decoding Rule
    With Applications to Transmission Error Robust Speech Recognition.” <i>IEEE Transactions
    on Audio, Speech, and Language Processing</i>, vol. 16, no. 5, 2008, pp. 1047–60,
    doi:<a href="https://doi.org/10.1109/TASL.2008.925879">10.1109/TASL.2008.925879</a>.
  short: V. Ion, R. Haeb-Umbach, IEEE Transactions on Audio, Speech, and Language
    Processing 16 (2008) 1047–1060.
date_created: 2019-07-12T05:28:53Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
doi: 10.1109/TASL.2008.925879
intvolume: '        16'
issue: '5'
keyword:
- automatic speech recognition
- bit errors
- codecs
- communication links
- corrupted observations
- decoding
- distributed speech recognition
- error-prone communication network
- feature vector sequence
- hidden Markov model-based ASR
- hidden Markov models
- inter-frame correlation
- Internet telephony
- network speech recognition
- packet loss
- speech posterior
- speech recognition
- transmission error robust speech recognition
- uncertainty decoding
- voice-over-IP codecs
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2008/IoHa08-1.pdf
oa: '1'
page: 1047-1060
publication: IEEE Transactions on Audio, Speech, and Language Processing
status: public
title: A Novel Uncertainty Decoding Rule With Applications to Transmission Error Robust
  Speech Recognition
type: journal_article
user_id: '44006'
volume: 16
year: '2008'
...
---
_id: '11821'
abstract:
- lang: eng
  text: This paper addresses the robustness of automatic speech recognition to environmental
    noise. In order to account for reliability of the clean feature estimate we employ
    the feature posterior density conditioned on observed noisy features to perform
    uncertainty decoding. We investigate two approaches to estimate the posterior
    using a discrete feature space, first conditioning only on the current observation,
    and second on the whole feature sequence of an utterance. Experiments with Aurora
    2 showed that the latter provides slightly better performance, as it allows for
    exploiting the temporal correlations between consecutive features.
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. Investigations into Uncertainty Decoding Employing a
    Discrete Feature Space for Noise Robust Automatic Speech Recognition. <i>2008
    ITG Conference on Voice Communication (SprachKommunikation)</i>. 2008:1-4.
  apa: Ion, V., &#38; Haeb-Umbach, R. (2008). Investigations into Uncertainty Decoding
    Employing a Discrete Feature Space for Noise Robust Automatic Speech Recognition.
    <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>, 1–4.
  bibtex: '@article{Ion_Haeb-Umbach_2008, title={Investigations into Uncertainty Decoding
    Employing a Discrete Feature Space for Noise Robust Automatic Speech Recognition},
    journal={2008 ITG Conference on Voice Communication (SprachKommunikation)}, author={Ion,
    Valentin and Haeb-Umbach, Reinhold}, year={2008}, pages={1–4} }'
  chicago: Ion, Valentin, and Reinhold Haeb-Umbach. “Investigations into Uncertainty
    Decoding Employing a Discrete Feature Space for Noise Robust Automatic Speech
    Recognition.” <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>,
    2008, 1–4.
  ieee: V. Ion and R. Haeb-Umbach, “Investigations into Uncertainty Decoding Employing
    a Discrete Feature Space for Noise Robust Automatic Speech Recognition,” <i>2008
    ITG Conference on Voice Communication (SprachKommunikation)</i>, pp. 1–4, 2008.
  mla: Ion, Valentin, and Reinhold Haeb-Umbach. “Investigations into Uncertainty Decoding
    Employing a Discrete Feature Space for Noise Robust Automatic Speech Recognition.”
    <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>, 2008,
    pp. 1–4.
  short: V. Ion, R. Haeb-Umbach, 2008 ITG Conference on Voice Communication (SprachKommunikation)
    (2008) 1–4.
date_created: 2019-07-12T05:28:54Z
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/2008/IoHa08-2.pdf
oa: '1'
page: 1-4
publication: 2008 ITG Conference on Voice Communication (SprachKommunikation)
status: public
title: Investigations into Uncertainty Decoding Employing a Discrete Feature Space
  for Noise Robust Automatic Speech Recognition
type: journal_article
user_id: '44006'
year: '2008'
...
---
_id: '11851'
abstract:
- lang: eng
  text: 'In diesem Beitrag werden zwei neuartige akustische Strahlformungsalgorithmen
    fuer einen Einsatz im KFZ diskutiert. In beiden Verfahren ist fuer jede Frequenzkomponente
    der Eigenvektor zum groessten Eigenwert eines verallgemeinerten Eigenwertproblems
    zu bestimmen: bei der einen Varianate stellen die Filterkoeffizienten gerade obigen
    Eigenvektor dar. Sprachverzerrungen werden mit einem einkanaligen Nachfilter kompensiert.
    In der anderen Variante, welche die Struktur eines ''''Generalized Sidelobe Cancellers''''
    (GSC) hat, basiert die Blockiermatrix auf obiger Eigenvektorzerlegung.'
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. Blinde Akustische Strahlformung fuer
    Anwendungen im KFZ. In: <i>34. Deutsche Jahrestagung Fuer Akustik (DAGA 2008)</i>.
    ; 2008.'
  apa: Krueger, A., Warsitz, E., &#38; Haeb-Umbach, R. (2008). Blinde Akustische Strahlformung
    fuer Anwendungen im KFZ. In <i>34. Deutsche Jahrestagung fuer Akustik (DAGA 2008)</i>.
  bibtex: '@inproceedings{Krueger_Warsitz_Haeb-Umbach_2008, title={Blinde Akustische
    Strahlformung fuer Anwendungen im KFZ}, booktitle={34. Deutsche Jahrestagung fuer
    Akustik (DAGA 2008)}, author={Krueger, Alexander and Warsitz, Ernst and Haeb-Umbach,
    Reinhold}, year={2008} }'
  chicago: Krueger, Alexander, Ernst Warsitz, and Reinhold Haeb-Umbach. “Blinde Akustische
    Strahlformung Fuer Anwendungen Im KFZ.” In <i>34. Deutsche Jahrestagung Fuer Akustik
    (DAGA 2008)</i>, 2008.
  ieee: A. Krueger, E. Warsitz, and R. Haeb-Umbach, “Blinde Akustische Strahlformung
    fuer Anwendungen im KFZ,” in <i>34. Deutsche Jahrestagung fuer Akustik (DAGA 2008)</i>,
    2008.
  mla: Krueger, Alexander, et al. “Blinde Akustische Strahlformung Fuer Anwendungen
    Im KFZ.” <i>34. Deutsche Jahrestagung Fuer Akustik (DAGA 2008)</i>, 2008.
  short: 'A. Krueger, E. Warsitz, R. Haeb-Umbach, in: 34. Deutsche Jahrestagung Fuer
    Akustik (DAGA 2008), 2008.'
date_created: 2019-07-12T05:29:29Z
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/2008/KrWaHa08.pdf
oa: '1'
publication: 34. Deutsche Jahrestagung fuer Akustik (DAGA 2008)
status: public
title: Blinde Akustische Strahlformung fuer Anwendungen im KFZ
type: conference
user_id: '44006'
year: '2008'
...
---
_id: '11914'
abstract:
- lang: eng
  text: This paper considers the convolutive blind source separation of speech sources
    in the presence of spatially correlated noise. We introduce a method for estimating
    the scaled mixing matrix from the sources to the microphones even if coherent
    noise is present. This is achieved by combining time-frequency sparseness with
    the generalized eigenvalue decomposition of the power spectral density matrix
    (PSD) of the noisy speech and noise-only microphone signals. Separation is performed
    by spatial filtering with coefficients constructed by Gram-Schmidt orthogonalization
    which places spatial nulls at the interferer{^A}?`s direction. Experimental results
    show that our approach is capable of separating 2 sources in a reverberant environment
    (RT60=0ms..500ms) degraded by significant directional noise.
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 in Presence of Correlated
    Noise with Generalized Eigenvector Beamforming. <i>2008 ITG Conference on Voice
    Communication (SprachKommunikation)</i>. 2008:1-4.
  apa: Tran Vu, D. H., &#38; Haeb-Umbach, R. (2008). Blind Speech Separation in Presence
    of Correlated Noise with Generalized Eigenvector Beamforming. <i>2008 ITG Conference
    on Voice Communication (SprachKommunikation)</i>, 1–4.
  bibtex: '@article{Tran Vu_Haeb-Umbach_2008, title={Blind Speech Separation in Presence
    of Correlated Noise with Generalized Eigenvector Beamforming}, journal={2008 ITG
    Conference on Voice Communication (SprachKommunikation)}, author={Tran Vu, Dang
    Hai and Haeb-Umbach, Reinhold}, year={2008}, pages={1–4} }'
  chicago: Tran Vu, Dang Hai, and Reinhold Haeb-Umbach. “Blind Speech Separation in
    Presence of Correlated Noise with Generalized Eigenvector Beamforming.” <i>2008
    ITG Conference on Voice Communication (SprachKommunikation)</i>, 2008, 1–4.
  ieee: D. H. Tran Vu and R. Haeb-Umbach, “Blind Speech Separation in Presence of
    Correlated Noise with Generalized Eigenvector Beamforming,” <i>2008 ITG Conference
    on Voice Communication (SprachKommunikation)</i>, pp. 1–4, 2008.
  mla: Tran Vu, Dang Hai, and Reinhold Haeb-Umbach. “Blind Speech Separation in Presence
    of Correlated Noise with Generalized Eigenvector Beamforming.” <i>2008 ITG Conference
    on Voice Communication (SprachKommunikation)</i>, 2008, pp. 1–4.
  short: D.H. Tran Vu, R. Haeb-Umbach, 2008 ITG Conference on Voice Communication
    (SprachKommunikation) (2008) 1–4.
date_created: 2019-07-12T05:30:42Z
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/2008/DaHa08.pdf
oa: '1'
page: 1-4
publication: 2008 ITG Conference on Voice Communication (SprachKommunikation)
status: public
title: Blind Speech Separation in Presence of Correlated Noise with Generalized Eigenvector
  Beamforming
type: journal_article
user_id: '44006'
year: '2008'
...
---
_id: '11915'
abstract:
- lang: eng
  text: This Paper deals with a new Technique for multi-channel separation of speech
    signals from convolutive mixtures under coherent noise. We demonstrate how the
    scaled transfer functions from the sources to the microphones can be estimated
    even in the presence of stationary coherent noise. The key to this are generalized
    eigenvalue decompositions of the power spectral density (PSD) matrices of the
    noisy speech and noise-only microphone signals with a controlled estimation of
    these matrices exploiting time-frequency sparseness of the speech sources. Separation
    is further improved by subsequent Gram-Schmidt orthogonalization which places
    spatial nulls at the interferers{\rq} directions, while noise reduction is improved
    by employing a novel blocking matrix and adaptive interference canceller in a
    Generalized Sidelobe Canceller (GSC)-like structure. We report promising experimental
    results for 2 speech sources with significant coherent noise in reverberant environments
    (RT60=0oms..500ms).
author:
- first_name: Dang Hai
  full_name: Tran Vu, Dang Hai
  last_name: Tran Vu
- 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: 'Tran Vu DH, Krueger A, Haeb-Umbach R. Generalized Eigenvector Blind Speech
    Separation Under Coherent Noise In A GSC Configuration. In: <i>International Workshop
    on Acoustic Echo and Noise Control (IWAENC 2008)</i>. ; 2008.'
  apa: Tran Vu, D. H., Krueger, A., &#38; Haeb-Umbach, R. (2008). Generalized Eigenvector
    Blind Speech Separation Under Coherent Noise In A GSC Configuration. In <i>International
    Workshop on Acoustic Echo and Noise Control (IWAENC 2008)</i>.
  bibtex: '@inproceedings{Tran Vu_Krueger_Haeb-Umbach_2008, title={Generalized Eigenvector
    Blind Speech Separation Under Coherent Noise In A GSC Configuration}, booktitle={International
    Workshop on Acoustic Echo and Noise Control (IWAENC 2008)}, author={Tran Vu, Dang
    Hai and Krueger, Alexander and Haeb-Umbach, Reinhold}, year={2008} }'
  chicago: Tran Vu, Dang Hai, Alexander Krueger, and Reinhold Haeb-Umbach. “Generalized
    Eigenvector Blind Speech Separation Under Coherent Noise In A GSC Configuration.”
    In <i>International Workshop on Acoustic Echo and Noise Control (IWAENC 2008)</i>,
    2008.
  ieee: D. H. Tran Vu, A. Krueger, and R. Haeb-Umbach, “Generalized Eigenvector Blind
    Speech Separation Under Coherent Noise In A GSC Configuration,” in <i>International
    Workshop on Acoustic Echo and Noise Control (IWAENC 2008)</i>, 2008.
  mla: Tran Vu, Dang Hai, et al. “Generalized Eigenvector Blind Speech Separation
    Under Coherent Noise In A GSC Configuration.” <i>International Workshop on Acoustic
    Echo and Noise Control (IWAENC 2008)</i>, 2008.
  short: 'D.H. Tran Vu, A. Krueger, R. Haeb-Umbach, in: International Workshop on
    Acoustic Echo and Noise Control (IWAENC 2008), 2008.'
date_created: 2019-07-12T05:30:43Z
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/2008/DaKrHa08.pdf
oa: '1'
publication: International Workshop on Acoustic Echo and Noise Control (IWAENC 2008)
status: public
title: Generalized Eigenvector Blind Speech Separation Under Coherent Noise In A GSC
  Configuration
type: conference
user_id: '44006'
year: '2008'
...
---
_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: '11939'
abstract:
- lang: eng
  text: In this paper a switching linear dynamical model (SLDM) approach for speech
    feature enhancement is improved by employing more accurate models for the dynamics
    of speech and noise. The model of the clean speech feature trajectory is improved
    by augmenting the state vector to capture information derived from the delta features.
    Further a hidden noise state variable is introduced to obtain a more elaborated
    model for the noise dynamics. Approximate Bayesian inference in the SLDM is carried
    out by a bank of extended Kalman filters, whose outputs are combined according
    to the a posteriori probability of the individual state models. Experimental results
    on the AURORA2 database show improved recognition accuracy.
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. Modeling the dynamics of speech and noise for speech
    feature enhancement in ASR. In: <i>IEEE International Conference on Acoustics,
    Speech and Signal Processing (ICASSP 2008)</i>. ; 2008:4409-4412. doi:<a href="https://doi.org/10.1109/ICASSP.2008.4518633">10.1109/ICASSP.2008.4518633</a>'
  apa: Windmann, S., &#38; Haeb-Umbach, R. (2008). Modeling the dynamics of speech
    and noise for speech feature enhancement in ASR. In <i>IEEE International Conference
    on Acoustics, Speech and Signal Processing (ICASSP 2008)</i> (pp. 4409–4412).
    <a href="https://doi.org/10.1109/ICASSP.2008.4518633">https://doi.org/10.1109/ICASSP.2008.4518633</a>
  bibtex: '@inproceedings{Windmann_Haeb-Umbach_2008, title={Modeling the dynamics
    of speech and noise for speech feature enhancement in ASR}, DOI={<a href="https://doi.org/10.1109/ICASSP.2008.4518633">10.1109/ICASSP.2008.4518633</a>},
    booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
    (ICASSP 2008)}, author={Windmann, Stefan and Haeb-Umbach, Reinhold}, year={2008},
    pages={4409–4412} }'
  chicago: Windmann, Stefan, and Reinhold Haeb-Umbach. “Modeling the Dynamics of Speech
    and Noise for Speech Feature Enhancement in ASR.” In <i>IEEE International Conference
    on Acoustics, Speech and Signal Processing (ICASSP 2008)</i>, 4409–12, 2008. <a
    href="https://doi.org/10.1109/ICASSP.2008.4518633">https://doi.org/10.1109/ICASSP.2008.4518633</a>.
  ieee: S. Windmann and R. Haeb-Umbach, “Modeling the dynamics of speech and noise
    for speech feature enhancement in ASR,” in <i>IEEE International Conference on
    Acoustics, Speech and Signal Processing (ICASSP 2008)</i>, 2008, pp. 4409–4412.
  mla: Windmann, Stefan, and Reinhold Haeb-Umbach. “Modeling the Dynamics of Speech
    and Noise for Speech Feature Enhancement in ASR.” <i>IEEE International Conference
    on Acoustics, Speech and Signal Processing (ICASSP 2008)</i>, 2008, pp. 4409–12,
    doi:<a href="https://doi.org/10.1109/ICASSP.2008.4518633">10.1109/ICASSP.2008.4518633</a>.
  short: 'S. Windmann, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
    Speech and Signal Processing (ICASSP 2008), 2008, pp. 4409–4412.'
date_created: 2019-07-12T05:31:11Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2008.4518633
keyword:
- a posteriori probability
- AURORA2 database
- Bayesian inference
- Bayes methods
- channel bank filters
- extended Kalman filter banks
- hidden noise state variable
- Kalman filters
- noise dynamics
- speech enhancement
- speech feature enhancement
- speech feature trajectory
- switching linear dynamical model approach
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2008/WiHa08-1.pdf
oa: '1'
page: 4409-4412
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
  (ICASSP 2008)
status: public
title: Modeling the dynamics of speech and noise for speech feature enhancement in
  ASR
type: conference
user_id: '44006'
year: '2008'
...
---
_id: '11940'
abstract:
- lang: eng
  text: 'In this paper, the noise estimation for model-based speech feature enhancement
    in automatic speech recognition (ASR) is investigated. Beside a stationary noise
    prior, three linear state space models for the (cepstral) noise process are considered.
    We have derived novel EM algorithms for the estimation of the noise model parameters:
    A blockwise EM algorithm is applied on noise-only input data. It is supposed to
    be used during the offline training mode of the recognizer. Further a sequential
    online EM algorithm is employed to adapt the observation variance in recognition
    mode which works as well under the asumption of a stationary noise prior and a
    linear state model for the noise. Experiments on the AURORA4 database lead to
    improved recognition results with the new state model compared to the assumption
    of stationary noise.'
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. A novel approach to noise estimation in model-based
    speech feature enhancement. <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>.
    2008:1-4.
  apa: Windmann, S., &#38; Haeb-Umbach, R. (2008). A novel approach to noise estimation
    in model-based speech feature enhancement. <i>2008 ITG Conference on Voice Communication
    (SprachKommunikation)</i>, 1–4.
  bibtex: '@article{Windmann_Haeb-Umbach_2008, title={A novel approach to noise estimation
    in model-based speech feature enhancement}, journal={2008 ITG Conference on Voice
    Communication (SprachKommunikation)}, author={Windmann, Stefan and Haeb-Umbach,
    Reinhold}, year={2008}, pages={1–4} }'
  chicago: Windmann, Stefan, and Reinhold Haeb-Umbach. “A Novel Approach to Noise
    Estimation in Model-Based Speech Feature Enhancement.” <i>2008 ITG Conference
    on Voice Communication (SprachKommunikation)</i>, 2008, 1–4.
  ieee: S. Windmann and R. Haeb-Umbach, “A novel approach to noise estimation in model-based
    speech feature enhancement,” <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>,
    pp. 1–4, 2008.
  mla: Windmann, Stefan, and Reinhold Haeb-Umbach. “A Novel Approach to Noise Estimation
    in Model-Based Speech Feature Enhancement.” <i>2008 ITG Conference on Voice Communication
    (SprachKommunikation)</i>, 2008, pp. 1–4.
  short: S. Windmann, R. Haeb-Umbach, 2008 ITG Conference on Voice Communication (SprachKommunikation)
    (2008) 1–4.
date_created: 2019-07-12T05:31:12Z
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/2008/WiHa08-2.pdf
oa: '1'
page: 1-4
publication: 2008 ITG Conference on Voice Communication (SprachKommunikation)
status: public
title: A novel approach to noise estimation in model-based speech feature enhancement
type: journal_article
user_id: '44006'
year: '2008'
...
---
_id: '11944'
abstract:
- lang: eng
  text: In this paper, a novel segmental Hidden Markov Model (HMM) is proposed. The
    model is based on a modified emission density where additional statistical dependencies
    between subsequent frames of the speech signal are considered. In the following
    we derive an effective search strategy for the modified statistical model. Further
    an approach to parameter reduction is introduced. Experiments were carried out
    on the AURORA2 database where consistent im
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
- first_name: Volker
  full_name: Leutnant, Volker
  last_name: Leutnant
citation:
  ama: Windmann S, Haeb-Umbach R, Leutnant V. A segmental HMM based on a modified
    emission probability. <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>.
    2008:1-4.
  apa: Windmann, S., Haeb-Umbach, R., &#38; Leutnant, V. (2008). A segmental HMM based
    on a modified emission probability. <i>2008 ITG Conference on Voice Communication
    (SprachKommunikation)</i>, 1–4.
  bibtex: '@article{Windmann_Haeb-Umbach_Leutnant_2008, title={A segmental HMM based
    on a modified emission probability}, journal={2008 ITG Conference on Voice Communication
    (SprachKommunikation)}, author={Windmann, Stefan and Haeb-Umbach, Reinhold and
    Leutnant, Volker}, year={2008}, pages={1–4} }'
  chicago: Windmann, Stefan, Reinhold Haeb-Umbach, and Volker Leutnant. “A Segmental
    HMM Based on a Modified Emission Probability.” <i>2008 ITG Conference on Voice
    Communication (SprachKommunikation)</i>, 2008, 1–4.
  ieee: S. Windmann, R. Haeb-Umbach, and V. Leutnant, “A segmental HMM based on a
    modified emission probability,” <i>2008 ITG Conference on Voice Communication
    (SprachKommunikation)</i>, pp. 1–4, 2008.
  mla: Windmann, Stefan, et al. “A Segmental HMM Based on a Modified Emission Probability.”
    <i>2008 ITG Conference on Voice Communication (SprachKommunikation)</i>, 2008,
    pp. 1–4.
  short: S. Windmann, R. Haeb-Umbach, V. Leutnant, 2008 ITG Conference on Voice Communication
    (SprachKommunikation) (2008) 1–4.
date_created: 2019-07-12T05:31:16Z
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/2008/WiHaLe08.pdf
oa: '1'
page: 1-4
publication: 2008 ITG Conference on Voice Communication (SprachKommunikation)
status: public
title: A segmental HMM based on a modified emission probability
type: journal_article
user_id: '44006'
year: '2008'
...
---
_id: '11720'
author:
- first_name: Maik
  full_name: Bevermeier, Maik
  last_name: Bevermeier
- first_name: Tobias
  full_name: Ebel, Tobias
  last_name: Ebel
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Bevermeier M, Ebel T, Haeb-Umbach R. Channel Estimation by Exploiting Sublayer
    Information in OFDM Systems. In: <i>Multi-Carrier Spread Spectrum 2007</i>. ;
    2007.'
  apa: Bevermeier, M., Ebel, T., &#38; Haeb-Umbach, R. (2007). Channel Estimation
    by Exploiting Sublayer Information in OFDM Systems. In <i>Multi-Carrier Spread
    Spectrum 2007</i>.
  bibtex: '@inproceedings{Bevermeier_Ebel_Haeb-Umbach_2007, title={Channel Estimation
    by Exploiting Sublayer Information in OFDM Systems}, booktitle={Multi-Carrier
    Spread Spectrum 2007}, author={Bevermeier, Maik and Ebel, Tobias and Haeb-Umbach,
    Reinhold}, year={2007} }'
  chicago: Bevermeier, Maik, Tobias Ebel, and Reinhold Haeb-Umbach. “Channel Estimation
    by Exploiting Sublayer Information in OFDM Systems.” In <i>Multi-Carrier Spread
    Spectrum 2007</i>, 2007.
  ieee: M. Bevermeier, T. Ebel, and R. Haeb-Umbach, “Channel Estimation by Exploiting
    Sublayer Information in OFDM Systems,” in <i>Multi-Carrier Spread Spectrum 2007</i>,
    2007.
  mla: Bevermeier, Maik, et al. “Channel Estimation by Exploiting Sublayer Information
    in OFDM Systems.” <i>Multi-Carrier Spread Spectrum 2007</i>, 2007.
  short: 'M. Bevermeier, T. Ebel, R. Haeb-Umbach, in: Multi-Carrier Spread Spectrum
    2007, 2007.'
date_created: 2019-07-12T05:26:57Z
date_updated: 2022-01-06T06:51:07Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/BeEbHa07.pdf
oa: '1'
publication: Multi-Carrier Spread Spectrum 2007
status: public
title: Channel Estimation by Exploiting Sublayer Information in OFDM Systems
type: conference
user_id: '44006'
year: '2007'
...
---
_id: '11722'
author:
- first_name: Maik
  full_name: Bevermeier, Maik
  last_name: Bevermeier
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
citation:
  ama: 'Bevermeier M, Haeb-Umbach R. Combined Time and Frequency Domain OFDM Channel
    Estimation. In: <i>Multi-Carrier Spread Spectrum 2007</i>. ; 2007.'
  apa: Bevermeier, M., &#38; Haeb-Umbach, R. (2007). Combined Time and Frequency Domain
    OFDM Channel Estimation. In <i>Multi-Carrier Spread Spectrum 2007</i>.
  bibtex: '@inproceedings{Bevermeier_Haeb-Umbach_2007, title={Combined Time and Frequency
    Domain OFDM Channel Estimation}, booktitle={Multi-Carrier Spread Spectrum 2007},
    author={Bevermeier, Maik and Haeb-Umbach, Reinhold}, year={2007} }'
  chicago: Bevermeier, Maik, and Reinhold Haeb-Umbach. “Combined Time and Frequency
    Domain OFDM Channel Estimation.” In <i>Multi-Carrier Spread Spectrum 2007</i>,
    2007.
  ieee: M. Bevermeier and R. Haeb-Umbach, “Combined Time and Frequency Domain OFDM
    Channel Estimation,” in <i>Multi-Carrier Spread Spectrum 2007</i>, 2007.
  mla: Bevermeier, Maik, and Reinhold Haeb-Umbach. “Combined Time and Frequency Domain
    OFDM Channel Estimation.” <i>Multi-Carrier Spread Spectrum 2007</i>, 2007.
  short: 'M. Bevermeier, R. Haeb-Umbach, in: Multi-Carrier Spread Spectrum 2007, 2007.'
date_created: 2019-07-12T05:27:00Z
date_updated: 2022-01-06T06:51:07Z
department:
- _id: '54'
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/BeEb07.pdf
oa: '1'
publication: Multi-Carrier Spread Spectrum 2007
status: public
title: Combined Time and Frequency Domain OFDM Channel Estimation
type: conference
user_id: '44006'
year: '2007'
...
---
_id: '11785'
abstract:
- lang: eng
  text: 'In this paper we present a novel channel impulse response estimation technique
    for block-oriented OFDM transmission based on combining estimators: the estimates
    provided by a Kalman filter operating in the time domain and a Wiener filter in
    the frequency domain are optimally combined by taking into account their estimated
    error covariances. The resulting estimator turns out to be identical to the MAP
    estimator of correlated jointly Gaussian mean vectors. Different variants of the
    proposed scheme are experimentally investigated in an EEEE 802.11a-like system
    setup. They compare favourably with known approaches from the literature resulting
    in reduced mean square estimation error and bit error rate. Further, robustness
    and complexity issues are discussed'
author:
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
- first_name: Maik
  full_name: Bevermeier, Maik
  last_name: Bevermeier
citation:
  ama: 'Haeb-Umbach R, Bevermeier M. OFDM Channel Estimation Based on Combined Estimation
    in Time and Frequency Domain. In: <i>IEEE International Conference on Acoustics,
    Speech and Signal Processing (ICASSP 2007)</i>. Vol 3. ; 2007:III-277-III-280.
    doi:<a href="https://doi.org/10.1109/ICASSP.2007.366526">10.1109/ICASSP.2007.366526</a>'
  apa: Haeb-Umbach, R., &#38; Bevermeier, M. (2007). OFDM Channel Estimation Based
    on Combined Estimation in Time and Frequency Domain. In <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2007)</i> (Vol.
    3, pp. III-277-III–280). <a href="https://doi.org/10.1109/ICASSP.2007.366526">https://doi.org/10.1109/ICASSP.2007.366526</a>
  bibtex: '@inproceedings{Haeb-Umbach_Bevermeier_2007, title={OFDM Channel Estimation
    Based on Combined Estimation in Time and Frequency Domain}, volume={3}, DOI={<a
    href="https://doi.org/10.1109/ICASSP.2007.366526">10.1109/ICASSP.2007.366526</a>},
    booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
    (ICASSP 2007)}, author={Haeb-Umbach, Reinhold and Bevermeier, Maik}, year={2007},
    pages={III-277-III–280} }'
  chicago: Haeb-Umbach, Reinhold, and Maik Bevermeier. “OFDM Channel Estimation Based
    on Combined Estimation in Time and Frequency Domain.” In <i>IEEE International
    Conference on Acoustics, Speech and Signal Processing (ICASSP 2007)</i>, 3:III-277-III–280,
    2007. <a href="https://doi.org/10.1109/ICASSP.2007.366526">https://doi.org/10.1109/ICASSP.2007.366526</a>.
  ieee: R. Haeb-Umbach and M. Bevermeier, “OFDM Channel Estimation Based on Combined
    Estimation in Time and Frequency Domain,” in <i>IEEE International Conference
    on Acoustics, Speech and Signal Processing (ICASSP 2007)</i>, 2007, vol. 3, pp.
    III-277-III–280.
  mla: Haeb-Umbach, Reinhold, and Maik Bevermeier. “OFDM Channel Estimation Based
    on Combined Estimation in Time and Frequency Domain.” <i>IEEE International Conference
    on Acoustics, Speech and Signal Processing (ICASSP 2007)</i>, vol. 3, 2007, pp.
    III-277-III–280, doi:<a href="https://doi.org/10.1109/ICASSP.2007.366526">10.1109/ICASSP.2007.366526</a>.
  short: 'R. Haeb-Umbach, M. Bevermeier, in: IEEE International Conference on Acoustics,
    Speech and Signal Processing (ICASSP 2007), 2007, pp. III-277-III–280.'
date_created: 2019-07-12T05:28:13Z
date_updated: 2022-01-06T06:51:08Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2007.366526
intvolume: '         3'
keyword:
- bit error rate
- block-oriented OFDM transmission
- channel estimation
- channel impulse response estimation
- combining estimators
- error statistics
- frequency domain estimation
- Gaussian mean vectors
- Gaussian processes
- Kalman filter
- Kalman filters
- MAP estimator
- maximum likelihood estimation
- OFDM channel estimation
- OFDM modulation
- time domain estimation
- time-frequency analysis
- Wiener filter
- Wiener filters
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/HaBe07.pdf
oa: '1'
page: III-277-III-280
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
  (ICASSP 2007)
status: public
title: OFDM Channel Estimation Based on Combined Estimation in Time and Frequency
  Domain
type: conference
user_id: '44006'
volume: 3
year: '2007'
...
---
_id: '11799'
abstract:
- lang: eng
  text: In this paper, we propose a novel similarity measure to be used for localizing
    mobile terminals by comparing measured signal power levels with a database of
    predictions. The proposed measure provides the possibility to incorporate inherent
    information about signal power level measurements requested by the serving base
    station but not reported by the mobile terminal. Increased positioning accuracy
    was observed both in simulations and with real field data
author:
- first_name: Reinhold
  full_name: Haeb-Umbach, Reinhold
  id: '242'
  last_name: Haeb-Umbach
- first_name: Sven
  full_name: Peschke, Sven
  last_name: Peschke
citation:
  ama: Haeb-Umbach R, Peschke S. A Novel Similarity Measure for Positioning Cellular
    Phones by a Comparison With a Database of Signal Power Levels. <i>IEEE Transactions
    on Vehicular Technology</i>. 2007;56(1):368-372. doi:<a href="https://doi.org/10.1109/TVT.2006.889563">10.1109/TVT.2006.889563</a>
  apa: Haeb-Umbach, R., &#38; Peschke, S. (2007). A Novel Similarity Measure for Positioning
    Cellular Phones by a Comparison With a Database of Signal Power Levels. <i>IEEE
    Transactions on Vehicular Technology</i>, <i>56</i>(1), 368–372. <a href="https://doi.org/10.1109/TVT.2006.889563">https://doi.org/10.1109/TVT.2006.889563</a>
  bibtex: '@article{Haeb-Umbach_Peschke_2007, title={A Novel Similarity Measure for
    Positioning Cellular Phones by a Comparison With a Database of Signal Power Levels},
    volume={56}, DOI={<a href="https://doi.org/10.1109/TVT.2006.889563">10.1109/TVT.2006.889563</a>},
    number={1}, journal={IEEE Transactions on Vehicular Technology}, author={Haeb-Umbach,
    Reinhold and Peschke, Sven}, year={2007}, pages={368–372} }'
  chicago: 'Haeb-Umbach, Reinhold, and Sven Peschke. “A Novel Similarity Measure for
    Positioning Cellular Phones by a Comparison With a Database of Signal Power Levels.”
    <i>IEEE Transactions on Vehicular Technology</i> 56, no. 1 (2007): 368–72. <a
    href="https://doi.org/10.1109/TVT.2006.889563">https://doi.org/10.1109/TVT.2006.889563</a>.'
  ieee: R. Haeb-Umbach and S. Peschke, “A Novel Similarity Measure for Positioning
    Cellular Phones by a Comparison With a Database of Signal Power Levels,” <i>IEEE
    Transactions on Vehicular Technology</i>, vol. 56, no. 1, pp. 368–372, 2007.
  mla: Haeb-Umbach, Reinhold, and Sven Peschke. “A Novel Similarity Measure for Positioning
    Cellular Phones by a Comparison With a Database of Signal Power Levels.” <i>IEEE
    Transactions on Vehicular Technology</i>, vol. 56, no. 1, 2007, pp. 368–72, doi:<a
    href="https://doi.org/10.1109/TVT.2006.889563">10.1109/TVT.2006.889563</a>.
  short: R. Haeb-Umbach, S. Peschke, IEEE Transactions on Vehicular Technology 56
    (2007) 368–372.
date_created: 2019-07-12T05:28:29Z
date_updated: 2022-01-06T06:51:08Z
department:
- _id: '54'
doi: 10.1109/TVT.2006.889563
intvolume: '        56'
issue: '1'
keyword:
- cellular phone positioning
- cellular radio
- measured signal power levels
- mobile handsets
- mobility management (mobile radio)
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/HaPe07.pdf
oa: '1'
page: 368-372
publication: IEEE Transactions on Vehicular Technology
status: public
title: A Novel Similarity Measure for Positioning Cellular Phones by a Comparison
  With a Database of Signal Power Levels
type: journal_article
user_id: '44006'
volume: 56
year: '2007'
...
---
_id: '11822'
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. Multi-Resolution Soft Features for Channel-Robust Distributed
    Speech Recognition. In: <i>Interspeech 2007</i>. ; 2007.'
  apa: Ion, V., &#38; Haeb-Umbach, R. (2007). Multi-Resolution Soft Features for Channel-Robust
    Distributed Speech Recognition. In <i>Interspeech 2007</i>.
  bibtex: '@inproceedings{Ion_Haeb-Umbach_2007, title={Multi-Resolution Soft Features
    for Channel-Robust Distributed Speech Recognition}, booktitle={Interspeech 2007},
    author={Ion, Valentin and Haeb-Umbach, Reinhold}, year={2007} }'
  chicago: Ion, Valentin, and Reinhold Haeb-Umbach. “Multi-Resolution Soft Features
    for Channel-Robust Distributed Speech Recognition.” In <i>Interspeech 2007</i>,
    2007.
  ieee: V. Ion and R. Haeb-Umbach, “Multi-Resolution Soft Features for Channel-Robust
    Distributed Speech Recognition,” in <i>Interspeech 2007</i>, 2007.
  mla: Ion, Valentin, and Reinhold Haeb-Umbach. “Multi-Resolution Soft Features for
    Channel-Robust Distributed Speech Recognition.” <i>Interspeech 2007</i>, 2007.
  short: 'V. Ion, R. Haeb-Umbach, in: Interspeech 2007, 2007.'
date_created: 2019-07-12T05:28:55Z
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/2007/IoHa07.pdf
oa: '1'
publication: Interspeech 2007
status: public
title: Multi-Resolution Soft Features for Channel-Robust Distributed Speech Recognition
type: conference
user_id: '44006'
year: '2007'
...
---
_id: '11883'
abstract:
- lang: eng
  text: In this paper, we experimentally evaluate algorithms for velocity estimation
    of a GSM 900 mobile terminal which are based on the analysis of the statistical
    properties of the fast fading process. It is shown how theses statistics can be
    obtained from the training sequences present in downlink transmission bursts without
    establishing an active connection. Realistic simulations of a GSM channel according
    to the COST 207 channel models have been conducted. These models incorporate effects
    like multipath propagation, fading, cochannel interference and additive noise.
    It is shown that velocity estimation by searching for the maximum slope of the
    power density spectrum of the fast fading performs best.
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. Velocity Estimation of Mobile Terminals by Exploiting
    GSM Downlink Signalling. In: <i>4th Workshop on Positioning Navigation and Communication
    (WPNC 2007)</i>. ; 2007:217-222. doi:<a href="https://doi.org/10.1109/WPNC.2007.353637">10.1109/WPNC.2007.353637</a>'
  apa: Peschke, S., &#38; Haeb-Umbach, R. (2007). Velocity Estimation of Mobile Terminals
    by Exploiting GSM Downlink Signalling. In <i>4th Workshop on Positioning Navigation
    and Communication (WPNC 2007)</i> (pp. 217–222). <a href="https://doi.org/10.1109/WPNC.2007.353637">https://doi.org/10.1109/WPNC.2007.353637</a>
  bibtex: '@inproceedings{Peschke_Haeb-Umbach_2007, title={Velocity Estimation of
    Mobile Terminals by Exploiting GSM Downlink Signalling}, DOI={<a href="https://doi.org/10.1109/WPNC.2007.353637">10.1109/WPNC.2007.353637</a>},
    booktitle={4th Workshop on Positioning Navigation and Communication (WPNC 2007)},
    author={Peschke, Sven and Haeb-Umbach, Reinhold}, year={2007}, pages={217–222}
    }'
  chicago: Peschke, Sven, and Reinhold Haeb-Umbach. “Velocity Estimation of Mobile
    Terminals by Exploiting GSM Downlink Signalling.” In <i>4th Workshop on Positioning
    Navigation and Communication (WPNC 2007)</i>, 217–22, 2007. <a href="https://doi.org/10.1109/WPNC.2007.353637">https://doi.org/10.1109/WPNC.2007.353637</a>.
  ieee: S. Peschke and R. Haeb-Umbach, “Velocity Estimation of Mobile Terminals by
    Exploiting GSM Downlink Signalling,” in <i>4th Workshop on Positioning Navigation
    and Communication (WPNC 2007)</i>, 2007, pp. 217–222.
  mla: Peschke, Sven, and Reinhold Haeb-Umbach. “Velocity Estimation of Mobile Terminals
    by Exploiting GSM Downlink Signalling.” <i>4th Workshop on Positioning Navigation
    and Communication (WPNC 2007)</i>, 2007, pp. 217–22, doi:<a href="https://doi.org/10.1109/WPNC.2007.353637">10.1109/WPNC.2007.353637</a>.
  short: 'S. Peschke, R. Haeb-Umbach, in: 4th Workshop on Positioning Navigation and
    Communication (WPNC 2007), 2007, pp. 217–222.'
date_created: 2019-07-12T05:30:06Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
doi: 10.1109/WPNC.2007.353637
keyword:
- additive noise
- cellular radio
- channel estimation
- cochannel interference
- COST 207 channel models
- downlink transmission bursts
- fading channels
- fading process
- GSM downlink signalling
- mobile terminals
- multipath channels
- multipath propagation
- power density spectrum
- statistical analysis
- statistical properties
- telecommunication links
- telecommunication terminals
- velocity estimation
language:
- iso: eng
main_file_link:
- open_access: '1'
  url: https://groups.uni-paderborn.de/nt/pubs/2007/PeHa07.pdf
oa: '1'
page: 217-222
publication: 4th Workshop on Positioning Navigation and Communication (WPNC 2007)
status: public
title: Velocity Estimation of Mobile Terminals by Exploiting GSM Downlink Signalling
type: conference
user_id: '44006'
year: '2007'
...
---
_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'
...