---
_id: '33398'
author:
- first_name: E.
full_name: Mulder, E.
last_name: Mulder
- first_name: G.
full_name: Clément, G.
last_name: Clément
- first_name: D.
full_name: Linnarsson, D.
last_name: Linnarsson
- first_name: W. H.
full_name: Paloski, W. H.
last_name: Paloski
- first_name: F. P.
full_name: Wuyts, F. P.
last_name: Wuyts
- first_name: J.
full_name: Zange, J.
last_name: Zange
- first_name: P.
full_name: Frings-Meuthen, P.
last_name: Frings-Meuthen
- first_name: B.
full_name: Johannes, B.
last_name: Johannes
- first_name: V.
full_name: Shushakov, V.
last_name: Shushakov
- first_name: M.
full_name: Grunewald, M.
last_name: Grunewald
- first_name: N.
full_name: Maassen, N.
last_name: Maassen
- first_name: Judith
full_name: Bühlmeier, Judith
id: '89838'
last_name: Bühlmeier
- first_name: J.
full_name: Rittweger, J.
last_name: Rittweger
citation:
ama: Mulder E, Clément G, Linnarsson D, et al. Musculoskeletal effects of 5 days
of bed rest with and without locomotion replacement training. European Journal
of Applied Physiology. 2014;115(4):727-738. doi:10.1007/s00421-014-3045-0
apa: Mulder, E., Clément, G., Linnarsson, D., Paloski, W. H., Wuyts, F. P., Zange,
J., Frings-Meuthen, P., Johannes, B., Shushakov, V., Grunewald, M., Maassen, N.,
Bühlmeier, J., & Rittweger, J. (2014). Musculoskeletal effects of 5 days of
bed rest with and without locomotion replacement training. European Journal
of Applied Physiology, 115(4), 727–738. https://doi.org/10.1007/s00421-014-3045-0
bibtex: '@article{Mulder_Clément_Linnarsson_Paloski_Wuyts_Zange_Frings-Meuthen_Johannes_Shushakov_Grunewald_et
al._2014, title={Musculoskeletal effects of 5 days of bed rest with and without
locomotion replacement training}, volume={115}, DOI={10.1007/s00421-014-3045-0},
number={4}, journal={European Journal of Applied Physiology}, publisher={Springer
Science and Business Media LLC}, author={Mulder, E. and Clément, G. and Linnarsson,
D. and Paloski, W. H. and Wuyts, F. P. and Zange, J. and Frings-Meuthen, P. and
Johannes, B. and Shushakov, V. and Grunewald, M. and et al.}, year={2014}, pages={727–738}
}'
chicago: 'Mulder, E., G. Clément, D. Linnarsson, W. H. Paloski, F. P. Wuyts, J.
Zange, P. Frings-Meuthen, et al. “Musculoskeletal Effects of 5 Days of Bed Rest
with and without Locomotion Replacement Training.” European Journal of Applied
Physiology 115, no. 4 (2014): 727–38. https://doi.org/10.1007/s00421-014-3045-0.'
ieee: 'E. Mulder et al., “Musculoskeletal effects of 5 days of bed rest with
and without locomotion replacement training,” European Journal of Applied Physiology,
vol. 115, no. 4, pp. 727–738, 2014, doi: 10.1007/s00421-014-3045-0.'
mla: Mulder, E., et al. “Musculoskeletal Effects of 5 Days of Bed Rest with and
without Locomotion Replacement Training.” European Journal of Applied Physiology,
vol. 115, no. 4, Springer Science and Business Media LLC, 2014, pp. 727–38, doi:10.1007/s00421-014-3045-0.
short: E. Mulder, G. Clément, D. Linnarsson, W.H. Paloski, F.P. Wuyts, J. Zange,
P. Frings-Meuthen, B. Johannes, V. Shushakov, M. Grunewald, N. Maassen, J. Bühlmeier,
J. Rittweger, European Journal of Applied Physiology 115 (2014) 727–738.
date_created: 2022-09-15T09:36:45Z
date_updated: 2022-09-15T09:55:02Z
doi: 10.1007/s00421-014-3045-0
extern: '1'
intvolume: ' 115'
issue: '4'
keyword:
- Physiology (medical)
- Public Health
- Environmental and Occupational Health
- Orthopedics and Sports Medicine
- General Medicine
- Public Health
- Environmental and Occupational Health
- Physiology
language:
- iso: eng
page: 727-738
publication: European Journal of Applied Physiology
publication_identifier:
issn:
- 1439-6319
- 1439-6327
publication_status: published
publisher: Springer Science and Business Media LLC
status: public
title: Musculoskeletal effects of 5 days of bed rest with and without locomotion replacement
training
type: journal_article
user_id: '89838'
volume: 115
year: '2014'
...
---
_id: '11716'
abstract:
- lang: eng
text: The accuracy of automatic speech recognition systems in noisy and reverberant
environments can be improved notably by exploiting the uncertainty of the estimated
speech features using so-called uncertainty-of-observation techniques. In this
paper, we introduce a new Bayesian decision rule that can serve as a mathematical
framework from which both known and new uncertainty-of-observation techniques
can be either derived or approximated. The new decision rule in its direct form
leads to the new significance decoding approach for Gaussian mixture models, which
results in better performance compared to standard uncertainty-of-observation
techniques in different additive and convolutive noise scenarios.
author:
- first_name: Ahmed H.
full_name: Abdelaziz, Ahmed H.
last_name: Abdelaziz
- first_name: Steffen
full_name: Zeiler, Steffen
last_name: Zeiler
- first_name: Dorothea
full_name: Kolossa, Dorothea
last_name: Kolossa
- 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: 'Abdelaziz AH, Zeiler S, Kolossa D, Leutnant V, Haeb-Umbach R. GMM-based significance
decoding. In: Acoustics, Speech and Signal Processing (ICASSP), 2013 IEEE International
Conference On. ; 2013:6827-6831. doi:10.1109/ICASSP.2013.6638984'
apa: Abdelaziz, A. H., Zeiler, S., Kolossa, D., Leutnant, V., & Haeb-Umbach,
R. (2013). GMM-based significance decoding. In Acoustics, Speech and Signal
Processing (ICASSP), 2013 IEEE International Conference on (pp. 6827–6831).
https://doi.org/10.1109/ICASSP.2013.6638984
bibtex: '@inproceedings{Abdelaziz_Zeiler_Kolossa_Leutnant_Haeb-Umbach_2013, title={GMM-based
significance decoding}, DOI={10.1109/ICASSP.2013.6638984},
booktitle={Acoustics, Speech and Signal Processing (ICASSP), 2013 IEEE International
Conference on}, author={Abdelaziz, Ahmed H. and Zeiler, Steffen and Kolossa, Dorothea
and Leutnant, Volker and Haeb-Umbach, Reinhold}, year={2013}, pages={6827–6831}
}'
chicago: Abdelaziz, Ahmed H., Steffen Zeiler, Dorothea Kolossa, Volker Leutnant,
and Reinhold Haeb-Umbach. “GMM-Based Significance Decoding.” In Acoustics,
Speech and Signal Processing (ICASSP), 2013 IEEE International Conference On,
6827–31, 2013. https://doi.org/10.1109/ICASSP.2013.6638984.
ieee: A. H. Abdelaziz, S. Zeiler, D. Kolossa, V. Leutnant, and R. Haeb-Umbach, “GMM-based
significance decoding,” in Acoustics, Speech and Signal Processing (ICASSP),
2013 IEEE International Conference on, 2013, pp. 6827–6831.
mla: Abdelaziz, Ahmed H., et al. “GMM-Based Significance Decoding.” Acoustics,
Speech and Signal Processing (ICASSP), 2013 IEEE International Conference On,
2013, pp. 6827–31, doi:10.1109/ICASSP.2013.6638984.
short: 'A.H. Abdelaziz, S. Zeiler, D. Kolossa, V. Leutnant, R. Haeb-Umbach, in:
Acoustics, Speech and Signal Processing (ICASSP), 2013 IEEE International Conference
On, 2013, pp. 6827–6831.'
date_created: 2019-07-12T05:26:53Z
date_updated: 2022-01-06T06:51:07Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2013.6638984
keyword:
- Bayes methods
- Gaussian processes
- convolution
- decision theory
- decoding
- noise
- reverberation
- speech coding
- speech recognition
- Bayesian decision rule
- GMM
- Gaussian mixture models
- additive noise scenarios
- automatic speech recognition systems
- convolutive noise scenarios
- decoding approach
- mathematical framework
- reverberant environments
- significance decoding
- speech feature estimation
- uncertainty-of-observation techniques
- Hidden Markov models
- Maximum likelihood decoding
- Noise
- Speech
- Speech recognition
- Uncertainty
- Uncertainty-of-observation
- modified imputation
- noise robust speech recognition
- significance decoding
- uncertainty decoding
language:
- iso: eng
page: 6827-6831
publication: Acoustics, Speech and Signal Processing (ICASSP), 2013 IEEE International
Conference on
publication_identifier:
issn:
- 1520-6149
status: public
title: GMM-based significance decoding
type: conference
user_id: '44006'
year: '2013'
...
---
_id: '11841'
abstract:
- lang: eng
text: Recently, substantial progress has been made in the field of reverberant speech
signal processing, including both single- and multichannel de-reverberation techniques,
and automatic speech recognition (ASR) techniques robust to reverberation. To
evaluate state-of-the-art algorithms and obtain new insights regarding potential
future research directions, we propose a common evaluation framework including
datasets, tasks, and evaluation metrics for both speech enhancement and ASR techniques.
The proposed framework will be used as a common basis for the REVERB (REverberant
Voice Enhancement and Recognition Benchmark) challenge. This paper describes the
rationale behind the challenge, and provides a detailed description of the evaluation
framework and benchmark results.
author:
- first_name: Keisuke
full_name: Kinoshita, Keisuke
last_name: Kinoshita
- first_name: Marc
full_name: Delcroix, Marc
last_name: Delcroix
- first_name: Takuya
full_name: Yoshioka, Takuya
last_name: Yoshioka
- first_name: Tomohiro
full_name: Nakatani, Tomohiro
last_name: Nakatani
- first_name: Emanuel
full_name: Habets, Emanuel
last_name: Habets
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
- first_name: Volker
full_name: Leutnant, Volker
last_name: Leutnant
- first_name: Armin
full_name: Sehr, Armin
last_name: Sehr
- first_name: Walter
full_name: Kellermann, Walter
last_name: Kellermann
- first_name: Roland
full_name: Maas, Roland
last_name: Maas
- first_name: Sharon
full_name: Gannot, Sharon
last_name: Gannot
- first_name: Bhiksha
full_name: Raj, Bhiksha
last_name: Raj
citation:
ama: 'Kinoshita K, Delcroix M, Yoshioka T, et al. The reverb challenge: a common
evaluation framework for dereverberation and recognition of reverberant speech.
In: IEEE Workshop on Applications of Signal Processing to Audio and Acoustics
. ; 2013:22-23.'
apa: 'Kinoshita, K., Delcroix, M., Yoshioka, T., Nakatani, T., Habets, E., Haeb-Umbach,
R., … Raj, B. (2013). The reverb challenge: a common evaluation framework for
dereverberation and recognition of reverberant speech. In IEEE Workshop on
Applications of Signal Processing to Audio and Acoustics (pp. 22–23).'
bibtex: '@inproceedings{Kinoshita_Delcroix_Yoshioka_Nakatani_Habets_Haeb-Umbach_Leutnant_Sehr_Kellermann_Maas_et
al._2013, title={The reverb challenge: a common evaluation framework for dereverberation
and recognition of reverberant speech}, booktitle={ IEEE Workshop on Applications
of Signal Processing to Audio and Acoustics }, author={Kinoshita, Keisuke and
Delcroix, Marc and Yoshioka, Takuya and Nakatani, Tomohiro and Habets, Emanuel
and Haeb-Umbach, Reinhold and Leutnant, Volker and Sehr, Armin and Kellermann,
Walter and Maas, Roland and et al.}, year={2013}, pages={22–23} }'
chicago: 'Kinoshita, Keisuke, Marc Delcroix, Takuya Yoshioka, Tomohiro Nakatani,
Emanuel Habets, Reinhold Haeb-Umbach, Volker Leutnant, et al. “The Reverb Challenge:
A Common Evaluation Framework for Dereverberation and Recognition of Reverberant
Speech.” In IEEE Workshop on Applications of Signal Processing to Audio and
Acoustics , 22–23, 2013.'
ieee: 'K. Kinoshita et al., “The reverb challenge: a common evaluation framework
for dereverberation and recognition of reverberant speech,” in IEEE Workshop
on Applications of Signal Processing to Audio and Acoustics , 2013, pp. 22–23.'
mla: 'Kinoshita, Keisuke, et al. “The Reverb Challenge: A Common Evaluation Framework
for Dereverberation and Recognition of Reverberant Speech.” IEEE Workshop
on Applications of Signal Processing to Audio and Acoustics , 2013, pp. 22–23.'
short: 'K. Kinoshita, M. Delcroix, T. Yoshioka, T. Nakatani, E. Habets, R. Haeb-Umbach,
V. Leutnant, A. Sehr, W. Kellermann, R. Maas, S. Gannot, B. Raj, in: IEEE Workshop
on Applications of Signal Processing to Audio and Acoustics , 2013, pp. 22–23.'
date_created: 2019-07-12T05:29:17Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
keyword:
- Reverberant speech
- dereverberation
- ASR
- evaluation
- challenge
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2013/Reverb2013.pdf
oa: '1'
page: ' 22-23 '
publication: ' IEEE Workshop on Applications of Signal Processing to Audio and Acoustics '
status: public
title: 'The reverb challenge: a common evaluation framework for dereverberation and
recognition of reverberant speech'
type: conference
user_id: '44006'
year: '2013'
...
---
_id: '11862'
abstract:
- lang: eng
text: In this contribution we extend a previously proposed Bayesian approach for
the enhancement of reverberant logarithmic mel power spectral coefficients for
robust automatic speech recognition to the additional compensation of background
noise. A recently proposed observation model is employed whose time-variant observation
error statistics are obtained as a side product of the inference of the a posteriori
probability density function of the clean speech feature vectors. Further a reduction
of the computational effort and the memory requirements are achieved by using
a recursive formulation of the observation model. The performance of the proposed
algorithms is first experimentally studied on a connected digits recognition task
with artificially created noisy reverberant data. It is shown that the use of
the time-variant observation error model leads to a significant error rate reduction
at low signal-to-noise ratios compared to a time-invariant model. Further experiments
were conducted on a 5000 word task recorded in a reverberant and noisy environment.
A significant word error rate reduction was obtained demonstrating the effectiveness
of the approach on real-world data.
author:
- first_name: Volker
full_name: Leutnant, Volker
last_name: Leutnant
- 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: Leutnant V, Krueger A, Haeb-Umbach R. Bayesian Feature Enhancement for Reverberation
and Noise Robust Speech Recognition. IEEE Transactions on Audio, Speech, and
Language Processing. 2013;21(8):1640-1652. doi:10.1109/TASL.2013.2258013
apa: Leutnant, V., Krueger, A., & Haeb-Umbach, R. (2013). Bayesian Feature Enhancement
for Reverberation and Noise Robust Speech Recognition. IEEE Transactions on
Audio, Speech, and Language Processing, 21(8), 1640–1652. https://doi.org/10.1109/TASL.2013.2258013
bibtex: '@article{Leutnant_Krueger_Haeb-Umbach_2013, title={Bayesian Feature Enhancement
for Reverberation and Noise Robust Speech Recognition}, volume={21}, DOI={10.1109/TASL.2013.2258013},
number={8}, journal={IEEE Transactions on Audio, Speech, and Language Processing},
author={Leutnant, Volker and Krueger, Alexander and Haeb-Umbach, Reinhold}, year={2013},
pages={1640–1652} }'
chicago: 'Leutnant, Volker, Alexander Krueger, and Reinhold Haeb-Umbach. “Bayesian
Feature Enhancement for Reverberation and Noise Robust Speech Recognition.” IEEE
Transactions on Audio, Speech, and Language Processing 21, no. 8 (2013): 1640–52.
https://doi.org/10.1109/TASL.2013.2258013.'
ieee: V. Leutnant, A. Krueger, and R. Haeb-Umbach, “Bayesian Feature Enhancement
for Reverberation and Noise Robust Speech Recognition,” IEEE Transactions on
Audio, Speech, and Language Processing, vol. 21, no. 8, pp. 1640–1652, 2013.
mla: Leutnant, Volker, et al. “Bayesian Feature Enhancement for Reverberation and
Noise Robust Speech Recognition.” IEEE Transactions on Audio, Speech, and Language
Processing, vol. 21, no. 8, 2013, pp. 1640–52, doi:10.1109/TASL.2013.2258013.
short: V. Leutnant, A. Krueger, R. Haeb-Umbach, IEEE Transactions on Audio, Speech,
and Language Processing 21 (2013) 1640–1652.
date_created: 2019-07-12T05:29:42Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
doi: 10.1109/TASL.2013.2258013
intvolume: ' 21'
issue: '8'
keyword:
- Bayes methods
- compensation
- error statistics
- reverberation
- speech recognition
- Bayesian feature enhancement
- background noise
- clean speech feature vectors
- compensation
- connected digits recognition task
- error statistics
- memory requirements
- noisy reverberant data
- posteriori probability density function
- recursive formulation
- reverberant logarithmic mel power spectral coefficients
- robust automatic speech recognition
- signal-to-noise ratios
- time-variant observation
- word error rate reduction
- Robust automatic speech recognition
- model-based Bayesian feature enhancement
- observation model for reverberant and noisy speech
- recursive observation model
language:
- iso: eng
page: 1640-1652
publication: IEEE Transactions on Audio, Speech, and Language Processing
status: public
title: Bayesian Feature Enhancement for Reverberation and Noise Robust Speech Recognition
type: journal_article
user_id: '44006'
volume: 21
year: '2013'
...
---
_id: '11917'
abstract:
- lang: eng
text: In this paper we present a speech presence probability (SPP) estimation algorithmwhich
exploits both temporal and spectral correlations of speech. To this end, the SPP
estimation is formulated as the posterior probability estimation of the states
of a two-dimensional (2D) Hidden Markov Model (HMM). We derive an iterative algorithm
to decode the 2D-HMM which is based on the turbo principle. The experimental results
show that indeed the SPP estimates improve from iteration to iteration, and further
clearly outperform another state-of-the-art SPP estimation algorithm.
author:
- first_name: Dang Hai Tran
full_name: Vu, Dang Hai Tran
last_name: Vu
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: 'Vu DHT, Haeb-Umbach R. Using the turbo principle for exploiting temporal and
spectral correlations in speech presence probability estimation. In: 38th International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2013). ; 2013:863-867.
doi:10.1109/ICASSP.2013.6637771'
apa: Vu, D. H. T., & Haeb-Umbach, R. (2013). Using the turbo principle for exploiting
temporal and spectral correlations in speech presence probability estimation.
In 38th International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2013) (pp. 863–867). https://doi.org/10.1109/ICASSP.2013.6637771
bibtex: '@inproceedings{Vu_Haeb-Umbach_2013, title={Using the turbo principle for
exploiting temporal and spectral correlations in speech presence probability estimation},
DOI={10.1109/ICASSP.2013.6637771},
booktitle={38th International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2013)}, author={Vu, Dang Hai Tran and Haeb-Umbach, Reinhold}, year={2013},
pages={863–867} }'
chicago: Vu, Dang Hai Tran, and Reinhold Haeb-Umbach. “Using the Turbo Principle
for Exploiting Temporal and Spectral Correlations in Speech Presence Probability
Estimation.” In 38th International Conference on Acoustics, Speech and Signal
Processing (ICASSP 2013), 863–67, 2013. https://doi.org/10.1109/ICASSP.2013.6637771.
ieee: D. H. T. Vu and R. Haeb-Umbach, “Using the turbo principle for exploiting
temporal and spectral correlations in speech presence probability estimation,”
in 38th International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2013), 2013, pp. 863–867.
mla: Vu, Dang Hai Tran, and Reinhold Haeb-Umbach. “Using the Turbo Principle for
Exploiting Temporal and Spectral Correlations in Speech Presence Probability Estimation.”
38th International Conference on Acoustics, Speech and Signal Processing (ICASSP
2013), 2013, pp. 863–67, doi:10.1109/ICASSP.2013.6637771.
short: 'D.H.T. Vu, R. Haeb-Umbach, in: 38th International Conference on Acoustics,
Speech and Signal Processing (ICASSP 2013), 2013, pp. 863–867.'
date_created: 2019-07-12T05:30:45Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2013.6637771
keyword:
- correlation methods
- estimation theory
- hidden Markov models
- iterative methods
- probability
- spectral analysis
- speech processing
- 2D HMM
- SPP estimates
- iterative algorithm
- posterior probability estimation
- spectral correlation
- speech presence probability estimation
- state-of-the-art SPP estimation algorithm
- temporal correlation
- turbo principle
- two-dimensional hidden Markov model
- Correlation
- Decoding
- Estimation
- Iterative decoding
- Noise
- Speech
- Vectors
language:
- iso: eng
page: 863-867
publication: 38th International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2013)
publication_identifier:
issn:
- 1520-6149
status: public
title: Using the turbo principle for exploiting temporal and spectral correlations
in speech presence probability estimation
type: conference
user_id: '44006'
year: '2013'
...
---
_id: '11745'
abstract:
- lang: eng
text: In this paper we present a novel noise power spectral density tracking algorithm
and its use in single-channel speech enhancement. It has the unique feature that
it is able to track the noise statistics even if speech is dominant in a given
time-frequency bin. As a consequence it can follow non-stationary noise superposed
by speech, even in the critical case of rising noise power. The algorithm requires
an initial estimate of the power spectrum of speech and is thus meant to be used
as a postprocessor to a first speech enhancement stage. An experimental comparison
with a state-of-the-art noise tracking algorithm demonstrates lower estimation
errors under low SNR conditions and smaller fluctuations of the estimated values,
resulting in improved speech quality as measured by PESQ scores.
author:
- first_name: Aleksej
full_name: Chinaev, Aleksej
last_name: Chinaev
- first_name: Alexander
full_name: Krueger, Alexander
last_name: Krueger
- 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: 'Chinaev A, Krueger A, Tran Vu DH, Haeb-Umbach R. Improved Noise Power Spectral
Density Tracking by a MAP-based Postprocessor. In: 37th International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2012). ; 2012.'
apa: Chinaev, A., Krueger, A., Tran Vu, D. H., & Haeb-Umbach, R. (2012). Improved
Noise Power Spectral Density Tracking by a MAP-based Postprocessor. In 37th
International Conference on Acoustics, Speech and Signal Processing (ICASSP 2012).
bibtex: '@inproceedings{Chinaev_Krueger_Tran Vu_Haeb-Umbach_2012, title={Improved
Noise Power Spectral Density Tracking by a MAP-based Postprocessor}, booktitle={37th
International Conference on Acoustics, Speech and Signal Processing (ICASSP 2012)},
author={Chinaev, Aleksej and Krueger, Alexander and Tran Vu, Dang Hai and Haeb-Umbach,
Reinhold}, year={2012} }'
chicago: Chinaev, Aleksej, Alexander Krueger, Dang Hai Tran Vu, and Reinhold Haeb-Umbach.
“Improved Noise Power Spectral Density Tracking by a MAP-Based Postprocessor.”
In 37th International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2012), 2012.
ieee: A. Chinaev, A. Krueger, D. H. Tran Vu, and R. Haeb-Umbach, “Improved Noise
Power Spectral Density Tracking by a MAP-based Postprocessor,” in 37th International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2012), 2012.
mla: Chinaev, Aleksej, et al. “Improved Noise Power Spectral Density Tracking by
a MAP-Based Postprocessor.” 37th International Conference on Acoustics, Speech
and Signal Processing (ICASSP 2012), 2012.
short: 'A. Chinaev, A. Krueger, D.H. Tran Vu, R. Haeb-Umbach, in: 37th International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2012), 2012.'
date_created: 2019-07-12T05:27:26Z
date_updated: 2022-01-06T06:51:08Z
department:
- _id: '54'
keyword:
- MAP parameter estimation
- noise power estimation
- speech enhancement
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2012/ChKrDaHa12.pdf
oa: '1'
publication: 37th International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2012)
related_material:
link:
- description: Presentation
relation: supplementary_material
url: https://groups.uni-paderborn.de/nt/pubs/2012/ChKrDaHa12_Talk.pdf
status: public
title: Improved Noise Power Spectral Density Tracking by a MAP-based Postprocessor
type: conference
user_id: '44006'
year: '2012'
...
---
_id: '11864'
abstract:
- lang: eng
text: In this work, an observation model for the joint compensation of noise and
reverberation in the logarithmic mel power spectral density domain is considered.
It relates the features of the noisy reverberant speech to those of the non-reverberant
speech and the noise. In contrast to enhancement of features only corrupted by
reverberation (reverberant features), enhancement of noisy reverberant features
requires a more sophisticated model for the error introduced by the proposed observation
model. In a first consideration, it will be shown that this error is highly dependent
on the instantaneous ratio of the power of reverberant speech to the power of
the noise and, moreover, sensitive to the phase between reverberant speech and
noise in the short-time discrete Fourier domain. Afterwards, a statistically motivated
approach will be presented allowing for the model of the observation error to
be inferred from the error model previously used for the reverberation only case.
Finally, the developed observation error model will be utilized in a Bayesian
feature enhancement scheme, leading to improvements in word accuracy on the AURORA5
database.
author:
- first_name: Volker
full_name: Leutnant, Volker
last_name: Leutnant
- 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: 'Leutnant V, Krueger A, Haeb-Umbach R. A Statistical Observation Model For
Noisy Reverberant Speech Features and its Application to Robust ASR. In: Signal
Processing, Communications and Computing (ICSPCC), 2012 IEEE International Conference
On. ; 2012.'
apa: Leutnant, V., Krueger, A., & Haeb-Umbach, R. (2012). A Statistical Observation
Model For Noisy Reverberant Speech Features and its Application to Robust ASR.
In Signal Processing, Communications and Computing (ICSPCC), 2012 IEEE International
Conference on.
bibtex: '@inproceedings{Leutnant_Krueger_Haeb-Umbach_2012, title={A Statistical
Observation Model For Noisy Reverberant Speech Features and its Application to
Robust ASR}, booktitle={Signal Processing, Communications and Computing (ICSPCC),
2012 IEEE International Conference on}, author={Leutnant, Volker and Krueger,
Alexander and Haeb-Umbach, Reinhold}, year={2012} }'
chicago: Leutnant, Volker, Alexander Krueger, and Reinhold Haeb-Umbach. “A Statistical
Observation Model For Noisy Reverberant Speech Features and Its Application to
Robust ASR.” In Signal Processing, Communications and Computing (ICSPCC), 2012
IEEE International Conference On, 2012.
ieee: V. Leutnant, A. Krueger, and R. Haeb-Umbach, “A Statistical Observation Model
For Noisy Reverberant Speech Features and its Application to Robust ASR,” in Signal
Processing, Communications and Computing (ICSPCC), 2012 IEEE International Conference
on, 2012.
mla: Leutnant, Volker, et al. “A Statistical Observation Model For Noisy Reverberant
Speech Features and Its Application to Robust ASR.” Signal Processing, Communications
and Computing (ICSPCC), 2012 IEEE International Conference On, 2012.
short: 'V. Leutnant, A. Krueger, R. Haeb-Umbach, in: Signal Processing, Communications
and Computing (ICSPCC), 2012 IEEE International Conference On, 2012.'
date_created: 2019-07-12T05:29:44Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
keyword:
- Robust Automatic Speech Recognition
- Bayesian feature enhancement
- observation model for reverberant and noisy speech
language:
- iso: eng
main_file_link:
- open_access: '1'
url: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=6335731
oa: '1'
publication: Signal Processing, Communications and Computing (ICSPCC), 2012 IEEE International
Conference on
status: public
title: A Statistical Observation Model For Noisy Reverberant Speech Features and its
Application to Robust ASR
type: conference
user_id: '44006'
year: '2012'
...
---
_id: '11850'
abstract:
- lang: eng
text: In this paper, we present a novel blocking matrix and fixed beamformer design
for a generalized sidelobe canceler for speech enhancement in a reverberant enclosure.
They are based on a new method for estimating the acoustical transfer function
ratios in the presence of stationary noise. The estimation method relies on solving
a generalized eigenvalue problem in each frequency bin. An adaptive eigenvector
tracking utilizing the power iteration method is employed and shown to achieve
a high convergence speed. Simulation results demonstrate that the proposed beamformer
leads to better noise and interference reduction and reduced speech distortions
compared to other blocking matrix designs from the literature.
author:
- first_name: Alexander
full_name: Krueger, Alexander
last_name: Krueger
- first_name: Ernst
full_name: Warsitz, Ernst
last_name: Warsitz
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: Krueger A, Warsitz E, Haeb-Umbach R. Speech Enhancement With a GSC-Like Structure
Employing Eigenvector-Based Transfer Function Ratios Estimation. IEEE Transactions
on Audio, Speech, and Language Processing. 2011;19(1):206-219. doi:10.1109/TASL.2010.2047324
apa: Krueger, A., Warsitz, E., & Haeb-Umbach, R. (2011). Speech Enhancement
With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios
Estimation. IEEE Transactions on Audio, Speech, and Language Processing,
19(1), 206–219. https://doi.org/10.1109/TASL.2010.2047324
bibtex: '@article{Krueger_Warsitz_Haeb-Umbach_2011, title={Speech Enhancement With
a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios Estimation},
volume={19}, DOI={10.1109/TASL.2010.2047324},
number={1}, journal={IEEE Transactions on Audio, Speech, and Language Processing},
author={Krueger, Alexander and Warsitz, Ernst and Haeb-Umbach, Reinhold}, year={2011},
pages={206–219} }'
chicago: 'Krueger, Alexander, Ernst Warsitz, and Reinhold Haeb-Umbach. “Speech Enhancement
With a GSC-Like Structure Employing Eigenvector-Based Transfer Function Ratios
Estimation.” IEEE Transactions on Audio, Speech, and Language Processing
19, no. 1 (2011): 206–19. https://doi.org/10.1109/TASL.2010.2047324.'
ieee: A. Krueger, E. Warsitz, and R. Haeb-Umbach, “Speech Enhancement With a GSC-Like
Structure Employing Eigenvector-Based Transfer Function Ratios Estimation,” IEEE
Transactions on Audio, Speech, and Language Processing, vol. 19, no. 1, pp.
206–219, 2011.
mla: Krueger, Alexander, et al. “Speech Enhancement With a GSC-Like Structure Employing
Eigenvector-Based Transfer Function Ratios Estimation.” IEEE Transactions on
Audio, Speech, and Language Processing, vol. 19, no. 1, 2011, pp. 206–19,
doi:10.1109/TASL.2010.2047324.
short: A. Krueger, E. Warsitz, R. Haeb-Umbach, IEEE Transactions on Audio, Speech,
and Language Processing 19 (2011) 206–219.
date_created: 2019-07-12T05:29:28Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
doi: 10.1109/TASL.2010.2047324
intvolume: ' 19'
issue: '1'
keyword:
- acoustical transfer function ratio
- adaptive eigenvector tracking
- array signal processing
- beamformer design
- blocking matrix
- eigenvalues and eigenfunctions
- eigenvector-based transfer function ratios estimation
- generalized sidelobe canceler
- interference reduction
- iterative methods
- power iteration method
- reduced speech distortions
- reverberant enclosure
- reverberation
- speech enhancement
- stationary noise
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2011/KrWaHa11.pdf
oa: '1'
page: 206-219
publication: IEEE Transactions on Audio, Speech, and Language Processing
status: public
title: Speech Enhancement With a GSC-Like Structure Employing Eigenvector-Based Transfer
Function Ratios Estimation
type: journal_article
user_id: '44006'
volume: 19
year: '2011'
...
---
_id: '33400'
abstract:
- lang: eng
text: 'We examined, in immobilization, the effect of a diet high in sodium
chloride (NaCl) on bone markers, nitrogen balance, and acid-base status. Eight
healthy male test subjects participated in a 14-day head-down-tilt bed rest (HDBR)
study. During the bed rest period they received, in a randomized crossover design,
a high (7.7 meq Na+/kg body wt per day) and a low (0.7 meq
Na+/kg body wt per day) NaCl diet. As expected, 24-h excretion
of urinary calcium was significantly greater in the high-NaCl-intake HDBR phase
than in the low-NaCl-intake HDBR phase ( P < 0.001). High NaCl intake caused
a 43–50% greater excretion of the bone resorption markers COOH- (CTX) and NH2-
(NTX) terminal telopeptide of type I collagen in HDBR than low NaCl in HDBR (CTX/NTX:
P < 0.001). Serum concentrations of the bone formation markers bone-specific
alkaline phosphatase (bAP) and NH2-terminal propeptide of
type I procollagen (PINP) were identical in both NaCl intake phases. High NaCl
intake led to a more negative nitrogen balance in HDBR ( P < 0.001). Changes
were accompanied by increased serum chloride concentration ( P = 0.008), reduced
blood bicarbonate ( P = 0.017), and base excess ( P = 0.009) whereas net acid
excretion was lower during high than during low NaCl intake in immobilization
( P < 0.001). High NaCl intake during immobilization exacerbates disuse-induced
bone and muscle loss by causing further protein wasting and an increase in bone
resorption. Changes in the acid-base status, mainly caused by disturbances in
electrolyte metabolism, seem to determine NaCl-induced degradation processes.'
author:
- first_name: Petra
full_name: Frings-Meuthen, Petra
last_name: Frings-Meuthen
- first_name: Judith
full_name: Bühlmeier, Judith
id: '89838'
last_name: Bühlmeier
- first_name: Natalie
full_name: Baecker, Natalie
last_name: Baecker
- first_name: Peter
full_name: Stehle, Peter
last_name: Stehle
- first_name: Rolf
full_name: Fimmers, Rolf
last_name: Fimmers
- first_name: Francisca
full_name: May, Francisca
last_name: May
- first_name: Goetz
full_name: Kluge, Goetz
last_name: Kluge
- first_name: Martina
full_name: Heer, Martina
last_name: Heer
citation:
ama: Frings-Meuthen P, Bühlmeier J, Baecker N, et al. High sodium chloride intake
exacerbates immobilization-induced bone resorption and protein losses. Journal
of Applied Physiology. 2011;111(2):537-542. doi:10.1152/japplphysiol.00454.2011
apa: Frings-Meuthen, P., Bühlmeier, J., Baecker, N., Stehle, P., Fimmers, R., May,
F., Kluge, G., & Heer, M. (2011). High sodium chloride intake exacerbates
immobilization-induced bone resorption and protein losses. Journal of Applied
Physiology, 111(2), 537–542. https://doi.org/10.1152/japplphysiol.00454.2011
bibtex: '@article{Frings-Meuthen_Bühlmeier_Baecker_Stehle_Fimmers_May_Kluge_Heer_2011,
title={High sodium chloride intake exacerbates immobilization-induced bone resorption
and protein losses}, volume={111}, DOI={10.1152/japplphysiol.00454.2011},
number={2}, journal={Journal of Applied Physiology}, publisher={American Physiological
Society}, author={Frings-Meuthen, Petra and Bühlmeier, Judith and Baecker, Natalie
and Stehle, Peter and Fimmers, Rolf and May, Francisca and Kluge, Goetz and Heer,
Martina}, year={2011}, pages={537–542} }'
chicago: 'Frings-Meuthen, Petra, Judith Bühlmeier, Natalie Baecker, Peter Stehle,
Rolf Fimmers, Francisca May, Goetz Kluge, and Martina Heer. “High Sodium Chloride
Intake Exacerbates Immobilization-Induced Bone Resorption and Protein Losses.”
Journal of Applied Physiology 111, no. 2 (2011): 537–42. https://doi.org/10.1152/japplphysiol.00454.2011.'
ieee: 'P. Frings-Meuthen et al., “High sodium chloride intake exacerbates
immobilization-induced bone resorption and protein losses,” Journal of Applied
Physiology, vol. 111, no. 2, pp. 537–542, 2011, doi: 10.1152/japplphysiol.00454.2011.'
mla: Frings-Meuthen, Petra, et al. “High Sodium Chloride Intake Exacerbates Immobilization-Induced
Bone Resorption and Protein Losses.” Journal of Applied Physiology, vol.
111, no. 2, American Physiological Society, 2011, pp. 537–42, doi:10.1152/japplphysiol.00454.2011.
short: P. Frings-Meuthen, J. Bühlmeier, N. Baecker, P. Stehle, R. Fimmers, F. May,
G. Kluge, M. Heer, Journal of Applied Physiology 111 (2011) 537–542.
date_created: 2022-09-15T09:37:29Z
date_updated: 2022-09-15T09:43:45Z
doi: 10.1152/japplphysiol.00454.2011
extern: '1'
intvolume: ' 111'
issue: '2'
keyword:
- Physiology (medical)
- Physiology
language:
- iso: eng
page: 537-542
publication: Journal of Applied Physiology
publication_identifier:
issn:
- 8750-7587
- 1522-1601
publication_status: published
publisher: American Physiological Society
status: public
title: High sodium chloride intake exacerbates immobilization-induced bone resorption
and protein losses
type: journal_article
user_id: '89838'
volume: 111
year: '2011'
...
---
_id: '17233'
abstract:
- lang: eng
text: 'It has been proposed that the design of robots might benefit from interactions
that are similar to caregiver–child interactions, which is tailored to children’s
respective capacities to a high degree. However, so far little is known about
how people adapt their tutoring behaviour to robots and whether robots can evoke
input that is similar to child-directed interaction. The paper presents detailed
analyses of speakers’ linguistic and non-linguistic behaviour, such as action
demonstration, in two comparable situations: In one experiment, parents described
and explained to their nonverbal infants the use of certain everyday objects;
in the other experiment, participants tutored a simulated robot on the same objects.
The results, which show considerable differences between the two situations on
almost all measures, are discussed in the light of the computer-as-social-actor
paradigm and the register hypothesis.'
author:
- first_name: Kerstin
full_name: Fischer, Kerstin
last_name: Fischer
- first_name: Kilian
full_name: Foth, Kilian
last_name: Foth
- first_name: Katharina
full_name: Rohlfing, Katharina
id: '50352'
last_name: Rohlfing
- first_name: Britta
full_name: Wrede, Britta
last_name: Wrede
citation:
ama: 'Fischer K, Foth K, Rohlfing K, Wrede B. Mindful tutors: Linguistic choice
and action demonstration in speech to infants and a simulated robot. Interaction
Studies. 2011;12(1):134-161. doi:10.1075/is.12.1.06fis'
apa: 'Fischer, K., Foth, K., Rohlfing, K., & Wrede, B. (2011). Mindful tutors:
Linguistic choice and action demonstration in speech to infants and a simulated
robot. Interaction Studies, 12(1), 134–161. https://doi.org/10.1075/is.12.1.06fis'
bibtex: '@article{Fischer_Foth_Rohlfing_Wrede_2011, title={Mindful tutors: Linguistic
choice and action demonstration in speech to infants and a simulated robot}, volume={12},
DOI={10.1075/is.12.1.06fis},
number={1}, journal={Interaction Studies}, publisher={John Benjamins Publishing
Company}, author={Fischer, Kerstin and Foth, Kilian and Rohlfing, Katharina and
Wrede, Britta}, year={2011}, pages={134–161} }'
chicago: 'Fischer, Kerstin, Kilian Foth, Katharina Rohlfing, and Britta Wrede. “Mindful
Tutors: Linguistic Choice and Action Demonstration in Speech to Infants and a
Simulated Robot.” Interaction Studies 12, no. 1 (2011): 134–61. https://doi.org/10.1075/is.12.1.06fis.'
ieee: 'K. Fischer, K. Foth, K. Rohlfing, and B. Wrede, “Mindful tutors: Linguistic
choice and action demonstration in speech to infants and a simulated robot,” Interaction
Studies, vol. 12, no. 1, pp. 134–161, 2011, doi: 10.1075/is.12.1.06fis.'
mla: 'Fischer, Kerstin, et al. “Mindful Tutors: Linguistic Choice and Action Demonstration
in Speech to Infants and a Simulated Robot.” Interaction Studies, vol.
12, no. 1, John Benjamins Publishing Company, 2011, pp. 134–61, doi:10.1075/is.12.1.06fis.'
short: K. Fischer, K. Foth, K. Rohlfing, B. Wrede, Interaction Studies 12 (2011)
134–161.
date_created: 2020-06-24T13:01:57Z
date_updated: 2023-02-01T12:56:04Z
department:
- _id: '749'
doi: 10.1075/is.12.1.06fis
intvolume: ' 12'
issue: '1'
keyword:
- human–robot interaction (HRI)
- social communication
- register theory
- motionese
- robotese
- child-directed speech (CDS)
- motherese
- mindless transfer
- computers-as-social-actors
language:
- iso: eng
page: 134-161
publication: Interaction Studies
publication_identifier:
issn:
- 1572-0381
publisher: John Benjamins Publishing Company
status: public
title: 'Mindful tutors: Linguistic choice and action demonstration in speech to infants
and a simulated robot'
type: journal_article
user_id: '14931'
volume: 12
year: '2011'
...
---
_id: '11846'
abstract:
- lang: eng
text: In this paper, we present a new technique for automatic speech recognition
(ASR) in reverberant environments. Our approach is aimed at the enhancement of
the logarithmic Mel power spectrum, which is computed at an intermediate stage
to obtain the widely used Mel frequency cepstral coefficients (MFCCs). Given the
reverberant logarithmic Mel power spectral coefficients (LMPSCs), a minimum mean
square error estimate of the clean LMPSCs is computed by carrying out Bayesian
inference. We employ switching linear dynamical models as an a priori model for
the dynamics of the clean LMPSCs. Further, we derive a stochastic observation
model which relates the clean to the reverberant LMPSCs through a simplified model
of the room impulse response (RIR). This model requires only two parameters, namely
RIR energy and reverberation time, which can be estimated from the captured microphone
signal. The performance of the proposed enhancement technique is studied on the
AURORA5 database and compared to that of constrained maximum-likelihood linear
regression (CMLLR). It is shown by experimental results that our approach significantly
outperforms CMLLR and that up to 80\% of the errors caused by the reverberation
are recovered. In addition to the fact that the approach is compatible with the
standard MFCC feature vectors, it leaves the ASR back-end unchanged. It is of
moderate computational complexity and suitable for real time applications.
author:
- 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: Krueger A, Haeb-Umbach R. Model-Based Feature Enhancement for Reverberant Speech
Recognition. IEEE Transactions on Audio, Speech, and Language Processing.
2010;18(7):1692-1707. doi:10.1109/TASL.2010.2049684
apa: Krueger, A., & Haeb-Umbach, R. (2010). Model-Based Feature Enhancement
for Reverberant Speech Recognition. IEEE Transactions on Audio, Speech, and
Language Processing, 18(7), 1692–1707. https://doi.org/10.1109/TASL.2010.2049684
bibtex: '@article{Krueger_Haeb-Umbach_2010, title={Model-Based Feature Enhancement
for Reverberant Speech Recognition}, volume={18}, DOI={10.1109/TASL.2010.2049684},
number={7}, journal={IEEE Transactions on Audio, Speech, and Language Processing},
author={Krueger, Alexander and Haeb-Umbach, Reinhold}, year={2010}, pages={1692–1707}
}'
chicago: 'Krueger, Alexander, and Reinhold Haeb-Umbach. “Model-Based Feature Enhancement
for Reverberant Speech Recognition.” IEEE Transactions on Audio, Speech, and
Language Processing 18, no. 7 (2010): 1692–1707. https://doi.org/10.1109/TASL.2010.2049684.'
ieee: A. Krueger and R. Haeb-Umbach, “Model-Based Feature Enhancement for Reverberant
Speech Recognition,” IEEE Transactions on Audio, Speech, and Language Processing,
vol. 18, no. 7, pp. 1692–1707, 2010.
mla: Krueger, Alexander, and Reinhold Haeb-Umbach. “Model-Based Feature Enhancement
for Reverberant Speech Recognition.” IEEE Transactions on Audio, Speech, and
Language Processing, vol. 18, no. 7, 2010, pp. 1692–707, doi:10.1109/TASL.2010.2049684.
short: A. Krueger, R. Haeb-Umbach, IEEE Transactions on Audio, Speech, and Language
Processing 18 (2010) 1692–1707.
date_created: 2019-07-12T05:29:23Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
doi: 10.1109/TASL.2010.2049684
intvolume: ' 18'
issue: '7'
keyword:
- ASR
- AURORA5 database
- automatic speech recognition
- Bayesian inference
- belief networks
- CMLLR
- computational complexity
- constrained maximum likelihood linear regression
- least mean squares methods
- LMPSC computation
- logarithmic Mel power spectrum
- maximum likelihood estimation
- Mel frequency cepstral coefficients
- MFCC feature vectors
- microphone signal
- minimum mean square error estimation
- model-based feature enhancement
- regression analysis
- reverberant speech recognition
- reverberation
- RIR energy
- room impulse response
- speech recognition
- stochastic observation model
- stochastic processes
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2010/KrHa10.pdf
oa: '1'
page: 1692-1707
publication: IEEE Transactions on Audio, Speech, and Language Processing
status: public
title: Model-Based Feature Enhancement for Reverberant Speech Recognition
type: journal_article
user_id: '44006'
volume: 18
year: '2010'
...
---
_id: '11913'
abstract:
- lang: eng
text: In this paper we propose to employ directional statistics in a complex vector
space to approach the problem of blind speech separation in the presence of spatially
correlated noise. We interpret the values of the short time Fourier transform
of the microphone signals to be draws from a mixture of complex Watson distributions,
a probabilistic model which naturally accounts for spatial aliasing. The parameters
of the density are related to the a priori source probabilities, the power of
the sources and the transfer function ratios from sources to sensors. Estimation
formulas are derived for these parameters by employing the Expectation Maximization
(EM) algorithm. The E-step corresponds to the estimation of the source presence
probabilities for each time-frequency bin, while the M-step leads to a maximum
signal-to-noise ratio (MaxSNR) beamformer in the presence of uncertainty about
the source activity. Experimental results are reported for an implementation in
a generalized sidelobe canceller (GSC) like spatial beamforming configuration
for 3 speech sources with significant coherent noise in reverberant environments,
demonstrating the usefulness of the novel modeling framework.
author:
- first_name: Dang Hai
full_name: Tran Vu, Dang Hai
last_name: Tran Vu
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: 'Tran Vu DH, Haeb-Umbach R. Blind speech separation employing directional statistics
in an Expectation Maximization framework. In: IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2010). ; 2010:241-244.
doi:10.1109/ICASSP.2010.5495994'
apa: Tran Vu, D. H., & Haeb-Umbach, R. (2010). Blind speech separation employing
directional statistics in an Expectation Maximization framework. In IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2010) (pp. 241–244).
https://doi.org/10.1109/ICASSP.2010.5495994
bibtex: '@inproceedings{Tran Vu_Haeb-Umbach_2010, title={Blind speech separation
employing directional statistics in an Expectation Maximization framework}, DOI={10.1109/ICASSP.2010.5495994},
booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2010)}, author={Tran Vu, Dang Hai and Haeb-Umbach, Reinhold}, year={2010},
pages={241–244} }'
chicago: Tran Vu, Dang Hai, and Reinhold Haeb-Umbach. “Blind Speech Separation Employing
Directional Statistics in an Expectation Maximization Framework.” In IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2010), 241–44,
2010. https://doi.org/10.1109/ICASSP.2010.5495994.
ieee: D. H. Tran Vu and R. Haeb-Umbach, “Blind speech separation employing directional
statistics in an Expectation Maximization framework,” in IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2010), 2010,
pp. 241–244.
mla: Tran Vu, Dang Hai, and Reinhold Haeb-Umbach. “Blind Speech Separation Employing
Directional Statistics in an Expectation Maximization Framework.” IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2010), 2010,
pp. 241–44, doi:10.1109/ICASSP.2010.5495994.
short: 'D.H. Tran Vu, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
Speech and Signal Processing (ICASSP 2010), 2010, pp. 241–244.'
date_created: 2019-07-12T05:30:40Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2010.5495994
keyword:
- array signal processing
- blind source separation
- blind speech separation
- complex vector space
- complex Watson distribution
- directional statistics
- expectation-maximisation algorithm
- expectation maximization algorithm
- Fourier transform
- Fourier transforms
- generalized sidelobe canceller
- interference suppression
- maximum signal-to-noise ratio beamformer
- microphone signal
- probabilistic model
- spatial aliasing
- spatial beamforming configuration
- speech enhancement
- statistical distributions
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2010/DaHa10-2.pdf
oa: '1'
page: 241-244
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2010)
status: public
title: Blind speech separation employing directional statistics in an Expectation
Maximization framework
type: conference
user_id: '44006'
year: '2010'
...
---
_id: '11892'
abstract:
- lang: eng
text: For an environment to be perceived as being smart, contextual information
has to be gathered to adapt the system's behavior and its interface towards the
user. Being a rich source of context information speech can be acquired unobtrusively
by microphone arrays and then processed to extract information about the user
and his environment. In this paper, a system for joint temporal segmentation,
speaker localization, and identification is presented, which is supported by face
identification from video data obtained from a steerable camera. Special attention
is paid to latency aspects and online processing capabilities, as they are important
for the application under investigation, namely ambient communication. It describes
the vision of terminal-less, session-less and multi-modal telecommunication with
remote partners, where the user can move freely within his home while the communication
follows him. The speaker diarization serves as a context source, which has been
integrated in a service-oriented middleware architecture and provided to the application
to select the most appropriate I/O device and to steer the camera towards the
speaker during ambient communication.
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 Diarization of Streaming Audio-Visual
Data for Smart Environments. IEEE Journal of Selected Topics in Signal Processing.
2010;4(5):845-856. doi:10.1109/JSTSP.2010.2050519
apa: Schmalenstroeer, J., & Haeb-Umbach, R. (2010). Online Diarization of Streaming
Audio-Visual Data for Smart Environments. IEEE Journal of Selected Topics in
Signal Processing, 4(5), 845–856. https://doi.org/10.1109/JSTSP.2010.2050519
bibtex: '@article{Schmalenstroeer_Haeb-Umbach_2010, title={Online Diarization of
Streaming Audio-Visual Data for Smart Environments}, volume={4}, DOI={10.1109/JSTSP.2010.2050519},
number={5}, journal={IEEE Journal of Selected Topics in Signal Processing}, author={Schmalenstroeer,
Joerg and Haeb-Umbach, Reinhold}, year={2010}, pages={845–856} }'
chicago: 'Schmalenstroeer, Joerg, and Reinhold Haeb-Umbach. “Online Diarization
of Streaming Audio-Visual Data for Smart Environments.” IEEE Journal of Selected
Topics in Signal Processing 4, no. 5 (2010): 845–56. https://doi.org/10.1109/JSTSP.2010.2050519.'
ieee: 'J. Schmalenstroeer and R. Haeb-Umbach, “Online Diarization of Streaming Audio-Visual
Data for Smart Environments,” IEEE Journal of Selected Topics in Signal Processing,
vol. 4, no. 5, pp. 845–856, 2010, doi: 10.1109/JSTSP.2010.2050519.'
mla: Schmalenstroeer, Joerg, and Reinhold Haeb-Umbach. “Online Diarization of Streaming
Audio-Visual Data for Smart Environments.” IEEE Journal of Selected Topics
in Signal Processing, vol. 4, no. 5, 2010, pp. 845–56, doi:10.1109/JSTSP.2010.2050519.
short: J. Schmalenstroeer, R. Haeb-Umbach, IEEE Journal of Selected Topics in Signal
Processing 4 (2010) 845–856.
date_created: 2019-07-12T05:30:16Z
date_updated: 2023-10-26T08:10:18Z
department:
- _id: '54'
doi: 10.1109/JSTSP.2010.2050519
intvolume: ' 4'
issue: '5'
keyword:
- audio streaming
- audio visual data streaming
- context information speech
- face identification
- face recognition
- image segmentation
- middleware
- multimodal telecommunication
- online diarization
- service oriented middleware architecture
- sessionless telecommunication
- software architecture
- speaker identification
- speaker localization
- speaker recognition
- steerable camera
- telecommunication computing
- temporal segmentation
- terminal-less telecommunication
- video streaming
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2010/ScHa10.pdf
oa: '1'
page: 845-856
publication: IEEE Journal of Selected Topics in Signal Processing
quality_controlled: '1'
status: public
title: Online Diarization of Streaming Audio-Visual Data for Smart Environments
type: journal_article
user_id: '460'
volume: 4
year: '2010'
...
---
_id: '11937'
abstract:
- lang: eng
text: In automatic speech recognition, hidden Markov models (HMMs) are commonly
used for speech decoding, while switching linear dynamic models (SLDMs) can be
employed for a preceding model-based speech feature enhancement. In this paper,
these model types are combined in order to obtain a novel iterative speech feature
enhancement and recognition architecture. It is shown that speech feature enhancement
with SLDMs can be improved by feeding back information from the HMM to the enhancement
stage. Two different feedback structures are derived. In the first, the posteriors
of the HMM states are used to control the model probabilities of the SLDMs, while
in the second they are employed to directly influence the estimate of the speech
feature distribution. Both approaches lead to improvements in recognition accuracy
both on the AURORA2 and AURORA4 databases compared to non-iterative speech feature
enhancement with SLDMs. It is also shown that a combination with uncertainty decoding
further enhances performance.
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. Approaches to Iterative Speech Feature Enhancement
and Recognition. IEEE Transactions on Audio, Speech, and Language Processing.
2009;17(5):974-984. doi:10.1109/TASL.2009.2014894
apa: Windmann, S., & Haeb-Umbach, R. (2009). Approaches to Iterative Speech
Feature Enhancement and Recognition. IEEE Transactions on Audio, Speech, and
Language Processing, 17(5), 974–984. https://doi.org/10.1109/TASL.2009.2014894
bibtex: '@article{Windmann_Haeb-Umbach_2009, title={Approaches to Iterative Speech
Feature Enhancement and Recognition}, volume={17}, DOI={10.1109/TASL.2009.2014894},
number={5}, journal={IEEE Transactions on Audio, Speech, and Language Processing},
author={Windmann, Stefan and Haeb-Umbach, Reinhold}, year={2009}, pages={974–984}
}'
chicago: 'Windmann, Stefan, and Reinhold Haeb-Umbach. “Approaches to Iterative Speech
Feature Enhancement and Recognition.” IEEE Transactions on Audio, Speech, and
Language Processing 17, no. 5 (2009): 974–84. https://doi.org/10.1109/TASL.2009.2014894.'
ieee: S. Windmann and R. Haeb-Umbach, “Approaches to Iterative Speech Feature Enhancement
and Recognition,” IEEE Transactions on Audio, Speech, and Language Processing,
vol. 17, no. 5, pp. 974–984, 2009.
mla: Windmann, Stefan, and Reinhold Haeb-Umbach. “Approaches to Iterative Speech
Feature Enhancement and Recognition.” IEEE Transactions on Audio, Speech, and
Language Processing, vol. 17, no. 5, 2009, pp. 974–84, doi:10.1109/TASL.2009.2014894.
short: S. Windmann, R. Haeb-Umbach, IEEE Transactions on Audio, Speech, and Language
Processing 17 (2009) 974–984.
date_created: 2019-07-12T05:31:08Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/TASL.2009.2014894
intvolume: ' 17'
issue: '5'
keyword:
- AURORA2 databases
- AURORA4 databases
- automatic speech recognition
- feedback structures
- hidden Markov models
- HMM
- iterative methods
- iterative speech feature enhancement
- model probabilities
- speech decoding
- speech enhancement
- speech feature distribution
- speech recognition
- switching linear dynamic models
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2009/WiHa09-1.pdf
oa: '1'
page: 974-984
publication: IEEE Transactions on Audio, Speech, and Language Processing
status: public
title: Approaches to Iterative Speech Feature Enhancement and Recognition
type: journal_article
user_id: '44006'
volume: 17
year: '2009'
...
---
_id: '11938'
abstract:
- lang: eng
text: In this paper, parameter estimation of a state-space model of noise or noisy
speech cepstra is investigated. A blockwise EM algorithm is derived for the estimation
of the state and observation noise covariance from noise-only input data. It is
supposed to be used during the offline training mode of a speech recognizer. Further
a sequential online EM algorithm is developed to adapt the observation noise covariance
on noisy speech cepstra at its input. The estimated parameters are then used in
model-based speech feature enhancement for noise-robust automatic speech recognition.
Experiments on the AURORA4 database lead to improved recognition results with
a linear 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. Parameter Estimation of a State-Space Model of Noise
for Robust Speech Recognition. IEEE Transactions on Audio, Speech, and Language
Processing. 2009;17(8):1577-1590. doi:10.1109/TASL.2009.2023172
apa: Windmann, S., & Haeb-Umbach, R. (2009). Parameter Estimation of a State-Space
Model of Noise for Robust Speech Recognition. IEEE Transactions on Audio, Speech,
and Language Processing, 17(8), 1577–1590. https://doi.org/10.1109/TASL.2009.2023172
bibtex: '@article{Windmann_Haeb-Umbach_2009, title={Parameter Estimation of a State-Space
Model of Noise for Robust Speech Recognition}, volume={17}, DOI={10.1109/TASL.2009.2023172},
number={8}, journal={IEEE Transactions on Audio, Speech, and Language Processing},
author={Windmann, Stefan and Haeb-Umbach, Reinhold}, year={2009}, pages={1577–1590}
}'
chicago: 'Windmann, Stefan, and Reinhold Haeb-Umbach. “Parameter Estimation of a
State-Space Model of Noise for Robust Speech Recognition.” IEEE Transactions
on Audio, Speech, and Language Processing 17, no. 8 (2009): 1577–90. https://doi.org/10.1109/TASL.2009.2023172.'
ieee: S. Windmann and R. Haeb-Umbach, “Parameter Estimation of a State-Space Model
of Noise for Robust Speech Recognition,” IEEE Transactions on Audio, Speech,
and Language Processing, vol. 17, no. 8, pp. 1577–1590, 2009.
mla: Windmann, Stefan, and Reinhold Haeb-Umbach. “Parameter Estimation of a State-Space
Model of Noise for Robust Speech Recognition.” IEEE Transactions on Audio,
Speech, and Language Processing, vol. 17, no. 8, 2009, pp. 1577–90, doi:10.1109/TASL.2009.2023172.
short: S. Windmann, R. Haeb-Umbach, IEEE Transactions on Audio, Speech, and Language
Processing 17 (2009) 1577–1590.
date_created: 2019-07-12T05:31:09Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/TASL.2009.2023172
intvolume: ' 17'
issue: '8'
keyword:
- AURORA4 database
- blockwise EM algorithm
- covariance analysis
- linear state model
- noise covariance
- noise-robust automatic speech recognition
- noisy speech cepstra
- offline training mode
- parameter estimation
- speech recognition
- speech recognition equipment
- speech recognizer
- state-space methods
- state-space model
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2009/WiHa09-2.pdf
oa: '1'
page: 1577-1590
publication: IEEE Transactions on Audio, Speech, and Language Processing
status: public
title: Parameter Estimation of a State-Space Model of Noise for Robust Speech Recognition
type: journal_article
user_id: '44006'
volume: 17
year: '2009'
...
---
_id: '17272'
abstract:
- lang: eng
text: In developmental research, tutoring behavior has been identified as scaffolding
infants' learning processes. It has been defined in terms of child-directed speech
(Motherese), child-directed motion (Motionese), and contingency. In the field
of developmental robotics, research often assumes that in human-robot interaction
(HRI), robots are treated similar to infants, because their immature cognitive
capabilities benefit from this behavior. However, according to our knowledge,
it has barely been studied whether this is true and how exactly humans alter their
behavior towards a robotic interaction partner. In this paper, we present results
concerning the acceptance of a robotic agent in a social learning scenario obtained
via comparison to adults and 8-11 months old infants in equal conditions. These
results constitute an important empirical basis for making use of tutoring behavior
in social robotics. In our study, we performed a detailed multimodal analysis
of HRI in a tutoring situation using the example of a robot simulation equipped
with a bottom-up saliency-based attention model. Our results reveal significant
differences in hand movement velocity, motion pauses, range of motion, and eye
gaze suggesting that for example adults decrease their hand movement velocity
in an Adult-Child Interaction (ACI), opposed to an Adult-Adult Interaction (AAI)
and this decrease is even higher in the Adult-Robot Interaction (ARI). We also
found important differences between ACI and ARI in how the behavior is modified
over time as the interaction unfolds. These findings indicate the necessity of
integrating top-down feedback structures into a bottom-up system for robots to
be fully accepted as interaction partners.
author:
- first_name: Anna-Lisa
full_name: Vollmer, Anna-Lisa
last_name: Vollmer
- first_name: Katrin Solveig
full_name: Lohan, Katrin Solveig
last_name: Lohan
- first_name: Kerstin
full_name: Fischer, Kerstin
last_name: Fischer
- first_name: Yukie
full_name: Nagai, Yukie
last_name: Nagai
- first_name: Karola
full_name: Pitsch, Karola
last_name: Pitsch
- first_name: Jannik
full_name: Fritsch, Jannik
last_name: Fritsch
- first_name: Katharina
full_name: Rohlfing, Katharina
id: '50352'
last_name: Rohlfing
- first_name: Britta
full_name: Wrede, Britta
last_name: Wrede
citation:
ama: 'Vollmer A-L, Lohan KS, Fischer K, et al. People modify their tutoring behavior
in robot-directed interaction for action learning. In: Development and Learning,
2009. ICDL 2009. IEEE 8th International Conference on Development and Learning.
IEEE; 2009:1-6. doi:10.1109/DEVLRN.2009.5175516'
apa: Vollmer, A.-L., Lohan, K. S., Fischer, K., Nagai, Y., Pitsch, K., Fritsch,
J., Rohlfing, K., & Wrede, B. (2009). People modify their tutoring behavior
in robot-directed interaction for action learning. Development and Learning,
2009. ICDL 2009. IEEE 8th International Conference on Development and Learning,
1–6. https://doi.org/10.1109/DEVLRN.2009.5175516
bibtex: '@inproceedings{Vollmer_Lohan_Fischer_Nagai_Pitsch_Fritsch_Rohlfing_Wrede_2009,
title={People modify their tutoring behavior in robot-directed interaction for
action learning}, DOI={10.1109/DEVLRN.2009.5175516},
booktitle={Development and Learning, 2009. ICDL 2009. IEEE 8th International Conference
on Development and Learning}, publisher={IEEE}, author={Vollmer, Anna-Lisa and
Lohan, Katrin Solveig and Fischer, Kerstin and Nagai, Yukie and Pitsch, Karola
and Fritsch, Jannik and Rohlfing, Katharina and Wrede, Britta}, year={2009}, pages={1–6}
}'
chicago: Vollmer, Anna-Lisa, Katrin Solveig Lohan, Kerstin Fischer, Yukie Nagai,
Karola Pitsch, Jannik Fritsch, Katharina Rohlfing, and Britta Wrede. “People Modify
Their Tutoring Behavior in Robot-Directed Interaction for Action Learning.” In
Development and Learning, 2009. ICDL 2009. IEEE 8th International Conference
on Development and Learning, 1–6. IEEE, 2009. https://doi.org/10.1109/DEVLRN.2009.5175516.
ieee: 'A.-L. Vollmer et al., “People modify their tutoring behavior in robot-directed
interaction for action learning,” in Development and Learning, 2009. ICDL 2009.
IEEE 8th International Conference on Development and Learning, 2009, pp. 1–6,
doi: 10.1109/DEVLRN.2009.5175516.'
mla: Vollmer, Anna-Lisa, et al. “People Modify Their Tutoring Behavior in Robot-Directed
Interaction for Action Learning.” Development and Learning, 2009. ICDL 2009.
IEEE 8th International Conference on Development and Learning, IEEE, 2009,
pp. 1–6, doi:10.1109/DEVLRN.2009.5175516.
short: 'A.-L. Vollmer, K.S. Lohan, K. Fischer, Y. Nagai, K. Pitsch, J. Fritsch,
K. Rohlfing, B. Wrede, in: Development and Learning, 2009. ICDL 2009. IEEE 8th
International Conference on Development and Learning, IEEE, 2009, pp. 1–6.'
date_created: 2020-06-24T13:02:43Z
date_updated: 2023-02-01T13:06:43Z
department:
- _id: '749'
doi: 10.1109/DEVLRN.2009.5175516
keyword:
- robot simulation
- hand movement velocity
- robotic interaction partner
- robotic agent
- robot-directed interaction
- multimodal analysis
- Motionese
- Motherese
- intelligent tutoring systems
- immature cognitive capability
- human computer interaction
- eye gaze
- child-directed speech
- child-directed motion
- bottom-up system
- bottom-up saliency-based attention model
- adult-robot interaction
- adult-child interaction
- adult-adult interaction
- human-robot interaction
- action learning
- social learning scenario
- social robotics
- software agents
- top-down feedback structures
- tutoring behavior
language:
- iso: eng
page: 1-6
publication: Development and Learning, 2009. ICDL 2009. IEEE 8th International Conference
on Development and Learning
publisher: IEEE
status: public
title: People modify their tutoring behavior in robot-directed interaction for action
learning
type: conference
user_id: '14931'
year: '2009'
...
---
_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. IEEE Transactions on Audio, Speech,
and Language Processing. 2008;16(5):1047-1060. doi:10.1109/TASL.2008.925879
apa: Ion, V., & Haeb-Umbach, R. (2008). A Novel Uncertainty Decoding Rule With
Applications to Transmission Error Robust Speech Recognition. IEEE Transactions
on Audio, Speech, and Language Processing, 16(5), 1047–1060. https://doi.org/10.1109/TASL.2008.925879
bibtex: '@article{Ion_Haeb-Umbach_2008, title={A Novel Uncertainty Decoding Rule
With Applications to Transmission Error Robust Speech Recognition}, volume={16},
DOI={10.1109/TASL.2008.925879},
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.” IEEE
Transactions on Audio, Speech, and Language Processing 16, no. 5 (2008): 1047–60.
https://doi.org/10.1109/TASL.2008.925879.'
ieee: V. Ion and R. Haeb-Umbach, “A Novel Uncertainty Decoding Rule With Applications
to Transmission Error Robust Speech Recognition,” IEEE Transactions on Audio,
Speech, and Language Processing, 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.” IEEE Transactions
on Audio, Speech, and Language Processing, vol. 16, no. 5, 2008, pp. 1047–60,
doi:10.1109/TASL.2008.925879.
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: '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: IEEE International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2008). ; 2008:73-76. doi:10.1109/ICASSP.2008.4517549'
apa: Warsitz, E., Krueger, A., & Haeb-Umbach, R. (2008). Speech enhancement
with a new generalized eigenvector blocking matrix for application in a generalized
sidelobe canceller. In IEEE International Conference on Acoustics, Speech and
Signal Processing (ICASSP 2008) (pp. 73–76). https://doi.org/10.1109/ICASSP.2008.4517549
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={10.1109/ICASSP.2008.4517549},
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 IEEE International Conference on Acoustics, Speech
and Signal Processing (ICASSP 2008), 73–76, 2008. https://doi.org/10.1109/ICASSP.2008.4517549.
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 IEEE International Conference on Acoustics, Speech and Signal
Processing (ICASSP 2008), 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.” IEEE
International Conference on Acoustics, Speech and Signal Processing (ICASSP 2008),
2008, pp. 73–76, doi:10.1109/ICASSP.2008.4517549.
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: IEEE International Conference on Acoustics,
Speech and Signal Processing (ICASSP 2008). ; 2008:4409-4412. doi:10.1109/ICASSP.2008.4518633'
apa: Windmann, S., & Haeb-Umbach, R. (2008). Modeling the dynamics of speech
and noise for speech feature enhancement in ASR. In IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2008) (pp. 4409–4412).
https://doi.org/10.1109/ICASSP.2008.4518633
bibtex: '@inproceedings{Windmann_Haeb-Umbach_2008, title={Modeling the dynamics
of speech and noise for speech feature enhancement in ASR}, DOI={10.1109/ICASSP.2008.4518633},
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 IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2008), 4409–12, 2008. https://doi.org/10.1109/ICASSP.2008.4518633.
ieee: S. Windmann and R. Haeb-Umbach, “Modeling the dynamics of speech and noise
for speech feature enhancement in ASR,” in IEEE International Conference on
Acoustics, Speech and Signal Processing (ICASSP 2008), 2008, pp. 4409–4412.
mla: Windmann, Stefan, and Reinhold Haeb-Umbach. “Modeling the Dynamics of Speech
and Noise for Speech Feature Enhancement in ASR.” IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2008), 2008, pp. 4409–12,
doi:10.1109/ICASSP.2008.4518633.
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: '17278'
abstract:
- lang: eng
text: This paper investigates the influence of feedback provided by an autonomous
robot (BIRON) on users’ discursive behavior. A user study is described during
which users show objects to the robot. The results of the experiment indicate,
that the robot’s verbal feedback utterances cause the humans to adapt their own
way of speaking. The changes in users’ verbal behavior are due to their beliefs
about the robots knowledge and abilities. In this paper they are identified and
grouped. Moreover, the data implies variations in user behavior regarding gestures.
Unlike speech, the robot was not able to give feedback with gestures. Due to the
lack of feedback, users did not seem to have a consistent mental representation
of the robot’s abilities to recognize gestures. As a result, changes between different
gestures are interpreted to be unconscious variations accompanying speech.
author:
- first_name: Manja
full_name: Lohse, Manja
last_name: Lohse
- first_name: Katharina
full_name: Rohlfing, Katharina
id: '50352'
last_name: Rohlfing
- first_name: Britta
full_name: Wrede, Britta
last_name: Wrede
- first_name: Gerhard
full_name: Sagerer, Gerhard
last_name: Sagerer
citation:
ama: 'Lohse M, Rohlfing K, Wrede B, Sagerer G. “Try something else!” — When users
change their discursive behavior in human-robot interaction. In: ; 2008:3481-3486.
doi:10.1109/ROBOT.2008.4543743'
apa: Lohse, M., Rohlfing, K., Wrede, B., & Sagerer, G. (2008). “Try something
else!” — When users change their discursive behavior in human-robot interaction.
3481–3486. https://doi.org/10.1109/ROBOT.2008.4543743
bibtex: '@inproceedings{Lohse_Rohlfing_Wrede_Sagerer_2008, title={“Try something
else!” — When users change their discursive behavior in human-robot interaction},
DOI={10.1109/ROBOT.2008.4543743},
author={Lohse, Manja and Rohlfing, Katharina and Wrede, Britta and Sagerer, Gerhard},
year={2008}, pages={3481–3486} }'
chicago: Lohse, Manja, Katharina Rohlfing, Britta Wrede, and Gerhard Sagerer. “‘Try
Something Else!’ — When Users Change Their Discursive Behavior in Human-Robot
Interaction,” 3481–86, 2008. https://doi.org/10.1109/ROBOT.2008.4543743.
ieee: 'M. Lohse, K. Rohlfing, B. Wrede, and G. Sagerer, “‘Try something else!’ —
When users change their discursive behavior in human-robot interaction,” 2008,
pp. 3481–3486, doi: 10.1109/ROBOT.2008.4543743.'
mla: Lohse, Manja, et al. “Try Something Else!” — When Users Change Their Discursive
Behavior in Human-Robot Interaction. 2008, pp. 3481–86, doi:10.1109/ROBOT.2008.4543743.
short: 'M. Lohse, K. Rohlfing, B. Wrede, G. Sagerer, in: 2008, pp. 3481–3486.'
date_created: 2020-06-24T13:02:49Z
date_updated: 2023-02-01T13:08:20Z
department:
- _id: '749'
doi: 10.1109/ROBOT.2008.4543743
keyword:
- discursive behavior
- autonomous robot
- BIRON
- man-machine systems
- robot abilities
- robot knowledge
- user gestures
- robot verbal feedback utterance
- speech processing
- user verbal behavior
- service robots
- human-robot interaction
- human computer interaction
- gesture recognition
language:
- iso: eng
page: 3481-3486
publication_identifier:
isbn:
- 1050-4729
status: public
title: “Try something else!” — When users change their discursive behavior in human-robot
interaction
type: conference
user_id: '14931'
year: '2008'
...