---
_id: '43045'
abstract:
- lang: eng
text: The pressure fields generated by two simultaneous discharges have
not been investigated on any notable scale for the electrohydraulic impulse forming
method. In this study, the synchronicity of two discharges is ensured by the sequential
connection of two wires mounted in two spark gaps in a common volume of liquid.
The objective is to experimentally confirm the equilibrium of the energies evolved
in two spark gaps by means of pressure measurements. In addition, multipoint membrane
pressure gauges demonstrated the feasibility of easily recording detailed pressure
maps. Based on the membrane deformation mechanism and material strengthening under
static and impulse conditions, the processing procedure is further developed so
as to achieve better accuracy in the determination of pressure field parameters.
The practical equality of the pressure fields on the left and right halves of
the flat-loaded area confirms the equality of energies evolved in the two spark
gaps. The direct shock waves create zones with the most intensive loading. These
results provide a basis for the development of new electrohydraulic technologies
involving the application of two simultaneous discharges with equal energy and
pressure parameters.
article_number: '40'
author:
- first_name: Mykhaylo
full_name: Knyazyev, Mykhaylo
last_name: Knyazyev
- first_name: Maik
full_name: Holzmüller, Maik
id: '82645'
last_name: Holzmüller
- first_name: Werner
full_name: Homberg, Werner
id: '233'
last_name: Homberg
citation:
ama: Knyazyev M, Holzmüller M, Homberg W. Investigation of Pressure Fields Generated
by Two Simultaneous Discharges in Liquid Initiated by Wires. Journal of Manufacturing
and Materials Processing. 2023;7(1). doi:10.3390/jmmp7010040
apa: Knyazyev, M., Holzmüller, M., & Homberg, W. (2023). Investigation of Pressure
Fields Generated by Two Simultaneous Discharges in Liquid Initiated by Wires.
Journal of Manufacturing and Materials Processing, 7(1), Article
40. https://doi.org/10.3390/jmmp7010040
bibtex: '@article{Knyazyev_Holzmüller_Homberg_2023, title={Investigation of Pressure
Fields Generated by Two Simultaneous Discharges in Liquid Initiated by Wires},
volume={7}, DOI={10.3390/jmmp7010040},
number={140}, journal={Journal of Manufacturing and Materials Processing}, publisher={MDPI
AG}, author={Knyazyev, Mykhaylo and Holzmüller, Maik and Homberg, Werner}, year={2023}
}'
chicago: Knyazyev, Mykhaylo, Maik Holzmüller, and Werner Homberg. “Investigation
of Pressure Fields Generated by Two Simultaneous Discharges in Liquid Initiated
by Wires.” Journal of Manufacturing and Materials Processing 7, no. 1 (2023).
https://doi.org/10.3390/jmmp7010040.
ieee: 'M. Knyazyev, M. Holzmüller, and W. Homberg, “Investigation of Pressure Fields
Generated by Two Simultaneous Discharges in Liquid Initiated by Wires,” Journal
of Manufacturing and Materials Processing, vol. 7, no. 1, Art. no. 40, 2023,
doi: 10.3390/jmmp7010040.'
mla: Knyazyev, Mykhaylo, et al. “Investigation of Pressure Fields Generated by Two
Simultaneous Discharges in Liquid Initiated by Wires.” Journal of Manufacturing
and Materials Processing, vol. 7, no. 1, 40, MDPI AG, 2023, doi:10.3390/jmmp7010040.
short: M. Knyazyev, M. Holzmüller, W. Homberg, Journal of Manufacturing and Materials
Processing 7 (2023).
date_created: 2023-03-17T10:31:23Z
date_updated: 2023-05-02T11:40:35Z
department:
- _id: '156'
doi: 10.3390/jmmp7010040
intvolume: ' 7'
issue: '1'
keyword:
- impulse
- forming
- electrohydraulic
- discharge
- wire
- pressure gauge
- pressure field
language:
- iso: eng
publication: Journal of Manufacturing and Materials Processing
publication_identifier:
issn:
- 2504-4494
publication_status: published
publisher: MDPI AG
quality_controlled: '1'
status: public
title: Investigation of Pressure Fields Generated by Two Simultaneous Discharges in
Liquid Initiated by Wires
type: journal_article
user_id: '82645'
volume: 7
year: '2023'
...
---
_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: '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: IEEE International Conference on Acoustics,
Speech and Signal Processing (ICASSP 2007). Vol 3. ; 2007:III-277-III-280.
doi:10.1109/ICASSP.2007.366526'
apa: Haeb-Umbach, R., & Bevermeier, M. (2007). OFDM Channel Estimation Based
on Combined Estimation in Time and Frequency Domain. In IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2007) (Vol.
3, pp. III-277-III–280). https://doi.org/10.1109/ICASSP.2007.366526
bibtex: '@inproceedings{Haeb-Umbach_Bevermeier_2007, title={OFDM Channel Estimation
Based on Combined Estimation in Time and Frequency Domain}, volume={3}, DOI={10.1109/ICASSP.2007.366526},
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 IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2007), 3:III-277-III–280,
2007. https://doi.org/10.1109/ICASSP.2007.366526.
ieee: R. Haeb-Umbach and M. Bevermeier, “OFDM Channel Estimation Based on Combined
Estimation in Time and Frequency Domain,” in IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2007), 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.” IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2007), vol. 3, 2007, pp.
III-277-III–280, doi:10.1109/ICASSP.2007.366526.
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: '11869'
abstract:
- lang: eng
text: Amongst several data driven approaches for designing filters for the time
sequence of spectral parameters, the linear discriminant analysis (LDA) based
method has been proposed for automatic speech recognition. Here we apply LDA-based
filter design to cepstral features, which better match the inherent assumption
of this method that feature vector components are uncorrelated. Extensive recognition
experiments have been conducted both on the standard TIMIT phone recognition task
and on a proprietary 130-words command word task under various adverse environmental
conditions, including reverberant data with real-life room impulse responses and
data processed by acoustic echo cancellation algorithms. Significant error rate
reductions have been achieved when applying the novel long-range feature filters
compared to standard approaches employing cepstral mean normalization and delta
and delta-delta features, in particular when facing acoustic echo cancellation
scenarios and room reverberation. For example, the phone accuracy on reverberated
TIMIT data could be increased from 50.7\% to 56.0\%
author:
- first_name: M.
full_name: Lieb, M.
last_name: Lieb
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: 'Lieb M, Haeb-Umbach R. LDA derived cepstral trajectory filters in adverse
environmental conditions. In: IEEE International Conference on Acoustics, Speech,
and Signal Processing (ICASSP 2000). Vol 2. ; 2000:II1105-II1108 vol.2. doi:10.1109/ICASSP.2000.859157'
apa: Lieb, M., & Haeb-Umbach, R. (2000). LDA derived cepstral trajectory filters
in adverse environmental conditions. In IEEE International Conference on Acoustics,
Speech, and Signal Processing (ICASSP 2000) (Vol. 2, pp. II1105-II1108 vol.2).
https://doi.org/10.1109/ICASSP.2000.859157
bibtex: '@inproceedings{Lieb_Haeb-Umbach_2000, title={LDA derived cepstral trajectory
filters in adverse environmental conditions}, volume={2}, DOI={10.1109/ICASSP.2000.859157},
booktitle={IEEE International Conference on Acoustics, Speech, and Signal Processing
(ICASSP 2000)}, author={Lieb, M. and Haeb-Umbach, Reinhold}, year={2000}, pages={II1105-II1108
vol.2} }'
chicago: Lieb, M., and Reinhold Haeb-Umbach. “LDA Derived Cepstral Trajectory Filters
in Adverse Environmental Conditions.” In IEEE International Conference on Acoustics,
Speech, and Signal Processing (ICASSP 2000), 2:II1105-II1108 vol.2, 2000.
https://doi.org/10.1109/ICASSP.2000.859157.
ieee: M. Lieb and R. Haeb-Umbach, “LDA derived cepstral trajectory filters in adverse
environmental conditions,” in IEEE International Conference on Acoustics, Speech,
and Signal Processing (ICASSP 2000), 2000, vol. 2, pp. II1105-II1108 vol.2.
mla: Lieb, M., and Reinhold Haeb-Umbach. “LDA Derived Cepstral Trajectory Filters
in Adverse Environmental Conditions.” IEEE International Conference on Acoustics,
Speech, and Signal Processing (ICASSP 2000), vol. 2, 2000, pp. II1105-II1108
vol.2, doi:10.1109/ICASSP.2000.859157.
short: 'M. Lieb, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
Speech, and Signal Processing (ICASSP 2000), 2000, pp. II1105-II1108 vol.2.'
date_created: 2019-07-12T05:29:50Z
date_updated: 2022-01-06T06:51:11Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2000.859157
intvolume: ' 2'
keyword:
- acoustic echo cancellation algorithms
- adverse environmental conditions
- automatic speech recognition
- cepstral analysis
- cepstral features
- cepstral mean normalization
- command word task
- delta-delta features
- delta features
- echo suppression
- error rate reductions
- feature vector components
- FIR filters
- LDA derived cepstral trajectory filters
- linear discriminant analysis
- long-range feature filters
- phone accuracy
- real-life room impulse responses
- reverberant data
- spectral parameters
- speech recognition
- standard TIMIT phone recognition task
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2000/LiHa00.pdf
oa: '1'
page: II1105-II1108 vol.2
publication: IEEE International Conference on Acoustics, Speech, and Signal Processing
(ICASSP 2000)
status: public
title: LDA derived cepstral trajectory filters in adverse environmental conditions
type: conference
user_id: '44006'
volume: 2
year: '2000'
...