---
_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: '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'
...
---
_id: '11824'
abstract:
- lang: eng
text: Soft-feature based speech recognition, which is an example of uncertainty
decoding, has been proven to be a robust error mitigation method for distributed
speech recognition over wireless channels exhibiting bit errors. In this paper
we extend this concept to packet-oriented transmissions. The a posteriori probability
density function of the lost feature vector, given the closest received neighbours,
is computed. In the experiments, the nearest frame repetition, which is shown
to be equivalent to the MAP estimate, outperforms the MMSE estimate for long bursts.
Taking the variance into account at the speech recognition stage results in superior
performance compared to classical schemes using point estimates. A computationally
and memory efficient implementation of the proposed packet loss compensation scheme
based on table lookup is presented
author:
- first_name: Valentin
full_name: Ion, Valentin
last_name: Ion
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: 'Ion V, Haeb-Umbach R. An Inexpensive Packet Loss Compensation Scheme for Distributed
Speech Recognition Based on Soft-Features. In: IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2006). Vol 1. ; 2006:I.
doi:10.1109/ICASSP.2006.1659984'
apa: Ion, V., & Haeb-Umbach, R. (2006). An Inexpensive Packet Loss Compensation
Scheme for Distributed Speech Recognition Based on Soft-Features. In IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2006) (Vol.
1, p. I). https://doi.org/10.1109/ICASSP.2006.1659984
bibtex: '@inproceedings{Ion_Haeb-Umbach_2006, title={An Inexpensive Packet Loss
Compensation Scheme for Distributed Speech Recognition Based on Soft-Features},
volume={1}, DOI={10.1109/ICASSP.2006.1659984},
booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2006)}, author={Ion, Valentin and Haeb-Umbach, Reinhold}, year={2006},
pages={I} }'
chicago: Ion, Valentin, and Reinhold Haeb-Umbach. “An Inexpensive Packet Loss Compensation
Scheme for Distributed Speech Recognition Based on Soft-Features.” In IEEE
International Conference on Acoustics, Speech and Signal Processing (ICASSP 2006),
1:I, 2006. https://doi.org/10.1109/ICASSP.2006.1659984.
ieee: V. Ion and R. Haeb-Umbach, “An Inexpensive Packet Loss Compensation Scheme
for Distributed Speech Recognition Based on Soft-Features,” in IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2006), 2006,
vol. 1, p. I.
mla: Ion, Valentin, and Reinhold Haeb-Umbach. “An Inexpensive Packet Loss Compensation
Scheme for Distributed Speech Recognition Based on Soft-Features.” IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2006), vol.
1, 2006, p. I, doi:10.1109/ICASSP.2006.1659984.
short: 'V. Ion, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
Speech and Signal Processing (ICASSP 2006), 2006, p. I.'
date_created: 2019-07-12T05:28:58Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2006.1659984
intvolume: ' 1'
keyword:
- distributed speech recognition
- least mean squares methods
- MAP estimate
- maximum likelihood estimation
- MMSE estimate
- packet loss compensation scheme
- packet switched communication
- posteriori probability density function
- robust error mitigation method
- soft-features
- speech recognition
- table lookup
- voice communication
- wireless channels
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2006/IoHa06-2.pdf
oa: '1'
page: I
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2006)
status: public
title: An Inexpensive Packet Loss Compensation Scheme for Distributed Speech Recognition
Based on Soft-Features
type: conference
user_id: '44006'
volume: 1
year: '2006'
...
---
_id: '11825'
abstract:
- lang: eng
text: In this paper, we propose an enhanced error concealment strategy at the server
side of a distributed speech recognition (DSR) system, which is fully compatible
with the existing DSR standard. It is based on a Bayesian approach, where the
a posteriori probability density of the error-free feature vector is computed,
given all received feature vectors which are possibly corrupted by transmission
errors. Rather than computing a point estimate, such as the MMSE estimate, and
plugging it into the Bayesian decision rule, we employ uncertainty decoding, which
results in an integration over the uncertainty in the feature domain. In a typical
scenario the communication between the thin client, often a mobile device, and
the recognition server spreads across heterogeneous networks. Both bit errors
on circuit-switched links and lost data packets on IP connections are mitigated
by our approach in a unified manner. The experiments reveal improved robustness
both for small- and large-vocabulary recognition tasks.
author:
- first_name: Valentin
full_name: Ion, Valentin
last_name: Ion
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: Ion V, Haeb-Umbach R. Uncertainty decoding for distributed speech recognition
over error-prone networks. Speech Communication. 2006;48(11):1435-1446.
doi:10.1016/j.specom.2006.03.007
apa: Ion, V., & Haeb-Umbach, R. (2006). Uncertainty decoding for distributed
speech recognition over error-prone networks. Speech Communication, 48(11),
1435–1446. https://doi.org/10.1016/j.specom.2006.03.007
bibtex: '@article{Ion_Haeb-Umbach_2006, title={Uncertainty decoding for distributed
speech recognition over error-prone networks}, volume={48}, DOI={10.1016/j.specom.2006.03.007},
number={11}, journal={Speech Communication}, author={Ion, Valentin and Haeb-Umbach,
Reinhold}, year={2006}, pages={1435–1446} }'
chicago: 'Ion, Valentin, and Reinhold Haeb-Umbach. “Uncertainty Decoding for Distributed
Speech Recognition over Error-Prone Networks.” Speech Communication 48,
no. 11 (2006): 1435–46. https://doi.org/10.1016/j.specom.2006.03.007.'
ieee: V. Ion and R. Haeb-Umbach, “Uncertainty decoding for distributed speech recognition
over error-prone networks,” Speech Communication, vol. 48, no. 11, pp.
1435–1446, 2006.
mla: Ion, Valentin, and Reinhold Haeb-Umbach. “Uncertainty Decoding for Distributed
Speech Recognition over Error-Prone Networks.” Speech Communication, vol.
48, no. 11, 2006, pp. 1435–46, doi:10.1016/j.specom.2006.03.007.
short: V. Ion, R. Haeb-Umbach, Speech Communication 48 (2006) 1435–1446.
date_created: 2019-07-12T05:28:59Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
doi: 10.1016/j.specom.2006.03.007
intvolume: ' 48'
issue: '11'
keyword:
- Channel error robustness
- Distributed speech recognition
- Soft features
- Uncertainty decoding
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2006/IoHa06-3.pdf
oa: '1'
page: 1435-1446
publication: Speech Communication
status: public
title: Uncertainty decoding for distributed speech recognition over error-prone networks
type: journal_article
user_id: '44006'
volume: 48
year: '2006'
...
---
_id: '11943'
abstract:
- lang: eng
text: A marginalized particle filter is proposed for performing single channel speech
enhancement with a non-linear dynamic state model. The system consists of a particle
filter for tracking line spectral pair (LSP) parameters and a Kalman filter per
particle for speech enhancement. The state model for the LSPs has been learnt
on clean speech training data. In our approach parameters and speech samples are
processed at different time scales by assuming the parameters to be constant for
small blocks of data. Further enhancement is obtained by an iteration which can
be applied on these small blocks. The experiments show that similar SNR gains
are obtained as with the Kalman-LM-iterative algorithm. However better values
of the noise level and the log-spectral distance are achieved
author:
- first_name: Stefan
full_name: Windmann, Stefan
last_name: Windmann
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: 'Windmann S, Haeb-Umbach R. Iterative Speech Enhancement using a Non-Linear
Dynamic State Model of Speech and its Parameters. In: IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2006). Vol 1. ; 2006:I.
doi:10.1109/ICASSP.2006.1660058'
apa: Windmann, S., & Haeb-Umbach, R. (2006). Iterative Speech Enhancement using
a Non-Linear Dynamic State Model of Speech and its Parameters. In IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2006) (Vol.
1, p. I). https://doi.org/10.1109/ICASSP.2006.1660058
bibtex: '@inproceedings{Windmann_Haeb-Umbach_2006, title={Iterative Speech Enhancement
using a Non-Linear Dynamic State Model of Speech and its Parameters}, volume={1},
DOI={10.1109/ICASSP.2006.1660058},
booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2006)}, author={Windmann, Stefan and Haeb-Umbach, Reinhold}, year={2006},
pages={I} }'
chicago: Windmann, Stefan, and Reinhold Haeb-Umbach. “Iterative Speech Enhancement
Using a Non-Linear Dynamic State Model of Speech and Its Parameters.” In IEEE
International Conference on Acoustics, Speech and Signal Processing (ICASSP 2006),
1:I, 2006. https://doi.org/10.1109/ICASSP.2006.1660058.
ieee: S. Windmann and R. Haeb-Umbach, “Iterative Speech Enhancement using a Non-Linear
Dynamic State Model of Speech and its Parameters,” in IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP 2006), 2006, vol. 1, p.
I.
mla: Windmann, Stefan, and Reinhold Haeb-Umbach. “Iterative Speech Enhancement Using
a Non-Linear Dynamic State Model of Speech and Its Parameters.” IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2006), vol.
1, 2006, p. I, doi:10.1109/ICASSP.2006.1660058.
short: 'S. Windmann, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
Speech and Signal Processing (ICASSP 2006), 2006, p. I.'
date_created: 2019-07-12T05:31:15Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2006.1660058
intvolume: ' 1'
keyword:
- clean speech training data
- iterative methods
- iterative speech enhancement
- Kalman filter
- Kalman filters
- Kalman-LM-iterative algorithm
- line spectral pair parameters
- log-spectral distance
- marginalized particle filter
- noise level
- nonlinear dynamic state speech model
- particle filtering (numerical methods)
- single channel speech enhancement
- SNR gains
- speech enhancement
- speech samples
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2006/WiHa06-2.pdf
oa: '1'
page: I
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2006)
status: public
title: Iterative Speech Enhancement using a Non-Linear Dynamic State Model of Speech
and its Parameters
type: conference
user_id: '44006'
volume: 1
year: '2006'
...
---
_id: '11828'
abstract:
- lang: eng
text: 'In this paper we present a comparison of the recently proposed Soft-Feature
Distributed Speech Recognition (SFDSR) with the two evaluated candidate codecs
for Speech Enabled Services over wireless networks: Adaptive Multirate Codec (AMR)
and the ETSI Extended Advanced Front-End for Distributed Speech Recognition (XAFE).
It is shown that SFDSR achieves the best recognition performance on a simulated
GSM transmission, followed by XAFE and AMR.We also present some new results concerning
SFDSR which demonstrate the versatility of the approach. Further, a simple method
is introduced which considerably reduces the computational effort.'
author:
- first_name: Valentin
full_name: Ion, Valentin
last_name: Ion
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: 'Ion V, Haeb-Umbach R. A Comparison of Soft-Feature Distributed Speech Recognition
with Candidate Codecs for Speech Enabled Mobile Services. In: IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP 2005). Vol 1.
; 2005:333-336. doi:10.1109/ICASSP.2005.1415118'
apa: Ion, V., & Haeb-Umbach, R. (2005). A Comparison of Soft-Feature Distributed
Speech Recognition with Candidate Codecs for Speech Enabled Mobile Services. In
IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP
2005) (Vol. 1, pp. 333–336). https://doi.org/10.1109/ICASSP.2005.1415118
bibtex: '@inproceedings{Ion_Haeb-Umbach_2005, title={A Comparison of Soft-Feature
Distributed Speech Recognition with Candidate Codecs for Speech Enabled Mobile
Services}, volume={1}, DOI={10.1109/ICASSP.2005.1415118},
booktitle={IEEE International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2005)}, author={Ion, Valentin and Haeb-Umbach, Reinhold}, year={2005},
pages={333–336} }'
chicago: Ion, Valentin, and Reinhold Haeb-Umbach. “A Comparison of Soft-Feature
Distributed Speech Recognition with Candidate Codecs for Speech Enabled Mobile
Services.” In IEEE International Conference on Acoustics, Speech and Signal
Processing (ICASSP 2005), 1:333–36, 2005. https://doi.org/10.1109/ICASSP.2005.1415118.
ieee: V. Ion and R. Haeb-Umbach, “A Comparison of Soft-Feature Distributed Speech
Recognition with Candidate Codecs for Speech Enabled Mobile Services,” in IEEE
International Conference on Acoustics, Speech and Signal Processing (ICASSP 2005),
2005, vol. 1, pp. 333–336.
mla: Ion, Valentin, and Reinhold Haeb-Umbach. “A Comparison of Soft-Feature Distributed
Speech Recognition with Candidate Codecs for Speech Enabled Mobile Services.”
IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP
2005), vol. 1, 2005, pp. 333–36, doi:10.1109/ICASSP.2005.1415118.
short: 'V. Ion, R. Haeb-Umbach, in: IEEE International Conference on Acoustics,
Speech and Signal Processing (ICASSP 2005), 2005, pp. 333–336.'
date_created: 2019-07-12T05:29:02Z
date_updated: 2022-01-06T06:51:10Z
department:
- _id: '54'
doi: 10.1109/ICASSP.2005.1415118
intvolume: ' 1'
keyword:
- adaptive codes
- adaptive multirate codec
- AMR
- distributed speech recognition
- ETSI
- extended advanced front-end
- recognition performance
- SFDSR
- simulated GSM transmission
- soft-feature distributed speech recognition
- speech codecs
- speech coding
- speech recognition
- variable rate codes
- XAFE
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2005/IoHa05-2.pdf
oa: '1'
page: 333-336
publication: IEEE International Conference on Acoustics, Speech and Signal Processing
(ICASSP 2005)
status: public
title: A Comparison of Soft-Feature Distributed Speech Recognition with Candidate
Codecs for Speech Enabled Mobile Services
type: conference
user_id: '44006'
volume: 1
year: '2005'
...
---
_id: '11931'
abstract:
- lang: eng
text: The paper is concerned with binaural signal processing for a bimodal human-robot
interface with hearing and vision. The two microphone signals are processed to
obtain an enhanced single-channel input signal for the subsequent speech recognizer
and to localize the acoustic source, an important information for establishing
a natural human-robot communication. We utilize a robust adaptive algorithm for
filter-and-sum beamforming (FSB) and extract speaker direction information from
the resulting FIR filter coefficients. Further, particle filtering is applied
which conducts a nonlinear Bayesian tracking of speaker movement. Good location
accuracy can be achieved even in highly reverberant environments. The results
obtained outperform the conventional generalized cross correlation (GCC) method.
author:
- first_name: Ernst
full_name: Warsitz, Ernst
last_name: Warsitz
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: 'Warsitz E, Haeb-Umbach R. Robust speaker direction estimation with particle
filtering. In: IEEE Workshop on Multimedia Signal Processing (MMSP 2004).
; 2004:367-370. doi:10.1109/MMSP.2004.1436569'
apa: Warsitz, E., & Haeb-Umbach, R. (2004). Robust speaker direction estimation
with particle filtering. In IEEE Workshop on Multimedia Signal Processing (MMSP
2004) (pp. 367–370). https://doi.org/10.1109/MMSP.2004.1436569
bibtex: '@inproceedings{Warsitz_Haeb-Umbach_2004, title={Robust speaker direction
estimation with particle filtering}, DOI={10.1109/MMSP.2004.1436569},
booktitle={IEEE Workshop on Multimedia Signal Processing (MMSP 2004)}, author={Warsitz,
Ernst and Haeb-Umbach, Reinhold}, year={2004}, pages={367–370} }'
chicago: Warsitz, Ernst, and Reinhold Haeb-Umbach. “Robust Speaker Direction Estimation
with Particle Filtering.” In IEEE Workshop on Multimedia Signal Processing
(MMSP 2004), 367–70, 2004. https://doi.org/10.1109/MMSP.2004.1436569.
ieee: E. Warsitz and R. Haeb-Umbach, “Robust speaker direction estimation with particle
filtering,” in IEEE Workshop on Multimedia Signal Processing (MMSP 2004),
2004, pp. 367–370.
mla: Warsitz, Ernst, and Reinhold Haeb-Umbach. “Robust Speaker Direction Estimation
with Particle Filtering.” IEEE Workshop on Multimedia Signal Processing (MMSP
2004), 2004, pp. 367–70, doi:10.1109/MMSP.2004.1436569.
short: 'E. Warsitz, R. Haeb-Umbach, in: IEEE Workshop on Multimedia Signal Processing
(MMSP 2004), 2004, pp. 367–370.'
date_created: 2019-07-12T05:31:01Z
date_updated: 2022-01-06T06:51:12Z
department:
- _id: '54'
doi: 10.1109/MMSP.2004.1436569
keyword:
- bimodal human-robot interface
- binaural signal processing
- enhanced single-channel input signal
- filter-and-sum beamforming
- filtering theory
- FIR filter coefficient
- generalized cross correlation method
- microphones
- microphone signal
- nonlinear Bayesian tracking
- particle filtering
- robust adaptive algorithm
- robust speaker direction estimation
- signal processing
- speech enhancement
- speech recognition
- speech recognizer
- user interfaces
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2004/WaHa04.pdf
oa: '1'
page: 367-370
publication: IEEE Workshop on Multimedia Signal Processing (MMSP 2004)
status: public
title: Robust speaker direction estimation with particle filtering
type: conference
user_id: '44006'
year: '2004'
...
---
_id: '39053'
abstract:
- lang: eng
text: Portable devices come with different limitations in user interaction like
limited display size, small keyboard, and different sorts of input and output
capabilities. With the advance of speech recognition and speech synthesis technologies,
their complementary use becomes attractive for mobile devices in order to implement
real multimodal user interaction. However, current systems and formats do not
sufficiently integrate advanced multimodal interactions. We introduce an advanced
generic multimodal interaction and rendering system (MIRS) dedicated for mobile
devices. MIRS incorporates efficient processing of XML specification languages
for limited, mobile devices and comes with the XML-based dialog and interface
specification language (DISL). DISL can be considered as an UIML subset, which
is enhanced by the means of state-oriented dialog specifications. The dialog specification
is based on ODSN (object oriented dialog specification notation), which has been
introduced to define user interface control by means of interaction states with
transition rules.
author:
- first_name: Wolfgang
full_name: Müller, Wolfgang
id: '16243'
last_name: Müller
- first_name: Robbie
full_name: Schäfer, Robbie
last_name: Schäfer
- first_name: Steffen
full_name: Bleul, Steffen
last_name: Bleul
citation:
ama: 'Müller W, Schäfer R, Bleul S. Interactive Multimodal User Interfaces for Mobile
Devices. In: Proceedings of HICCS-37. ; 2004. doi:10.1109/HICSS.2004.1265674'
apa: Müller, W., Schäfer, R., & Bleul, S. (2004). Interactive Multimodal User
Interfaces for Mobile Devices. Proceedings of HICCS-37. 37th Annual Hawaii
International Conference on System Sciences, Waikoloa, HI, USA. https://doi.org/10.1109/HICSS.2004.1265674
bibtex: '@inproceedings{Müller_Schäfer_Bleul_2004, place={Waikoloa, HI, USA}, title={Interactive
Multimodal User Interfaces for Mobile Devices}, DOI={10.1109/HICSS.2004.1265674},
booktitle={Proceedings of HICCS-37}, author={Müller, Wolfgang and Schäfer, Robbie
and Bleul, Steffen}, year={2004} }'
chicago: Müller, Wolfgang, Robbie Schäfer, and Steffen Bleul. “Interactive Multimodal
User Interfaces for Mobile Devices.” In Proceedings of HICCS-37. Waikoloa,
HI, USA, 2004. https://doi.org/10.1109/HICSS.2004.1265674.
ieee: 'W. Müller, R. Schäfer, and S. Bleul, “Interactive Multimodal User Interfaces
for Mobile Devices,” presented at the 37th Annual Hawaii International Conference
on System Sciences, Waikoloa, HI, USA, 2004, doi: 10.1109/HICSS.2004.1265674.'
mla: Müller, Wolfgang, et al. “Interactive Multimodal User Interfaces for Mobile
Devices.” Proceedings of HICCS-37, 2004, doi:10.1109/HICSS.2004.1265674.
short: 'W. Müller, R. Schäfer, S. Bleul, in: Proceedings of HICCS-37, Waikoloa,
HI, USA, 2004.'
conference:
location: Waikoloa, HI, USA
name: 37th Annual Hawaii International Conference on System Sciences
date_created: 2023-01-24T08:46:31Z
date_updated: 2023-01-24T08:46:37Z
department:
- _id: '672'
doi: 10.1109/HICSS.2004.1265674
keyword:
- User interfaces
- Speech recognition
- Streaming media
- Specification languages
- Keyboards
- Speech synthesis
- Rendering (computer graphics)
- Ambient intelligence
- Humans
- Displays
language:
- iso: eng
place: Waikoloa, HI, USA
publication: Proceedings of HICCS-37
publication_identifier:
isbn:
- 0-7695-2056-1
status: public
title: Interactive Multimodal User Interfaces for Mobile Devices
type: conference
user_id: '5786'
year: '2004'
...
---
_id: '11778'
abstract:
- lang: eng
text: In this paper, it is shown that a correlation criterion is the appropriate
criterion for bottom-up clustering to obtain broad phonetic class regression trees
for maximum likelihood linear regression (MLLR)-based speaker adaptation. The
correlation structure among speech units is estimated on the speaker-independent
training data. In adaptation experiments the tree outperformed a regression tree
obtained from clustering according to closeness in acoustic space and achieved
results comparable with those of a manually designed broad phonetic class tree
author:
- first_name: Reinhold
full_name: Haeb-Umbach, Reinhold
id: '242'
last_name: Haeb-Umbach
citation:
ama: Haeb-Umbach R. Automatic generation of phonetic regression class trees for
MLLR adaptation. IEEE Transactions on Speech and Audio Processing. 2001;9(3):299-302.
doi:10.1109/89.906003
apa: Haeb-Umbach, R. (2001). Automatic generation of phonetic regression class trees
for MLLR adaptation. IEEE Transactions on Speech and Audio Processing,
9(3), 299–302. https://doi.org/10.1109/89.906003
bibtex: '@article{Haeb-Umbach_2001, title={Automatic generation of phonetic regression
class trees for MLLR adaptation}, volume={9}, DOI={10.1109/89.906003},
number={3}, journal={IEEE Transactions on Speech and Audio Processing}, author={Haeb-Umbach,
Reinhold}, year={2001}, pages={299–302} }'
chicago: 'Haeb-Umbach, Reinhold. “Automatic Generation of Phonetic Regression Class
Trees for MLLR Adaptation.” IEEE Transactions on Speech and Audio Processing
9, no. 3 (2001): 299–302. https://doi.org/10.1109/89.906003.'
ieee: R. Haeb-Umbach, “Automatic generation of phonetic regression class trees for
MLLR adaptation,” IEEE Transactions on Speech and Audio Processing, vol.
9, no. 3, pp. 299–302, 2001.
mla: Haeb-Umbach, Reinhold. “Automatic Generation of Phonetic Regression Class Trees
for MLLR Adaptation.” IEEE Transactions on Speech and Audio Processing,
vol. 9, no. 3, 2001, pp. 299–302, doi:10.1109/89.906003.
short: R. Haeb-Umbach, IEEE Transactions on Speech and Audio Processing 9 (2001)
299–302.
date_created: 2019-07-12T05:28:04Z
date_updated: 2022-01-06T06:51:08Z
department:
- _id: '54'
doi: 10.1109/89.906003
intvolume: ' 9'
issue: '3'
keyword:
- acoustic space
- adaptation experiments
- automatic generation
- bottom-up clustering
- broad phonetic class regression trees
- correlation criterion
- correlation methods
- maximum likelihood estimation
- maximum likelihood linear regression based speaker adaptation
- MLLR adaptation
- pattern clustering
- phonetic regression class trees
- speaker-independent training data
- speech recognition
- speech units
- statistical analysis
- trees (mathematics)
language:
- iso: eng
main_file_link:
- open_access: '1'
url: https://groups.uni-paderborn.de/nt/pubs/2001/Ha01.pdf
oa: '1'
page: 299-302
publication: IEEE Transactions on Speech and Audio Processing
status: public
title: Automatic generation of phonetic regression class trees for MLLR adaptation
type: journal_article
user_id: '44006'
volume: 9
year: '2001'
...
---
_id: '2433'
author:
- first_name: Christian
full_name: Plessl, Christian
id: '16153'
last_name: Plessl
orcid: 0000-0001-5728-9982
- first_name: Simon
full_name: Maurer, Simon
last_name: Maurer
citation:
ama: Plessl C, Maurer S. Hardware/Software Codesign in Speech Compression Applications.
Computer Engineering and Networks Lab, ETH Zurich, Switzerland; 2000.
apa: Plessl, C., & Maurer, S. (2000). Hardware/Software Codesign in Speech
Compression Applications. Computer Engineering and Networks Lab, ETH Zurich,
Switzerland.
bibtex: '@book{Plessl_Maurer_2000, title={Hardware/Software Codesign in Speech Compression
Applications}, publisher={Computer Engineering and Networks Lab, ETH Zurich, Switzerland},
author={Plessl, Christian and Maurer, Simon}, year={2000} }'
chicago: Plessl, Christian, and Simon Maurer. Hardware/Software Codesign in Speech
Compression Applications. Computer Engineering and Networks Lab, ETH Zurich,
Switzerland, 2000.
ieee: C. Plessl and S. Maurer, Hardware/Software Codesign in Speech Compression
Applications. Computer Engineering and Networks Lab, ETH Zurich, Switzerland,
2000.
mla: Plessl, Christian, and Simon Maurer. Hardware/Software Codesign in Speech
Compression Applications. Computer Engineering and Networks Lab, ETH Zurich,
Switzerland, 2000.
short: C. Plessl, S. Maurer, Hardware/Software Codesign in Speech Compression Applications,
Computer Engineering and Networks Lab, ETH Zurich, Switzerland, 2000.
date_created: 2018-04-17T15:56:00Z
date_updated: 2022-01-06T06:56:17Z
department:
- _id: '518'
keyword:
- co-design
- speech processing
publisher: Computer Engineering and Networks Lab, ETH Zurich, Switzerland
status: public
title: Hardware/Software Codesign in Speech Compression Applications
type: mastersthesis
user_id: '24135'
year: '2000'
...