---
_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'
...
---
_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'
...