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Dennis Alexander Lehmann Thomsen

dblp:185/0469 · DBLP profile ↗
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6ranked-venue papers
0as first author
0since 2021 · last 2018
—ORCID · none

Domains — the database's venue-derived domains; a paper can count in several

Artificial intelligence and machine learning · 5Graphics, computer vision, multimedia, augmented reality and games · 5

Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.

Computer graphics and multimedia
1 paper
Audio and music processing · 100%

Topics — the 2 heaviest of 2, each with the papers that count most for it

TopicWeightPapersLastEvidence papers
Audio and music processing › speaker recognition
speaker verification
0.312018
Robust Voice Liveness Detection and Speaker Verification Using Throat Microphones · IEEE ACM Trans. Audio Speech Lang. Process. 2018
Audio and music processing › speaker recognition › speaker verification
spoofing detection
0.312018
Robust Voice Liveness Detection and Speaker Verification Using Throat Microphones · IEEE ACM Trans. Audio Speech Lang. Process. 2018

Methods — techniques the papers use, named apart from their topics

spectral features · 0.3i-vector · 0.3GMM-UBM · 0.3
YearPublicationVenuePosition
2018 Robust Voice Liveness Detection and Speaker Verification Using Throat Microphones
abstract
While having a wide range of applications, automatic speaker verification (ASV) systems are vulnerable to spoofing attacks, in particular, replay attacks that are effective and easy to implement. Most prior work on detecting replay attacks uses audio from a single acoustic microphone only, leading to difficulties in detecting high-end replay attacks close to indistinguishable from live human speech. In this paper, we study the use of a special body-conducted sensor, throat microphone (TM), for combined voice liveness detection (VLD) and ASV in order to improve both robustness and security of ASV against replay attacks. We first investigate the possibility and methods of attacking a TM-based ASV system, followed by a pilot data collection. Second, we study the use of spectral features for VLD using both single-channel and dual-channel ASV systems. We carry out speaker verification experiments using Gaussian mixture model with universal background model (GMM-UBM) and i-vector based systems on a dataset of 38 speakers collected by us. We have achieved considerable improvement in recognition accuracy, with the use of dual-microphone setup. In experiments with noisy test speech, the false acceptance rate (FAR) of the dual-microphone GMM-UBM based system for recorded speech reduces from 69.69% to 18.75%. The FAR of replay condition further drops to 0% when this dual-channel ASV system is integrated with the new dual-channel voice liveness detector.
Md. Sahidullah, Dennis Alexander Lehmann Thomsen, Rosa González Hautamäki, Tomi Kinnunen, Zheng-Hua Tan, Robert Parts, Martti Pitkänen
IEEE ACM Trans. Audio Speech Lang. Process.2
2017 RedDots replayed: A new replay spoofing attack corpus for text-dependent speaker verification research
abstract
This paper describes a new database for the assessment of automatic speaker verification (ASV) vulnerabilities to spoofing attacks. In contrast to other recent data collection efforts, the new database has been designed to support the development of replay spoofing countermeasures tailored towards the protection of text-dependent ASV systems from replay attacks in the face of variable recording and playback conditions. Derived from the re-recording of the original RedDots database, the effort is aligned with that in text-dependent ASV and thus well positioned for future assessments of replay spoofing countermeasures, not just in isolation, but in integration with ASV. The paper describes the database design and re-recording, a protocol and some early spoofing detection results. The new “RedDots Replayed” database is publicly available through a creative commons license.
Tomi Kinnunen, Md. Sahidullah, Mauro Falcone, Luca Costantini, Rosa González Hautamäki, Dennis Alexander Lehmann Thomsen, Achintya Kumar Sarkar, Zheng-Hua Tan, Héctor Delgado, Massimiliano Todisco, Nicholas W. D. Evans, Ville Hautamäki, Kong-Aik Lee
ICASSP6
2017 The I4U Mega Fusion and Collaboration for NIST Speaker Recognition Evaluation 2016
abstract
18th Annual Conference of the International Speech Communication Association, INTERSPEECH 2017, Stockholm, Sweden, 20-24 August 2017
Kong-Aik Lee, Ville Hautamäki, Tomi Kinnunen, Anthony Larcher, Andreas Nautsch, Themos Stafylakis, Gang Liu 0001, Mickael Rouvier, Wei Rao 0002, Federico Alegre, Man-Wai Mak, Achintya Kumar Sarkar, Héctor Delgado, Rahim Saeidi, Hagai Aronowitz, Aleksandr Sizov, Hanwu Sun, Trung Hieu Nguyen 0001, Guangsen Wang, Bin Ma 0001, Ville Vestman, Md. Sahidullah, M. Halonen, Anssi Kanervisto, Gaël Le Lan, Fahimeh Bahmaninezhad, Sergey Isadskiy, Christian Rathgeb, Christoph Busch 0001, Georgios Tzimiropoulos, Q. Qian, Q. Zhao, J. Xue, R. Jin, T. Zhao, Pierre-Michel Bousquet, Moez Ajili, Waad Ben Kheder, Driss Matrouf, Zhi Hao Lim, Chenglin Xu, Haihua Xu 0001, Chng Eng Siong, Benoit G. B. Fauve, Kaavya Sriskandaraja, Vidhyasaharan Sethu, W. W. Lin, Dennis Alexander Lehmann Thomsen, Zheng-Hua Tan, Massimiliano Todisco, Nicholas W. D. Evans, Haizhou Li 0001, John H. L. Hansen, Jean-François Bonastre, Eliathamby Ambikairajah
INTERSPEECH55
2016 HAPPY Team Entry to NIST OpenSAD Challenge: A Fusion of Short-Term Unsupervised and Segment i-Vector Based Speech Activity Detectors
abstract
Speech activity detection (SAD), the task of locating speech segments from a given recording, remains challenging under acoustically degraded conditions. In 2015, National Institute of Standards and Technology (NIST) coordinated OpenSAD bench-mark. We summarize “HAPPY” team effort to OpenSAD. SADs come in both unsupervised and supervised flavors, the latter requiring a labeled training set. Our solution fuses six base SADs (2 supervised and 4 unsupervised). The individually best SAD, in terms of detection cost function (DCF), is supervised and uses adaptive segmentation with i-vectors to represent the segments. Fusion of the six base SADs yields a relative decrease of 9.3% in DCF over this SAD. Further, relative decrease of 17.4% is obtained by incorporating channel detection side information.
Tomi Kinnunen, Alexey Sholokhov, Elie Khoury 0001, Dennis Alexander Lehmann Thomsen, Md. Sahidullah, Zheng-Hua Tan
INTERSPEECH4
2016 Robust Speaker Recognition with Combined Use of Acoustic and Throat Microphone Speech
abstract
Accuracy of automatic speaker recognition (ASV) systems degrades severely in the presence of background noise. In this paper, we study the use of additional side information provided by a body-conducted sensor, throat microphone. Throat microphone signal is much less affected by background noise in comparison to acoustic microphone signal. This makes throat microphones potentially useful for feature extraction or speech activity detection. This paper, firstly, proposes a new prototype system for simultaneous data-acquisition of acoustic and throat microphone signals. Secondly, we study the use of this additional information for both speech activity detection, feature extraction and fusion of the acoustic and throat microphone signals. We collect a pilot database consisting of 38 subjects including both clean and noisy sessions. We carry out speaker verification experiments using Gaussian mixture model with universal background model (GMM-UBM) and i-vector based system. We have achieved considerable improvement in recognition accuracy even in highly degraded conditions.
Md. Sahidullah, Rosa González Hautamäki, Dennis Alexander Lehmann Thomsen, Tomi Kinnunen, Zheng-Hua Tan, Ville Hautamäki, Robert Parts, Martti Pitkänen
INTERSPEECH3
2016 Speaker-Dependent Dictionary-Based Speech Enhancement for Text-Dependent Speaker Verification
abstract
The problem of text-dependent speaker verification under noisy conditions is becoming ever more relevant, due to increased usage for authentication in real-world applications. Classical methods for noise reduction such as spectral subtraction and Wiener filtering introduce distortion and do not perform well in this setting. In this work we compare the performance of different noise reduction methods under different noise conditions in terms of speaker verification when the text is known and the system is trained on clean data (mis-matched conditions). We furthermore propose a new approach based on dictionary-based noise reduction and compare it to the baseline methods.
Nicolai Bæk Thomsen, Dennis Alexander Lehmann Thomsen, Zheng-Hua Tan, Børge Lindberg, Søren Holdt Jensen
INTERSPEECH2