The Audio Analysis Lab’s annual workshop, the Audio Analysis Workshop, was held on August 21 this year. The workshop featured a keynote talk entitled Multi-Microphone Speaker Localization on Manifolds by Prof. Sharon Gannot of Bar-Ilan University, Israel, and 14 oral and 15 poster presentations about ongoing research. The 2018 edition of the workshop, which was co-sponsored by the GN Store Nord Foundation and the Independent Research Fund Denmark, was attended by 40+ people from both academia (Delft University of Technology, Lund University, KU Leuven, Aston University, and Aalborg University) and industry (e.g., B&O, Terma, Jabra GN Hearing, UniqiSense, Intel).
The Audio Analysis Lab is extremely happy to announce that Prof. Sharon Gannot of Bar-Ilan University in Israel will be affiliated with us for the next year as Guest Professor. Prof. Sharon Gannot is well-known to all in audio and acoustic signal processing, and he is a prominent figure in our community. He will work with the Audio Analysis Lab on the topics of microphone array signal processing and will be giving a Ph.D. course on this topic at Aalborg University.
Biography: Sharon Gannot eceived the B.Sc. degree (summa cum laude) from the Technion-Israel Institute of Technology, Haifa, Israel, in 1986, and the M.Sc. (cum laude) and Ph.D. degrees from Tel-Aviv University, Tel Aviv, Israel, in 1995 and 2000, respectively, all in electrical engineering. In 2001, he held a Postdoctoral position with the Department of Electrical Engineering, KU Leuven, Leuven, Belgium. From 2002 to 2003, he held a Research and Teaching position with the Faculty of Electrical Engineering, Technion–Israel Institute of Technology. He is currently, a Full Professor with the Faculty of Engineering, Bar-Ilan University, Ramat Gan, Israel, where he is heading the Speech and Signal Processing Laboratory and the Signal Processing Track. Since April 2018, he is also a part-time Professor at the Technical Faculty of IT and Design, Aalborg University, Denmark. His research interests include multi-microphone speech processing and specifically distributed algorithms for ad hoc microphone arrays for noise reduction and speaker separation, dereverberation, single microphone speech enhancement, and speaker localization and tracking. He was an Associate Editor for the EURASIP Journal of Advances in Signal Processing during 2003–2012, and an Editor for several special issues on multi-microphone speech processing of the same journal. He was a Guest Editor for the Elsevier Speech Communication and Signal Processing journals. He was an Associate Editor for the IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING during 2009–2013, and the Area Chair for the same journal during 2013–2017. He is currently a Moderator for arXiv in the field of audio and speech processing. He is also a Reviewer for many IEEE journals and conferences. Since January 2010, he has been a Member of the Audio and Acoustic Signal Processing technical committee of the IEEE and serves, since January 2017, as the Committee Chair. Since 2005, he has also been a Member of the technical and steering committee of the International Workshop on Acoustic Signal Enhancement (IWAENC) and was the General Co-Chair of the IWAENC held in Tel-Aviv, Israel in August 2010. He was the General Co-Chair of the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics in October 2013. He was selected (with colleagues) to present tutorial sessions at ICASSP 2012, EUSIPCO 2012, ICASSP 2013, and EUSIPCO 2013 and was a keynote speaker for IWAENC 2012 and LVA/ICA 2017. He was the recipient of the Bar-Ilan University Outstanding Lecturer Award in 2010 and 2014 and the Rector Innovation in Research Award in 2018. He is also a co-recipient of ten best paper awards.
The Audio Analysis Lab is organizing a Ph.D. course in August on Advanced Topics in Acoustic Array Signal Processing. The course will be given by Prof. Sharon Gannot from Bar-Ilan University, Israel, who is Guest Professor at Audio Analysis Lab and lab member Assistant Prof. Jesper Rindom Jensen. The course will be held August 13-17, 20 2018 at Rendsburggade 14 at CREATE in Aalborg. Participants from other universities and companies are welcome to participate. You can read more about the course here.
Description: Acoustic arrays are becoming a ubiquitous technology in many places, including in consumer electronics and healthcare technology. Microphone arrays are now found in smartphones, laptops, TVs, etc., and loudspeaker arrays are emerging as a promising technology in home entertainment systems, car audio systems, public announcement systems. Moreover, as wireless communication capabilities are becoming widespread, audio devices can now form ad hoc networks and cooperate when solving signal processing problems, such as estimation and filtering. This offers many new possibilities but also poses many new challenges, as it requires that many difficult, technical problems must be solved. In the course, a general introduction to acoustic array signal processing will be given, including commonly used models and assumptions as well as classical methods for solving problems such as localization, beamforming and noise reduction. The remainder of the course is then devoted to recent advances in acoustic array signal processing and applications. These include advances within, for example, model-based localization and beamforming, sound zone control with loudspeaker arrays, multi-channel noise reduction in ad hoc microphone arrays, noise statistics estimation, speech intelligibility prediction, and speech enhancement in binaural hearing aids.
The course is dedicated to the following subjects:
- Fundamentals: Definitions, narrow-band signals, near-filed and far-field, array manifold vector. Beamforming, uniform linear array, directivity pattern. Performance criteria (beam-width, sidelobe level, directivity, white noise gain). Sensitivity. Sampling of continuous aperture. Wide-band signals and nested arrays.
- Space-time random processes: Snapshots, spatial correlation matrix, signal and noise subspaces.
- Optimal array processors: MVDR (Capon), MPDR, Maximum SNR, MMSE, LCMV.
- Sensitivity and robustness: Noise fields and multi-path and their influence on performance. Superdirective beamformer. Diagonal loading.
- Adaptive spatial filtering: Frost method, generalized sidelobe canceller (GSC).
- Parameter estimation (DoA): ML estimation, resolution, Cramér-Rao lower bound.
- Classical methods for localization: Classical methods (Bartlett), method based on eigen-decomposition: Pisarenko, MUSIC, ESPRIT. Resolution. MVDR estimation. Performance evaluation and comparison.
- Advances: Model-based processing and estimation, multi-channel noise reduction, ad hoc microphone arrays.
- Applications: Speech processing, hearing aids, wireless acoustic sensor networks, loudspeaker arrays.
Audio Analysis Lab members Assistant Professors Jesper Rindom Jensen and Jesper Kjær Nielsen along with Professor Mads Græsbøll Christensen will be giving a tutorial at ICASSP 2018, which will be held in Calgary, Canada on April 15-20. The tutorial is entitled Model-based Speech and Audio Processing and is based on the lab’s work in various projects, including our work on pitch estimation, noise reduction, audio analysis, microphone arrays, etc. ICASSP, which is organized by the IEEE, is the top conference on signal processing in the world. You can read more about the tutorial using the following link: https://2018.ieeeicassp.org/Tutorials.asp#T-9.
The 43rd IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP) 2018 will be held April 15-20, 2018 in Calgary, Canada. As usual, the Audio Analysis Lab will be present at the top signal processing conference in the world with the following papers that will be presented at the conference:
- A PARAMETRIC APPROACH FOR CLASSIFICATION OF DISTORTIONS IN PATHOLOGICAL VOICES
- MULTIPITCH ESTIMATION USING BLOCK SPARSE BAYESIAN LEARNING AND INTRA-BLOCK CLUSTERING
- ESTIMATION OF SOURCE PANNING PARAMETERS AND SEGMENTATION OF STEREOPHONIC MIXTURES
- A STUDY OF NOISE PSD ESTIMATORS FOR SINGLE CHANNEL SPEECH ENHANCEMENT
- LOUDSPEAKER AND LISTENING POSITION ESTIMATION USING SMART SPEAKERS
- MODEL-BASED NOISE PSD ESTIMATION FROM SPEECH IN NON-STATIONARY NOISE
- A SUPERVISED APPROACH TO GLOBAL SIGNAL-TO-NOISE RATIO ESTIMATION FOR WHISPERED AND PATHOLOGICAL VOICES
- A UNIFIED APPROACH TO GENERATING SOUND ZONES USING VARIABLE SPAN LINEAR FILTERS
Audio Analysis Lab member Liming shi received a prestigious EliteForsk (Elite Research) Travel Stipend from the Ministry of Education and Science yesterday during the annual EliteForsk celebration at the beautiful Copenhagen Opera House. The stipends, which are awarded to current Ph.D. students in Denmark, consists of 200,000 DKK for visiting universities abroad and conference participation. The travel stipends are awarded by nomination and the candiates are evaluated by council members of Independent Research Fund Denmark. Only 20 of these highly competitive stipends are awarded annually, and only two were awarded to Ph.D. students at AAU. Liming Shi will be visiting Cambridge University, United Kingdom and Alto University, Finland. You can read more about the EliteForsk celebration here and about Liming and his work here.
At the Technical Faculty of IT and Design, Department of Architecture, Design and Media Technology one or more PhD stipends are available within the general research programme Electrical and Electronic Engineering. The stipend(s) are open for appointment from 1st of April 2018 or as soon thereafter as possible.
The open position(s) are with the research group Audio Analysis Lab. The PhD students will work on a research project entitled ‘Sound Processing for Robots and Drones in the Fourth Industrial Revolution’.
Sound is an often overlooked modality in robot and drones despite its potential to increase the application of robots and drones in industrial production. By using echolocation like bats, the robots may orient themselves and navigate blindly by emitting sound and listening for reflected echoes. Compared to commonly used sensors such as depth cameras and LIDAR, the usage of microphones (MEMS) can offer a cheaper solution, which is robust against, e.g., changing lighting conditions. Sound may therefore facilitate autonomous and indoor robot operation for production, which is a key to increase the competitiveness of high-tech countries. This is also known as the fourth industrial revolution. The aim of this project is, thus, to facilitate such operation via advanced signal processing for robots equipped with microphone arrays (multiple microphones in a known geometrical configuration).
The successful applicant should have a M.Sc. (or equivalent) in engineering within signal processing. Prior experience with audio and acoustic signal processing and mobile robot platforms is a plus but not required. Moreover, the successful applicant should be fluent in English, have strong programming and math skills, and be familiar with MATLAB (or similar tools). The applicant must submit his/her M.Sc. thesis (or a draft thereof) as part of the application. The degree must be completed at the time of the appointment.
The Audio Analysis Lab at Aalborg University conducts basic and applied research in signal processing theory and methods aimed at or involving analysis of audio signals. The research focuses on problems such as compression, analysis, classification, separation, and enhancement of audio signals, as well as localization, identification and tracking using microphone arrays. The lab and its members have in recent years been funded by grants from the Villum Foundation, the Danish Council for Strategic Research, the Danish Council for Independent Research, and Innovations Fund Denmark. The research projects are carried out in close collaboration with leading industrial partners and universities around the world.
You may obtain further information from Assistant Professor Jesper Rindom Jensen, Department of Architecture, Design and Media Technology, phone: +45 9940 7450, email: firstname.lastname@example.org concerning the scientific aspects of the stipend.
PhD stipends are allocated to individuals who hold a Master’s degree. PhD stipends are normally for a period of 3 years. It is a prerequisite for allocation of the stipend that the candidate will be enrolled as a PhD student at the Technical Doctoral School of IT adn Design in accordance with the regulations of Ministerial Order No. 1039 of August 27, 2013 on the PhD Programme at the Universities and Certain Higher Artistic Educational Institutions. According to the Ministerial Order, the progress of the PhD student shall be assessed every six months. It is a prerequisite for continuation of salary payment that the previous progress is approved at the time of the evaluation.
The qualifications of the applicant will be assessed by an assessment committee. On the basis of the recommendation of the assessment committee, the Dean of the Technical Faculty of IT and Design will make a decision for allocating the stipend.
For further information about stipends and salary as well as practical issues concerning the application procedure contact Ms. Helene Ulrich Pedersen, The Technical Faculty of IT and Design, email: email@example.com, phone: +45 9940 9803.
You can read more and apply at http://www.stillinger.aau.dk/vis-stilling/?vacancy=956268.
The Audio Analysis Lab is extremely proud to announce that founding member Assistant Professor Jesper Rindom Jensen has been accepted in AAU’s research talent program for younger researchers. The program is part of AAU’s strategy for 2016-2021 and provides funding for a research project proposed by the young researcher. This round of applications was extremely competitive and 10 out of 52 applications were approved by an independent evaluation committee. You can read more about the program and the young researchers who received funding here.
The Audio Analysis Lab presented two papers at the IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, which was held October 15-18 at Mohonk Mountain House, New Paltz, New York. The lab was also part of the organizing team for this edition of the workshop, as Mads Græsbøll Christensen was Technical Program Co-Chair. The papers presented were:
Experimental Study of Robust Beamforming Techniques for Acoustic Applications Yingke Zhao (Northwestern Polytechnical University, P.R. China); Jesper Rindom Jensen and Mads Græsbøll Christensen (Aalborg University, Denmark); Simon Doclo (University of Oldenburg, Germany); Jingdong Chen (Northwestern Polytechnical University, P.R: China)
In this paper, we investigate robust beamforming methods for wideband signal processing in noisy and reverberant environments. In such environments, the appearance of steering vector estimation errors is inevitable, which degrades the performance of beamformers. Here, we study two types of robust beamformers against this estimation inaccuracy. The first type includes the norm constrained Capon, the robust Capon, and the doubly constrained robust Capon beamformers. The underlying principle is to add steering vector uncertainty constraint and norm constraint to the optimization problem to improve the beamformer’s robustness. The second one is the amplitude and phase estimation method, which utilizes both time and spatial smoothing to obtain robust beamforming. Experiments are presented to demonstrate the performance of the robust beamformers in acoustic environments. The results show that the robust beamformers outperform the non-robust methods in many respects: 1) robust performance in reverberation and different noise levels; 2) resilience against steering vector and covariance matrix estimation errors; and 3) better speech quality and intelligibility.
A Kalman-Based Fundamental Frequency Estimation Algorithm Liming Shi, Jesper Kjær Nielsen and Jesper Rindom Jensen (Aalborg University, Denmark); Max Little (MIT, USA); Mads Græsbøll Christensen (Aalborg University, Denmark)
Current and former lab members Hendrik Purwins and Bob Sturm (Queen Mary University of London) along with Mark Plumbley (University of Surrey) are organizing the Workshop “Machine Learning for Audio Signal Processing (ML4Audio)” at NIPS 2017, on Friday, December 8th, 2017 in Los Angeles and invite you to submit contributions: