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You are interesed in working with us on ASFHEAR project? The following positions are open for application, so contact us

Postdoc position #2: Speech intelligibility tests using adaptive sound field synthesis

  • Duration: 12 months
  • Starting date: Spring 2027
  • Salary: 30-34k€ / year before tax, depending on experience (INM 518-577)
  • Location: Acoustic Team of LMFA - École Centrale de Lyon, Écully, France
  • Keywords: speech intelligibility, sound field synthesis, loudspeaker array, adaptive filtering, reverberant rooms, hearing aid research

Context and objectives

Speech intelligibility decreases drastically in reverberant environments, and hearing-impaired people often complain of the poor performances of their hearing devices in such situations. One reason for this mismatch is that clinical protocols for hearing aid calibration fail to reproduce real-life environments faithfully: they typically rely on a single loudspeaker and overly simplistic stimuli9. A genuine environment simulator is therefore needed. Moreover, speech intelligibility tests are usually conducted over headphones, which is problematic for hearing-aid users, as the hearing aid interacts with the headphone through feedback and instability (Larsen effect)4. In this context, the ASFHEAR project develops a physically-accurate acoustic environment simulator based on an adaptive transaural synthesis: the algorithm has already been implemented and physically validated10, but its perceptual validation through speech intelligibility tests remains to be carried out, to validate its use in place of headphone tests, considered here as the reference.

The postdoc will focus on conducting speech intelligibility tests using this adaptive transaural system, to validate it as a reference tool for hearing aid research: so far, only few systems using loudspeaker arrays exist for hearing aid research111213, and none of them has been compared to a headphone-based test, taken here as the reference.

The candidate will contribute to a bibliographical survey of speech intelligibility studies in virtual reverberant rooms141516, with an emphasis on studies using loudspeaker arrays111213. The candidate will then take over the adaptive transaural system deployed in the LMFA listening room, and conduct speech intelligibility listening tests with normal-hearing listeners. A first test will use the system developed in the project10 in a static condition (the listener uses a chin rest). A second test will allow listeners simple head movements, first controlled and then free, with adaptive filtering compensating for these movements in real time. This should demonstrate the benefit of the adaptive approach for speech intelligibility tests over loudspeaker arrays with moving listeners. The candidate will recruit participants, analyse the perceptual data, and disseminate the results through international conferences and scientific publications.

This work will open up the possibility of testing hearing-impaired listeners with the ASFHEAR adaptive transaural system, provided that a timely approval is obtained from the research ethics committee. As a further opening, a complementary approach based on adaptive sound field synthesis, rather than transaural, is currently under development within the ASFHEAR project and could possibly be evaluated as well.

Supervision and research environment

The postdoc will be supervised by Pierre Lecomte, Associate Professor at LMFA, and Mathieu Lavandier, Researcher at LTDS (ENTPE), as part of the ASFHEAR project. The postdoc will collaborate with a research intern hired to help set up the experimental protocol.

As part of the ASFHEAR project, the candidate will join a stimulating research team in the Lyon area (see the team), with opportunities of teaching at École Centrale de Lyon. Lyon and its surroundings offer rich opportunities for culture and outdoor activities.

Candidate profile

Applicants should have:

  • A PhD degree in acoustics, psychoacoustics, audiology or a related field,
  • Background in speech intelligibility or psychoacoustic testing with human listeners,
  • Fluent oral and written communication skills in English,
  • Coding skills in Python,
  • Laboratory experimental work background.

Application procedure

Applicants should send a CV (summarizing education, research background and other qualifying activity), a cover letter, copies of educational certificates, and a list of publications to pierre.lecomte@ec-lyon.fr and mathieu.lavandier@entpe.fr.

Informal enquiries about the position and the research project are very welcome and may be sent to the same addresses, or by phone: +33 4 72 18 60 13.

Please note that the application is subject to security clearance, as the laboratory is in a restricted access area.

PhD position: Adaptive Sound Field Synthesis — filled
  • Duration: 36 months
  • Starting date: Nov. 2024 - Jan. 2025 (flexible)
  • Salary: 25-27k€ / year before tax
  • Location: Acoustic Team of LMFA - École Centrale de Lyon, Écully, France
  • Keywords: acoustic, signal-processing, adaptive filtering, real-time, active control, hearing aids

This PhD project aims at developping and testing an adaptive Sound Field Synthesis (SFS) system for human listeners, to be used for hearing aid research.

A first task will consists of a bibliographical survey of SFS techniques using a loudspeaker array for hearing aid research1234.
Based on these knowledge, the candidate will developp an adaptive SFS algorithm based on state-of-the art transaural approaches4 using a loudspeaker array and a dummy head.
This algorithm will be implemented and tested in real-time in the lab, in collaboration with the Postdoc researcher #1 and the help of a Master 2 intern.
In a second time, adaptive SFS techniques using modal approachs (i.e. Ambisonics) will be investigated.
After validation, this system will be used for speech intelligibility tests in virtual reverberant room (see Postdoc researcher #2)

Qualification requirements

  • Master degree in acoustics and/or signal processing,
  • Fluent oral and written communication skills in English,
  • Strong coding skills in Python and Matlab/Simulink,
  • Laboratory experimental work background,
  • Knowledge in psychoacoustics are not required but appreciated.

Application

  • CV (summarizing education, research background and other qualifying activity),
  • Cover letter,
  • Copies of educational certificates, academic transcript of records,
  • An example of scientific document (scientific paper, master internship report)

Please note that the application is subject to security clearance, as the laboratory is in a restricted access area.

We offer

  • As part of the ASFHEAR project, we offer an stimulating working environment within a research team in the Lyon area (see the team).
  • The are opportunities of teaching at École Centrale de Lyon and/or in the international MSc in Acoustics.
  • The candidate will conduct his PhD inside the doctoral school MEGA. The latter help doctoral students develop a career plan throughout their thesis, by offering a wide range of training courses, and promoting interdisciplinary work, relations with companies and international contacts.
  • Lyon city and surroundings offer rich opportunities for culture and outdoor activities.

Contact person

For more information or to apply, you can use our contact form or contact Associate Processor Pierre Lecomte:

phone: +33 4 72 18 60 13
email: pierre <dot> lecomte <at> ec-lyon.fr

Postdoc position #1: Real-Time Massive Multichannel Adaptive Filtering Techniques — filled
  • Duration: 18 months
  • Starting date: ~Mar. 2025 (flexible)
  • Salary: 30-36k€ / year before tax
  • Keywords: real-time, filtering, active control systems, acoustic

This postdoctoral researcher fellowship aims at designing and implementing algorithms for massive multichannel filtering to be used in a adaptive filtering context.

Adaptive filtering is required for adaptive sound field synthesis algorithm developped in the ASFHEAR project. Usually, the adaptive filtering algorithms are formulated using Finite Impulse Response (FIR) filters.5 The larger the frequency band to be processed, the longer the FIRs (several thousand coefficients for processing speech signals). Usually, the number of convolutions increases drastically with the algorithm complexity, making the massive real-time operations unfeasible on modern controllers. We are therefore interested on developing a parallel adaptive filtering approach using Infinite Impulse Response (IIR) filters (i.e., biquad filters).6 These filters are individually inexpensive, and parallel computing is well suited to modern controllers.

The project will be segmented in 3 main tasks:
1. The first task is to provide a bibliographical survey on fast filtering algorithms, which could be implemented on a modern controller, available at the lab.8
2. The second task is to explore and improve current algorithms for FIR filters decomposition as parallel stable IIR filters.7
3. The third task is to impement on the lab controller a massive multichannel adaptive filtering process, at the heart of the adaptive SFS algorithm designed by the PhD student

Qualification requirements

  • PhD degree in audio signal processing, control systems,
  • Fluent oral and written communication skills in English,
  • Strong coding skills in Python and Matlab/Simulink,
  • Background with DSP programming or FPGA,
  • Laboratory experimental work background.

Application

  • CV (summarizing education, research background and other qualifying activity),
  • List of publication
  • Cover letter,
  • Copies of educational certificates, academic transcript of records,

Please note that the application is subject to security clearance, as the laboratory is in a restricted access area.

We offer

Contact person

For more information or to apply, you can use our contact form or contact Associate Processor Pierre Lecomte:

phone: +33 4 72 18 60 13
email: pierre <dot> lecomte <at> ec-lyon.fr

Master 2 Internship #1

  • Duration: 6 months
  • Starting date:
  • Salary: 609€/month

To be published soon

Master 2 Internship #2

  • Duration: 6 months
  • Starting date:
  • Salry: 609€/month

To be published soon

Master 2 Internship #3

  • Duration: 6 months
  • Starting date:
  • Salary: 609€/month

To be published soon

References

  1. G. Grimm, S. Ewert, V. Hohmann, Evaluation of spatial audio reproduction schemes for application in hearing aid research, Acta Acustica United with Acustica 101 (4) (2015) 842–854. ↩

  2. P. Minnaar, S. F. Albeck, C. S. Simonsen, B. Søndersted, S. A. D. Oakley, J. Bennedbæk, Reproducing real-life listening situations in the laboratory for testing hearing aids, in: Audio Engineering Society Convention 135, Audio Engineering Society, 2013. ↩

  3. C. Oreinos, J. M. Buchholz, Evaluation of loudspeaker-based virtual sound environments for testing directional hearing aids, Journal of the American Academy of Audiology 27 (07) (2016) 541–556. ↩

  4. F. Pausch, L. Aspöck, M. Vorländer, J. Fels, An Extended Binaural Real-Time Auralization System With an Interface to Research Hearing Aids for Experiments on Subjects With Hearing Loss, Trends in Hearing 22 (2018). ↩↩↩

  5. S. J. Elliott, Signal Processing For Active Control, Academic Press, London, 2001. ↩

  6. P. Regalia, Adaptive IIR Filtering in Signal Processing and Control, Routledge, 2018. ↩

  7. https://github.com/orchidas/Modal-estimation/tree/main ↩

  8. https://www.speedgoat.com/products-services/real-time-target-machines ↩

  9. ISO 8253-3: Acoustics. Audiometric test methods-Part 3 Speech Audiometry, 2012. ↩

  10. T. Fouchard, A. Rigaud, P. Lecomte, An Extended Multichannel Frequency-Domain FxLMS Algorithm for Real-Time Full-Band Adaptive Transaural Reproduction, in: AES 160th Convention, Audio Engineering Society, 2026. ↩↩

  11. A. Ahrens, M. Marschall, T. Dau, Measuring and modeling speech intelligibility in real and loudspeaker-based virtual sound environments, Hearing Research 377 (2019) 307–317. ↩↩

  12. E. Au, S. Xiao, J. Hui, Y. Hioka, H. Masuda, C. Watson, Speech intelligibility in noise with varying spatial acoustics under Ambisonics-based sound reproduction system, Applied Acoustics 174 (2021) 107707. ↩↩

  13. J. Hui, E. Au, S. Xiao, Y. Hioka, H. Masuda, C. Watson, Differences in speech intelligibility in noise between native and non-native listeners under Ambisonics-based sound reproduction system, Applied Acoustics 184 (2021) 108368. ↩↩

  14. M. Lavandier, S. Jelfs, J. F. Culling, A. J. Watkins, A. P. Raimond, S. J. Makin, Binaural prediction of speech intelligibility in reverberant rooms with multiple noise sources, The Journal of the Acoustical Society of America 131 (1) (2012) 218–231. ↩

  15. B. Collin, M. Lavandier, Binaural speech intelligibility in rooms with variations in spatial location of sources and modulation depth of noise interferers, The Journal of the Acoustical Society of America 134 (2) (2013) 1146–1159. ↩

  16. N. Prodi, M. Pellegatti, C. Visentin, Comparing the effects of scattered and specular sound reflections on speech intelligibility in rooms, Building and Environment 228 (2023) 109881. ↩