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Low-Complexity Signal Processing for Speech Enhancement and Audio Analysis
Blekinge Institute of Technology, Faculty of Engineering, Department of Mathematics and Natural Sciences.
2022 (English)Licentiate thesis, comprehensive summary (Other academic)
Abstract [en]

In real-time signal processing there is a constraint to finish processing of an audio signal before the next audio segment is received. This makes it important to have signal processing algorithms with low computational complexity while still maintaining high quality results. This thesis presents methods for audio signal processing used in real-time systems. The publications presented cover areas of noise reduction, network echo cancellation, noise dosimeter measurements and voice analysis.

A method for speech enhancement is presented with low amounts of speech distortion. The audio signal is split into several subbands, covering different frequency regions. For each subband, the noise level is estimated. A signal gain is calculated by comparing the total signal level with the noise level for each subband. The method presented here, improves performance compared to previously similar methods. Improvement is especially found in multi-speaker and noise-only scenarios.

When communicating on a telephone line, network echo is introduced by hybrids in the network. In cases where multiple echo sources exist, the time range for echoes can be quite long. In devices with limited storage, it is difficult to get good echo cancellation in such cases. This thesis presents a method for network echo cancellation suited for use in a device with a larger external memory.

Exposure to high noise levels will have negative health effects and methods for measuring noise level exposure is important. Included in this thesis is a study that remove the influence of own voice in noise dose measurements.

For certain medical conditions it is beneficial with daily voice exercises. Methods for grading voice in four different exercises is presented, based on pitch and loudness. Evaluation is done in real-time on test medical device.

Place, publisher, year, edition, pages
Karlskrona: Blekinge Tekniska Högskola, 2022.
Series
Blekinge Institute of Technology Licentiate Dissertation Series, ISSN 1650-2140 ; 2022:01
National Category
Signal Processing
Research subject
Applied Signal Processing
Identifiers
URN: urn:nbn:se:bth-22350ISBN: 978-91-7295-434-2 (print)OAI: oai:DiVA.org:bth-22350DiVA, id: diva2:1611499
Supervisors
Available from: 2021-11-15 Created: 2021-11-15 Last updated: 2022-04-29Bibliographically approved
List of papers
1. An improved adaptive gain equalizer for noise reduction with low speech distortion
Open this publication in new window or tab >>An improved adaptive gain equalizer for noise reduction with low speech distortion
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2011 (English)In: EURASIP Journal on Audio, Speech, and Music Processing, ISSN 1687-4714, E-ISSN 1687-4722, Vol. 7Article in journal (Refereed) Published
Abstract [en]

In high-quality conferencing systems, it is desired to perform noise reduction with as limited speech distortion as possible. Previous work, based on time varying amplification controlled by signal-to-noise ratio estimation in different frequency subbands, has shown promising results in this regard but can suffer from problems in situations with intense continuous speech. Further, the amount of noise reduction cannot exceed a certain level in order to avoid artifacts. This paper establishes the problems and proposes several improvements. The improved algorithm is evaluated with several different noise characteristics, and the results show that the algorithm provides even less speech distortion, better performance in a multi-speaker environment and improved noise suppression when speech is absent compared with previous work.

Place, publisher, year, edition, pages
Springer, 2011
Keywords
Speech enhancement, Noise reduction, Noise-level estimation
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-6967 (URN)10.1186/1687-4722-2011-7 (DOI)000299122500005 ()oai:bth.se:forskinfoEF3499AC58836F03C125798A007408EC (Local ID)oai:bth.se:forskinfoEF3499AC58836F03C125798A007408EC (Archive number)oai:bth.se:forskinfoEF3499AC58836F03C125798A007408EC (OAI)
Note

Open Access article Article 7

Available from: 2013-05-31 Created: 2012-01-19 Last updated: 2021-11-18Bibliographically approved
2. Low-complexity network echo cancellation approach for systems equipped with external memory
Open this publication in new window or tab >>Low-complexity network echo cancellation approach for systems equipped with external memory
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2011 (English)In: IEEE Transactions on Audio, Speech, and Language Processing, ISSN 1558-7916, E-ISSN 1558-7924, Vol. 19, no 8, p. 2506-2515Article in journal (Refereed) Published
Abstract [en]

Long delays and sparseness characterize impulse responses in telecommunication networks and a vast number of solutions for network echo cancellation have been proposed over the years. In this paper, an approach for detecting dispersive regions of a sparse impulse response and a proportionate normalized least mean square (PNLMS)-based selective updating approach are combined with an adaptive double-talk detector to form a complete solution for echo cancellation. The proposed solution has low computational complexity and is targeted for systems equipped with external memory.

Place, publisher, year, edition, pages
IEEE, 2011
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-7219 (URN)10.1109/TASL.2011.2144972 (DOI)000296889900024 ()oai:bth.se:forskinfo2702207CF5FD5B6AC12578B0002A3BAD (Local ID)oai:bth.se:forskinfo2702207CF5FD5B6AC12578B0002A3BAD (Archive number)oai:bth.se:forskinfo2702207CF5FD5B6AC12578B0002A3BAD (OAI)
Available from: 2012-11-06 Created: 2011-06-15 Last updated: 2021-11-18Bibliographically approved
3. The Effect of Own Voice on Noise Dosimeter Measurements: A Field Study in a Day-Care Environment, Including Adults and Children
Open this publication in new window or tab >>The Effect of Own Voice on Noise Dosimeter Measurements: A Field Study in a Day-Care Environment, Including Adults and Children
2008 (English)Conference paper, Published paper (Refereed)
Abstract [en]

Noise dosimeters are valuable tools in assessing the individual noise dose in the workplace. At non-industrial work places with a high degree of communication, such measurements would include the wearer’s own voice which would be registered as noise. This may not always be desirable. The purpose of this investigation was to study the effect of the wearers own voice in noise dosimeter measurements, and especially the difference between children and adults as test subjects. The study took place at a day-care center and sixteen children and thirteen adult female preschool teachers participated. The participants wore a digital recorder during the day, which recorded the sound signal and vibrations originating from an accelerometer attached to the neck of the test subjects, for distinguishing of whether the subject was speaking or not. Thus, average A-weighted noise levels with and without the influence of the subjects own voice could be obtained. The Leq for the measurements with and without the own voice was 84.6 dBA and 72.2 dBA for the children, respectively, and 79.3 dBA and 70.0 dBA for adults. Student’s t-test showed a significant (p<0.01) difference of 12.4 dBA for children and 9.3 dBA for adults when comparing measurements including and excluding the own voice and also that the difference was significantly larger for children. Thus, the study conclude that the influence from the own voice implied an augmentation of the Leq value and that there is a significant difference between children and adults in how large this augmentation is.

Place, publisher, year, edition, pages
Shanghai: , 2008
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-8438 (URN)oai:bth.se:forskinfoBE016998EA40D996C12574A90045FA92 (Local ID)oai:bth.se:forskinfoBE016998EA40D996C12574A90045FA92 (Archive number)oai:bth.se:forskinfoBE016998EA40D996C12574A90045FA92 (OAI)
Conference
Internoise
Available from: 2012-09-18 Created: 2008-08-18 Last updated: 2021-11-18Bibliographically approved
4. A Personal Voice Analyzer and Trainer
Open this publication in new window or tab >>A Personal Voice Analyzer and Trainer
2010 (English)Conference paper, Published paper (Refereed)
Abstract [en]

This paper presents a personal voice analyzer and trainer that allow the user to perform four daily exercises to improve the voice capacity. The system grades how well the user is performing the exercises by analyzing the duration, the intensity and the pitch of the user’s voice.

Place, publisher, year, edition, pages
Las Vegas, USA: IEEE, 2010
National Category
Signal Processing
Identifiers
urn:nbn:se:bth-7847 (URN)000286972900002 ()oai:bth.se:forskinfo7FFDD6F88E1D886BC12576DB00551A65 (Local ID)978-1-4244-4314-7 (ISBN)oai:bth.se:forskinfo7FFDD6F88E1D886BC12576DB00551A65 (Archive number)oai:bth.se:forskinfo7FFDD6F88E1D886BC12576DB00551A65 (OAI)
Conference
IEEE International Conference on Consumer Electronics
Available from: 2012-09-18 Created: 2010-03-03 Last updated: 2021-11-18Bibliographically approved

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