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Low-complexity network echo cancellation approach for systems equipped with external memory
Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för matematik och naturvetenskap. Limes Audio AB, Umeå.
Limes Audio AB, Umeå.
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2011 (engelsk)Inngår i: IEEE Transactions on Audio, Speech, and Language Processing, ISSN 1558-7916, E-ISSN 1558-7924, Vol. 19, nr 8, s. 2506-2515Artikkel i tidsskrift (Fagfellevurdert) 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.

sted, utgiver, år, opplag, sider
IEEE , 2011. Vol. 19, nr 8, s. 2506-2515
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Identifikatorer
URN: urn:nbn:se:bth-7219DOI: 10.1109/TASL.2011.2144972ISI: 000296889900024Lokal ID: oai:bth.se:forskinfo2702207CF5FD5B6AC12578B0002A3BADOAI: oai:DiVA.org:bth-7219DiVA, id: diva2:834802
Tilgjengelig fra: 2012-11-06 Laget: 2011-06-15 Sist oppdatert: 2021-11-18bibliografisk kontrollert
Inngår i avhandling
1. Low-Complexity Signal Processing for Speech Enhancement and Audio Analysis
Åpne denne publikasjonen i ny fane eller vindu >>Low-Complexity Signal Processing for Speech Enhancement and Audio Analysis
2022 (engelsk)Licentiatavhandling, med artikler (Annet vitenskapelig)
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.

sted, utgiver, år, opplag, sider
Karlskrona: Blekinge Tekniska Högskola, 2022
Serie
Blekinge Institute of Technology Licentiate Dissertation Series, ISSN 1650-2140 ; 2022:01
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Forskningsprogram
Tillämpad signalbehandling
Identifikatorer
urn:nbn:se:bth-22350 (URN)978-91-7295-434-2 (ISBN)
Veileder
Tilgjengelig fra: 2021-11-15 Laget: 2021-11-15 Sist oppdatert: 2022-04-29bibliografisk kontrollert

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Berggren, MagnusBorgh, MarkusClaesson, Ingvar

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