| 1. |
- Berggren, Magnus, et al.
(författare)
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Analysis of How the Noise Level Depends on Different Activities in a Child Day-Care Center
- 2008
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Konferensbidrag (refereegranskat)abstract
- In child day-care centers the noise level can rise to high levels and in some cases become so high that the people present risk hearing damage. The purpose of this investigation was to study how the noise level depends on the different activities during the day. The study was performed at a child day-care center and 6 children and 5 adult female teachers participated. The participants had a microphone attached next to the ear connected to a wearable digital recorder. A total of 32.5 hours of data was recorded. By listening tests the recorded data could be sorted by activity and by number of people present in the same room as the test subject. Activities were classified as belonging to one of the following: outdoor activity, indoor play, singing, storytelling and gathering. Further, by listening, the data was classified in small group/large group (3 or less/more than 3). The results show that the average noise level (LAeq) for outdoor activity was the highest and was measured to 88.1 dBA (average over 7h52min). Singing was 81.5 dBA (1h26min), indoor play 81.3 dBA (19h21min), storytelling 76.6 dBA (1h09min) and gathering 75.0 dBA (2h44min). The noise level difference between all activities except between singing and indoor play and gathering and storytelling could be verified using t-test (p<0.001). Further, the results showed that the average noise level was 86.6 dBA (14h11min) for the large group and 79.6 dBA (18h21min) for the small group. This difference, of 7.0 dB was statistically validated (p<0.001) using t-test.
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| 2. |
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| 3. |
- Borgh, Markus, 1983-, et al.
(författare)
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The Effect of Own Voice on Noise Dosimeter Measurements : A Field Study in a Day-Care Environment, Including Adults and Children
- 2008
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Konferensbidrag (refereegranskat)abstract
- 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.
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| 4. |
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| 5. |
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| 6. |
- Lindström, Fredric, et al.
(författare)
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A hybrid acoustic echo canceller and suppressor
- 2007
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Ingår i: Signal Processing. - : Elsevier. - 0165-1684. ; 87:4, s. 739-749
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Tidskriftsartikel (refereegranskat)abstract
- Wideband communication is becoming a desired feature in telephone conferencing systems. This paper proposes a computationally efficient echo suppression control algorithm to be used when increasing the bandwidth of an audio conferencing system, e.g. a conference telephone. The method presented in this paper gives a quality improvement, in the form of increased bandwidth, at a negligible extra computational cost. The increase in bandwidth is obtained through combining a conventional acoustic echo cancellation unit and an acoustic echo suppression unit, i.e. a hybrid echo canceller and suppressor. The proposed solution was implemented in a real-time system. Frequency analysis combined with subjective tests showed that the proposed method extends the bandwidth, while maintaining high quality.
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| 7. |
- Lindström, Fredric, et al.
(författare)
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A Method for Reduced Finite Precision Effects in Parallel Filtering Echo Cancellation
- 2007
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Ingår i: IEEE transactions on circuits and systems I-Regular Papers. - : IEEE. - 1057-7122. ; 54:9, s. 2011-2018
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Tidskriftsartikel (refereegranskat)abstract
- The two-path algorithm is an adaptive filter algorithm based on a parallel filter structure, which has been found to be useful for line echo cancellation as well as for acoustic echo cancellation. It is well known that in finite precision arithmetic, the adaptation process of adaptive algorithms can be reduced or even halted due to finite precision effects. This paper proposes a variant of the two-path scheme where the effects of quantization are reduced, without any significant increase in complexity. The improvement is shown by simulations using bandlimited flat spectrum noise as well as real speech signals.
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| 8. |
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| 9. |
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| 10. |
- Lindström, Fredric, et al.
(författare)
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An Improvement of the Two-Path Algorithm Transfer Logic for Acoustic Echo Cancellation
- 2007
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Ingår i: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 15:4, s. 1320-1326
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Tidskriftsartikel (refereegranskat)abstract
- Adaptive filters for echo cancellation generally need update control schemes to avoid divergence in case of significant disturbances. The two-path algorithm avoids the problem of unnecessary halting of the adaptive filter when the control scheme gives an erroneous output. Versions of this algorithm have previously been presented for echo cancellation. This paper presents a transfer logic which improves the convergence speed of the two-path algorithm for acoustic echo cancellation, while retaining the robustness. Results from simulations show an improved performance, and a fixed-point DSP implementation verifies the performance in real-time
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| 11. |
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| 12. |
- Lindström, Fredric, et al.
(författare)
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Comparison of two methods of voice activity detection in field studies
- 2009
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Ingår i: Journal of Speech, Language and Hearing Research. - Rockville : American speech-language-hearing association. - 1092-4388 .- 1558-9102. ; 52:6, s. 1658-1663
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Tidskriftsartikel (refereegranskat)abstract
- Purpose: To evaluate and compare the performance of 2 methods of voice activity detection (neck-attached accelerometer vs. binaural recordings) in field studies in environments where voice activity normally occurs.Method: A group of 11 healthy adults wore recording equipment during their lunch break. We used binary classification to analyze the results from the 2 methods. The output was compared to a gold standard, obtained through listening tests, and the probability for sensitivity (Ps) and false positive (Pf) was rated. The binary classifiers were set for consistent sensitivity of 99%; thus, the lower false positive rate would indicate the method with the better performance.Results: The neck-attached accelerometer (Pf = 0.5%) performed significantly (p < .001) better than the binaural method (Pf = 7%).Conclusion: The neck-attached accelerometer is more suitable than the binaural method for voice assessments in environments where people are speaking in close proximity to each other and where the signal-to-noise ratio is moderate to low.
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| 13. |
- Lindström, Fredric
(författare)
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Digital signal processing methods and algorithms for audio conferencing systems
- 2007
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Doktorsavhandling (övrigt vetenskapligt/konstnärligt)abstract
- Today, we are interconnected almost all over the planet. Large multinational companies operate worldwide, but also an increasing number of small and medium sized companies do business overseas. As people travel to meet and do businesses, the already exposed earth is subject to even more strain. Audio conferencing is an attractive alternative to travel, which is becoming more and more appreciated. Audio conferences can of course not replace all types of meetings, but can help companies to cut travel costs and to reduce the environmental strain. The continuously growing market for audio conferencing systems proves that audio conferencing will play an important part in future communication solutions. This thesis treats digital signal processing methods and algorithms for single microphone audio conferencing systems. Concrete real problems, all in relation to audio conferencing systems, are discussed. An intrinsic problem to an audio conferencing system is the acoustic echoes picked up by the microphone. Acoustic echoes are generally cancelled using adaptive fi ltering. In such adaptive filter systems, a major difficulty is to achieve robustness in situations where both participants in a conversation are talking simultaneously. This thesis presents methods and solutions, focusing on the use of parallel adaptive fi lters, which provides the desired robustness. Audio conferencing systems are consumer electronic products and the manufacturing cost is a constant issue. Therefore, it is desirable to implement solutions on low-cost fi nite precision processors. A method to reduce fi nite precision effects in parallel filter implementations is presented in he thesis. In order to run algorithms on low-cost processors it is necessary to keep the computational complexity low. The thesis proposes a number of different methods to reduce complexity,including specific methods targeted for wideband solutions and systems equipped with extension microphones. A high quality audio conferencing system should be equipped with some sort of noise reduction feature. In the end of the thesis a method for integrating such noise reduction with the acoustic echo cancellation is presented. The performance of the proposed methods and algorithms are demonstrated through simulations as well as on real acoustic systems.
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| 14. |
- Lindström, Fredric, et al.
(författare)
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Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation
- 2007
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Ingår i: EURASIP Journal on Audio, Speech, and Music Processing. - : Hindawi Publishing Corporation. - 1687-4714 .- 1687-4722. ; 2007
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Tidskriftsartikel (refereegranskat)abstract
- An acoustic echo cancellation structure with a single loudspeaker and multiple microphones is, from a system identification perspective, generally modelled as a single-input multiple-output system. Such a system thus implies specific echo-path models (adaptive filter) for every loudspeaker to microphone path. Due to the often large dimensionality of the filters, which is required to model rooms with standard reverberation time, the adaptation process can be computationally demanding. This paper presents a selective updating normalized least mean square (NLMS)-based method which reduces complexity to nearly half in practical situations, while showing superior convergence speed performance as compared to conventional complexity reduction schemes. Moreover, the method concentrates the filter adaptation to the filter which is most misadjusted, which is a typically desired feature.
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| 15. |
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| 16. |
- Lindström, Fredric, et al.
(författare)
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On the Design of a Sound System for a Mobile Audio Unit
- 2005
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Konferensbidrag (refereegranskat)abstract
- A mobile audio unit is a wireless, battery-driven unit, the main purpose of which is to reproduce acoustic signals. This kind of unit can be used in conjunction with a home server. For example, a radio station broadcasting can be received from the Internet and fed to the mobile audio unit via a central home server. The market for home servers is expected to grow leading to a possible expansion of the market for this type of mobile audio unit. This paper presents some design aspects for the sound system of an audio unit, adapted to the new demands of the market.
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| 17. |
- Lindström, Fredric, et al.
(författare)
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Reusing Data During Speech Pauses in an NLMS-based Acoustic Echo Canceller
- 2006
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Konferensbidrag (refereegranskat)abstract
- Fast convergence of the adaptive filter in an acoustic echo cancellation based hands-free communication system is desirable as it implies more periods of possible full-duplex communication. This paper presents a normalized least mean square (NLMS)-based algorithm, targeted for acoustic echo cancellation based units equipped with large external memory. The proposed algorithm utilizes unused processing resources in periods of silence, thus no extra complexity as compared with the conventional NLMS algorithm is required. The improvements obtained by the proposed algorithm are verified through simulated, as well as through real acoustic systems.
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| 18. |
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| 19. |
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| 20. |
- Schüldt, Christian, et al.
(författare)
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A Distortion Reducing Subband Limiter Implementation for Conference Phones
- 2008
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Konferensbidrag (refereegranskat)abstract
- Distortion, most often caused by the loudspeaker, is a practical problem in acoustic echo cancellation-based conference phones. Typically, this distortion varies significantly with the frequency. This paper presents an implementation of a distortion reducing subband limiter to be used in conjunction with an acoustic echo canceller in a commercial conference phone, yielding improved cancellation performance. Verification of the performance was performed through evaluation of a fix-point real-time implementation.
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| 21. |
- Schüldt, Christian, et al.
(författare)
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A Low-Complexity Delayless Selective Subband Adaptive Filtering Algorithm
- 2008
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Ingår i: IEEE Transactions on Signal Processing. - 1053-587X .- 1941-0476. ; 56:12, s. 5840-5850
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Tidskriftsartikel (refereegranskat)abstract
- Adaptive filters of significant order, requiring high computational complexity, are necessary in many applications such as acoustic echo cancellation and wideband active noise control. Successful approaches to lessen the computational complexity of such filters are subband methods, and partial updating schemes where only a part of the filter is updated at each instant. To avoid the time delay introduced by the subband-splitting, delayless structures which reconstructs a fullband filter, producing delayless output, from the adaptive subband filters have been proposed. This paper proposes a delayless subband adaptive filter partial updating scheme, where the general idea is to only update the most misadjusted subband filter(s). Analysis in terms of mean square deviation is presented and shows that the fullband filter convergence speed is significantly increased, even for flat spectrum signals, as compared to traditional periodic subband filter update with the same computational complexity. Echo cancellation simulations with an artificial system to verify the analysis, using both flat spectrum signals and speech, is also presented, as well as offline calculations using signals from a real system.
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| 22. |
- Schüldt, Christian, et al.
(författare)
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Adaptive filter length selection for acoustic echo cancellation
- 2009
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Ingår i: Signal Processing. - AMSTERDAM : Elsevier. - 0165-1684 .- 1872-7557. ; 89:6, s. 1185-1194
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Tidskriftsartikel (refereegranskat)abstract
- The number of coefficients in an adaptive finite impulse response filter-based acoustic echo cancellation setup is an important parameter, affecting the overall performance of the echo cancellation. Too few coefficients give undermodelling and too many cause slow convergence and an additional echo due to the mismatch of the extra coefficients. This paper proposes a method to adaptively determine the filter length, based on estimation of the mean square deviation. The method is primarily intended for identifying long non-sparse systems, such as a typical impulse response from an acoustic setup. Simulations with band limited flat spectrum signals are used for verification, showing the behavior and benefits of the proposed algorithm. Furthermore, off-line calculation using recorded speech signals show the behavior in real situations and comparison with another state-of-the-art variable filter length algorithm shows the advantages of the proposed method.
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| 23. |
- Schüldt, Christian, et al.
(författare)
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Low-Complexity Adaptive Filtering Implementation for Acoustic Echo Cancellation
- 2006
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Konferensbidrag (refereegranskat)abstract
- Acoustic echo cancellation is generally achieved with adaptive FIR filters. Due to the often large dimensionality of the adaptive filters, required to model rooms with standard reverberation time, the adaptation process can be computationally demanding. This paper presents a block based selective updating method which reduces the complexity with nearly a half in practical situations, while showing superior convergence speed performance as compared to conventional partial update complexity reduction schemes.
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| 24. |
- Wiklunda, Kenneth, et al.
(författare)
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EVALUATION OF A HANDS-FREE UNIT DURING DOUBLE-TALK
- 2005
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Konferensbidrag (refereegranskat)abstract
- The double-talk performance of a handsfree speech communication system is crucial to the overall quality. This paper presents a method to evaluate double-talk performance of conference phones and speakerphones. The method measures loudness, clipping and the amount of echo during double-talk.
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