| 1. |
- Ahlin, Kjell, et al.
(författare)
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Comparing Road Profiles with Vehicle Perceived Roughness
- 2004
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Ingår i: International Journal of Vehicle Design. - Geneva, Switzerland : Inderscience Enterprises. - 0143-3369 .- 1741-5314. ; 36:2-3, s. 270-286
-
Tidskriftsartikel (refereegranskat)abstract
- Accurate road profiles are useful in vehicle design, such as for simulation of durability and ride quality. Laser/inertial profilometers typically record I mm wide profiles. The question is how well such a profile matches perceived vehicle wheel roughness. The objective here was to create a more representative wheel track longitudinal profile. Simulated and measured wheel vibration was compared on a 6km long road. Simulations were made for several definitions of the profile. Results for single laser sensor profiles showed reasonable likeness to truck perceived roughness. By far the best likeness (14.5% better) was achieved when the profile was based on triangular 25%-50%-25% weighted data from three sensors in the wheel track. Clearly, vehicle engineers can benefit from using multiple laser profile sensors, instead of a single sensor. This will improve test accuracy, thus reducing vehicle design project lead times and costs.
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| 2. |
- Bengtsson, Ylva, et al.
(författare)
-
Ulriksdals slott under 350 år
- 1995
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Rapport (populärvet., debatt m.m.)abstract
- Studierna har utgått från Ulriksdals slott, uppfört av Jacob De la Gardie för 350 år sedan. Som kungligt slott har Ulriksdal genomgått ombyggnader under 1600 - och 1700-talet, men präglas idag mest av Karl XVs omfattande omdaning på 1850-talet och Gustav VI Adolfs lika genomgripande förändringar på 1920-talet.Slottet har bjudit på rika studiemöjligheter av stilideal, planlösning, interiörgestaltning, snickerikultur och stormtekniska system från hela 350-årsperioden. Genom noggranna uppmätningar och inventeringar på plats kopplade till arkivstudier har slottets komplicerade förändringshistoria kunnat analyseras,
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| 3. |
- Berggren, Magnus, et al.
(författare)
-
Analysis of How the Noise Level Depends on Different Activities in a Child Day-Care Center
- 2008
-
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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| 4. |
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| 5. |
- Berggren, Magnus, et al.
(författare)
-
Audio Processing Solution for Video Conference Based Aerobics
- 2010
-
Ingår i: 2010 Digest of Technical Papers, International Conference on Consumer Electronics. - Las Vegas : IEEE. - 9781424443147 - 9781424443161 ; , s. 407-408
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Konferensbidrag (refereegranskat)abstract
- In this paper an audio processing solution for video conference based aerobics is presented. The proposed solution leaves the workout music unaltered by separating it from the speech and processing each signal separately. The speech signal processing is also performed at a lower sample rate, which saves computational power. Real time evaluation of the system shows that high quality music as well as a good two-way communication is maintained during the aerobic session.
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| 6. |
- Berggren, Magnus, et al.
(författare)
-
Low-complexity network echo cancellation approach for systems equipped with external memory
- 2011
-
Ingår i: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 19:8, s. 2506-2515
-
Tidskriftsartikel (refereegranskat)abstract
- 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.
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| 7. |
- Borgh, Markus, 1983-, et al.
(författare)
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A Personal Voice Analyzer and Trainer
- 2010
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Konferensbidrag (refereegranskat)abstract
- 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.
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| 8. |
- Borgh, Markus, 1983-, et al.
(författare)
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An improved adaptive gain equalizer for noise reduction with low speech distortion
- 2011
-
Ingår i: EURASIP Journal on Audio, Speech, and Music Processing. - : Springer. - 1687-4714 .- 1687-4722. ; 7
-
Tidskriftsartikel (refereegranskat)abstract
- 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.
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| 9. |
- 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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| 10. |
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| 11. |
- Lindroth, Markus, 1983-
(författare)
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Low-Complexity Signal Processing for Speech Enhancement and Audio Analysis
- 2022
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Licentiatavhandling (övrigt vetenskapligt/konstnärligt)abstract
- 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.
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| 12. |
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| 13. |
- Lindström, Fredric, et al.
(författare)
-
A Computational Efficient Method for Assuring Full Duplex Feeling in Hands-free Communication
- 2003
-
Konferensbidrag (refereegranskat)abstract
- This report proposes a method for obtaining satisfying "full-duplex feeling" in hands-free communication units at low computational cost. The proposed method uses a combination of an acoustic echo cancellation unit and an adaptive gain unit. The core of the method is to perform the processing of the speech signal into two separate frequency bands and to process these in different manners. Acoustic echoes in the low frequency part of the signal are cancelled by means of an acoustic echo cancellation unit, while acoustic echoes in the high frequency part are suppressed by an adaptive gain unit.
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| 14. |
- Lindström, Fredric, et al.
(författare)
-
A Computational Efficient Method for Assuring Full Duplex Feeling in Hands-free Communication
- 2003
-
Rapport (övrigt vetenskapligt/konstnärligt)abstract
- This report proposes a method for obtaining satisfying "full-duplex feeling" in hands-free communication units at low computational cost. The proposed method uses a combination of an acoustic echo cancellation unit and an adaptive gain unit. The core of the method is to perform the processing of the speech signal into two separate frequency bands and to process these in different manners. Acoustic echoes in the low frequency part of the signal are cancelled by means of an acoustic echo cancellation unit, while acoustic echoes in the high frequency part are suppressed by an adaptive gain unit. The proposed method is well suited when extending the bandwidth of an existing hands-free phone. A real-time implementation of a conventional hands-free phone is compared with a real-time implementation according to the proposed method, where the later is an extended version of the first. The evaluation of the two implementations shows that the proposed method can be used to increase the quality, i.e. extended bandwidth, of a hands-free phone with only a small increase in computational demand.
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| 15. |
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| 16. |
- Lindström, Fredric, et al.
(författare)
-
A Computational Efficient Method For Bandwidth Extension of a Conference Phone
- 2003
-
Konferensbidrag (refereegranskat)abstract
- This paper presents a computationally efficient method for extension of the bandwidth of a conference telephone. The proposed method allows an improvement in quality, i.e. increased bandwidth, at a negligible extra computational cost. This is performed by a combination of an acoustic echo cancellation unit and an adaptive gain unit. The proposed method was implemented in a real-time system. Frequency analysis in combination with subjective tests showed that the proposed method extends the bandwidth with high quality.
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| 17. |
- Lindström, Fredric, et al.
(författare)
-
A Finite Precision LMS Algorithm for Increased Quantization Robustness
- 2003
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Konferensbidrag (refereegranskat)abstract
- The well known Least Mean Square (LMS) algorithm, or variations thereof are frequently used in adaptive systems. When the LMS algorithm is implemented in a finite precision environment it suffers from quantization effects. These effects can severely degrade the performance of the algorithm. This paper proposes a modification of the LMS algorithm that reduces the impact of quantization at virtually no extra computational cost. The paper contains an off-line evaluation of a system identification scheme where the presented algorithm outperforms the classical LMS algorithm yielding a better modelling of the unknown plant. This approach is well suited for adaptive system identification, e.g. beam-forming, electrocardiography, and echo cancelling.
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| 18. |
- Lindström, Fredric, et al.
(författare)
-
A hybrid acoustic echo canceller and suppressor
- 2007
-
Ingår i: Signal Processing. - : Elsevier. - 0165-1684. ; 87:4, s. 739-749
-
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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| 19. |
- Lindström, Fredric, et al.
(författare)
-
A Method for Reduced Finite Precision Effects in Parallel Filtering Echo Cancellation
- 2007
-
Ingår i: IEEE transactions on circuits and systems I-Regular Papers. - : IEEE. - 1057-7122. ; 54:9, s. 2011-2018
-
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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| 20. |
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| 21. |
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| 22. |
- Lindström, Fredric, et al.
(författare)
-
An Improvement of the Two-Path Algorithm Transfer Logic for Acoustic Echo Cancellation
- 2007
-
Ingår i: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 15:4, s. 1320-1326
-
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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| 23. |
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| 24. |
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| 25. |
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| 26. |
- Lindström, Fredric, et al.
(författare)
-
Comparison of two methods of voice activity detection in field studies
- 2009
-
Ingår i: Journal of Speech, Language and Hearing Research. - Rockville : American speech-language-hearing association. - 1092-4388 .- 1558-9102. ; 52:6, s. 1658-1663
-
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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| 27. |
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| 28. |
- Lindström, Fredric
(författare)
-
Digital signal processing methods and algorithms for audio conferencing systems
- 2007
-
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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| 29. |
- Lindström, Fredric, et al.
(författare)
-
Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation
- 2007
-
Ingår i: EURASIP Journal on Audio, Speech, and Music Processing. - : Hindawi Publishing Corporation. - 1687-4714 .- 1687-4722. ; 2007
-
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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| 30. |
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| 31. |
- Lindström, Fredric, et al.
(författare)
-
Long-term measurements of sound levels in child day-care centers
- 2010
-
Ingår i: 39th International Congress on Noise Control Engineering 2010 (INTER-NOISE 2010). - : Sociedade Portuguesa de Acustica. - 9781617823961 ; , s. 2992-3001
-
Konferensbidrag (refereegranskat)abstract
- Over the years numerous measurements of the sound levels in Swedish child care centers have been performed. However, most of these measurements consist of short one day measurements sessions. In this study we have measured the sound levels continuously during the whole day for several months in a row. Stationary sound level meters were used to measure the sound. In the study 6 day-care centers, with about 15 children in each, participated and for each center measurements were done in 4 different rooms, i.e. The sound levels in 24 rooms were measured continuously. Analysis of data show a wide spread between the different rooms and in time, which demonstrate the uncertainty in using single location short time, e.g. one day, measurements to assess the long-term sound levels in child day-care centers. The paper further demonstrates that long-term measurements can be used to identify e.g. differences between rooms, differences over time, etc, and that these differences can evoke questions about the activities generating them.
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| 32. |
- Lindström, Fredric, 1974, et al.
(författare)
-
Mean F0 values obtained through standard phrase pronunciation compared with values obtained from the normal work environment: A study on teacher and child voices performed in a pre-school environment
- 2010
-
Ingår i: Journal of Voice. - 0892-1997. ; 24:3, s. 319-323
-
Tidskriftsartikel (refereegranskat)abstract
- Mean fundamental frequency (F0) values are often used in research on vocal load. In this study, we examine how the mean F0 differs when evaluated through pronouncing a standard phrase as compared to the mean F0 obtained in a real work/play environment. We also examine how the F0 values change throughout the day. The study was performed in a preschool, nine adult female preschool teachers and 11 children participated. The participants wore a digital recorder equipped with an accelerometer, which was attached to the neck. In the study, the participant first pronounced a standard phrase in a controlled environment; thereafter, the voice was recorded in the environment where both children and adults normally reside throughout the day, denoted by the work/play environment. For each participant, the procedure was repeated four times throughout the day. Analyses showed that the F0 values of the children's and adult's voices were significantly higher when recorded in the work/play environment as compared to the controlled environment. The average difference was 36 Hz for adults and 24 Hz for children. Previous studies have shown an increase of F0 over the day for teachers. In this study, an increase between morning and afternoon values was found amounting to 8 Hz for adults and 24 Hz for children. For the child population, this increase was statistically significant. However, the total changes over the day revealed a somewhat more complex scheme, with an increase of F0 in the morning, a decrease during lunch, and finally an increase in the afternoon. This pattern was verified statistically for the joint child-adult population.
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| 33. |
- Lindström, Fredric, 1974, et al.
(författare)
-
Observations of the Relationship Between Noise Exposure and Preschool Teacher Voice Usage in Day-Care Center Environments
- 2011
-
Ingår i: Journal of Voice. - : Elsevier BV. - 0892-1997 .- 1873-4588. ; 25:2, s. 166-172
-
Tidskriftsartikel (refereegranskat)abstract
- Although the relationship between noise exposure and vocal behavior (the Lombard effect) is well established, actual vocal behavior in the workplace is still relatively unexamined. The first purpose of this study was to investigate correlations between noise level and both voice level and voice average fundamental frequency (F-0) for a population of preschool teachers in their normal workplace. The second purpose was to study the vocal behavior of each teacher to investigate whether individual vocal behaviors or certain patterns could be identified. Voice and noise data were obtained for female preschool teachers (n = 13) in their workplace, using wearable measurement equipment. Correlations between noise level and voice level, and between voice level and F-0, were calculated for each participant and ranged from 0.07 to 0.87 for voice level and from 0.11 to 0.78 for F-0. The large spread of the correlation coefficients indicates that the teachers react individually to the noise exposure. For example, some teachers increase their voice-to-noise level ratio when the noise is reduced, whereas others do not.
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| 34. |
- Lindström, Fredric, et al.
(författare)
-
On audio hands-free design
- 2004
-
Konferensbidrag (refereegranskat)abstract
- High-quality audio hands-free systems involve rather complex signal processing. The development of such a system is not a straightforward task. This paper proposes a step-by-step approach to the design and implementation of an audio hands-free system. The proposed design method facilitates the implementation process and leads to a robust audio hands-free system solution. The paper also provides an overview of the problems encountered when designing an audio hands-free system. State-of-the-art solutions as well as recently proposed solutions are referred to in addition to the hands-free system design problems.
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| 35. |
- Lindström, Fredric, et al.
(författare)
-
On the Design of a Sound System for a Mobile Audio Unit
- 2005
-
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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| 36. |
- Lindström, Fredric, et al.
(författare)
-
Reusing Data During Speech Pauses in an NLMS-based Acoustic Echo Canceller
- 2006
-
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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| 37. |
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| 38. |
- Lindström, Fredric, et al.
(författare)
-
The two-path algorithm for line echo cancellation
- 2004
-
Konferensbidrag (refereegranskat)abstract
- The two-path algorithm is an algorithm for line echo cancellation based on two parallel filters. This paper proposes a modification of the two-path algorithm that improves its performance. In the two-path algorithm a background filter is used for continuously adaptive estimation of the line echo, while a foreground filter is used for the actual cancellation. The coefficients of the background filter are copied into the foreground filter when the background filter is proven to perform better. A robust algorithm for line echo cancellation is thereby achieved. In this paper, the benefits and the drawbacks of the two-path algorithm are evaluated and demonstrated through simulations. A modification is proposed that reduces the negative effects of the two-path algorithm. This modification is compared to the original two-path algorithm. Simulations using real speech signals indicate that the proposed modification can improve the performance of the two-path algorithm. © 2004IEEE.
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| 39. |
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| 40. |
- Moberg, Christina, et al.
(författare)
-
De unga gör helt rätt när de stämmer staten
- 2022
-
Ingår i: Aftonbladet. - 1103-9000.
-
Tidskriftsartikel (populärvet., debatt m.m.)abstract
- 1 620 forskare och lärare i forskarvärlden: Vi ställer oss bakom Auroras klimatkrav
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| 41. |
- Nygren, M., et al.
(författare)
-
Gender Differences in Children's Voice Use in a Day Care Environment
- 2012
-
Ingår i: Journal of Voice. - : Elsevier BV. - 0892-1997 .- 1873-4588. ; 26:6
-
Tidskriftsartikel (refereegranskat)abstract
- Background. The prevalence of dysphonia is higher in boys than in girls before puberty. This could be because of the differences in boys' and girls' voice use. Previous research on gender differences in prepubescent children's voice parameters has been contradictory. Most studies have focused on examining fundamental frequency. Objectives. The purpose of this study was to investigate voice use in boys and girls in a day care environment based on the voice parameters fundamental frequency (Hz), vocal intensity (dB SPL), and phonation time (%) and to ascertain whether there were any significant gender differences. Study Design. Prospective comparative design. Method. The study was conducted in a day care environment where 30 children (17 boys and 13 girls aged 4-5 years) participated. The participants' voices were measured continuously for 4 hours with a voice accumulator that registered fundamental frequency, vocal intensity level, phonation time, and background noise. Results. Mean (standard deviation) fundamental frequency was 310 (22) and 321 (16) Hz, vocal intensity was 93 (4) and 91 (3) dB SPL, and phonation time was 7.7 (2.0)% and 7.6 (2.5)% for boys and girls, respectively. No differences between genders were statistically significant. Conclusion. The finding of no statistically significant gender differences for measurements of voice parameters in a group of children aged 4-5 years in a day care environment is an important finding that contributes to increased knowledge about young boys' and girls' voice use.
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| 42. |
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| 43. |
- Persson Waye, Kerstin, et al.
(författare)
-
Being in a pre-school sound environment - Annoyance and subjective symptoms among personnel and children
- 2010
-
Ingår i: 39th International Congress on Noise Control Engineering 2010 (INTER-NOISE 2010). - : Sociedade Portuguesa de Acustica. - 9781617823961 ; , s. 1026-1032
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Konferensbidrag (refereegranskat)abstract
- Pre-school noise may be a serious occupational and public health problem. The equivalent sound levels are high, the exposure time long and the sound environment highly intermittent with maximum levels in the range of 110-115 dB. In order to elucidate how children and personnel experience their pre-school sound environment, a cross sectional study was carried out. Interview data was obtained from 63 children from seven pre-schools and questionnaire data from 187 personnel from 67 pre-schools. The results showed that as many as 59% of the personnel reported being rather, very or extremely annoyed by noise. More than 50% of the children reported loud and angering/screaming sounds often or very often. They coped by avoidance, holding their ears and/or telling the teacher. The study gives clear indications of a loud and often disturbing pre-school sound environment and the health consequences need to be furthered assessed.
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| 44. |
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| 45. |
- Schüldt, Christian, et al.
(författare)
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A Delay-Based Double-Talk Detector
- 2012
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Ingår i: IEEE Transactions on Audio, Speech, and Language Processing. - 1558-7916 .- 1558-7924. ; 20:6, s. 1725-1733
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Tidskriftsartikel (refereegranskat)abstract
- When an adaptive filter is used for echo cancellation, it is essential to prevent the filter from diverging in situations when the echo signal is contaminated with near-end disturbance, i.e. during double-talk. This paper presents an extension of a previously proposed double-talk detector for improved performance. It is shown that the computational complexity of the proposed detector is lower than that of the well-used normalized cross correlation (NCC) double-talk detector, at the cost of performance. Further, it is shown that there can be a significant performance difference, in terms of detecting double-talk, between having a fixed echo cancellation filter, which is a common strategy in objective evaluation techniques, and an adaptive filter, which is more close to realistic conditions.
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| 46. |
- 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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| 47. |
- 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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| 48. |
- 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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| 49. |
- Schüldt, Christian, et al.
(författare)
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An Improved Deviation Measure for Two-Path Echo Cancellation
- 2010
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Konferensbidrag (refereegranskat)abstract
- Parallel adaptive filters have been proposed for echo cancellation to solve the dead-lock problem, occurring when the echo is detected as near-end speech after a severe echo-path change; causing the updating of the adaptive filter to halt. To control the parallel filters and monitor their performance, estimates of the filter deviation (i.e. the squared norm of the filter mismatch vector) are typically used. This paper presents a modification of a filter mismatch estimator. The proposed modification requires slightly more computational resources than the original measure, but provides a significantn improvement in terms of robustness during double-talk. This is shown both analytically and through simulations.
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| 50. |
- Schüldt, Christian, et al.
(författare)
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Evaluation of an Improved Deviation Measure for Two-Path Echo Cancellation
- 2010
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Konferensbidrag (refereegranskat)abstract
- The two-path algorithm is a well-known approach for overcoming the dead-lock problem in echo cancellation systems. Typically, a fixed foreground filter is producing the echo cancelled output while a continuously updating background filter adapts to the echo-path. When the background filter is considered to perform better than the foreground filter, the coefficients of the background filter are copied into the foreground filter. To determine which filter is better adjusted to the true echo-path, a filter deviation measure can be used. Recently, a method which introduces a delay in the calculation of the filter deviation measure, yielding a more reliable estimate has been proposed. However, a thorough evaluation of the effect of different delay settings has not yet been performed. Thus, in this paper a number of simulations with different delay parameter settings are carried out to show how this parameter affects the overall performance of the filter deviation measure.
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