| 1. |
- Kattge, Jens, et al.
(author)
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TRY plant trait database - enhanced coverage and open access
- 2020
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In: Global Change Biology. - : Wiley-Blackwell. - 1354-1013 .- 1365-2486. ; 26:1, s. 119-188
-
Journal article (peer-reviewed)abstract
- Plant traits-the morphological, anatomical, physiological, biochemical and phenological characteristics of plants-determine how plants respond to environmental factors, affect other trophic levels, and influence ecosystem properties and their benefits and detriments to people. Plant trait data thus represent the basis for a vast area of research spanning from evolutionary biology, community and functional ecology, to biodiversity conservation, ecosystem and landscape management, restoration, biogeography and earth system modelling. Since its foundation in 2007, the TRY database of plant traits has grown continuously. It now provides unprecedented data coverage under an open access data policy and is the main plant trait database used by the research community worldwide. Increasingly, the TRY database also supports new frontiers of trait-based plant research, including the identification of data gaps and the subsequent mobilization or measurement of new data. To support this development, in this article we evaluate the extent of the trait data compiled in TRY and analyse emerging patterns of data coverage and representativeness. Best species coverage is achieved for categorical traits-almost complete coverage for 'plant growth form'. However, most traits relevant for ecology and vegetation modelling are characterized by continuous intraspecific variation and trait-environmental relationships. These traits have to be measured on individual plants in their respective environment. Despite unprecedented data coverage, we observe a humbling lack of completeness and representativeness of these continuous traits in many aspects. We, therefore, conclude that reducing data gaps and biases in the TRY database remains a key challenge and requires a coordinated approach to data mobilization and trait measurements. This can only be achieved in collaboration with other initiatives.
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| 2. |
- Berggren, Magnus, et al.
(author)
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Low-complexity network echo cancellation approach for systems equipped with external memory
- 2011
-
In: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 19:8, s. 2506-2515
-
Journal article (peer-reviewed)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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| 3. |
- Borgh, Markus, 1983-, et al.
(author)
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An improved adaptive gain equalizer for noise reduction with low speech distortion
- 2011
-
In: EURASIP Journal on Audio, Speech, and Music Processing. - : Springer. - 1687-4714 .- 1687-4722. ; 7
-
Journal article (peer-reviewed)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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| 4. |
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| 5. |
- Cumlin, Fredrik, et al.
(author)
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Latent-based Neural Net for Non-intrusive Speech Quality Assessment
- 2023
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In: 31st European Signal Processing Conference, EUSIPCO 2023 - Proceedings. - : European Signal Processing Conference, EUSIPCO. ; , s. 226-230
-
Conference paper (peer-reviewed)abstract
- For non-intrusive speech quality assessment, we treat the mean-opinion-score (MOS) of a speech signal as a latent, and propose a latent MOS network (LaMOSNet) to estimate the MOS. At the time of training, the proposed LaMOSNet has two parts in series, with the first part providing the latent estimate, i.e. the MOS of an input speech signal, and the second part providing an estimated score by a given judge. Only the first part is used for testing. We address two inherent aspects - limited-data and noisy-data aspects - in training using stochastic gradient noise and a student-teacher type of training, motivated by semi-supervised learning. It is shown that LaMOSNet provides good performance on the Voice Conversion Challenge 2018 dataset, and state-of-the-art correlation performance on the Voice Conversion Challenge 2016 dataset.
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| 6. |
- Laptev, Ivan, et al.
(author)
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Local velocity-adapted motion events for spatio-temporal recognition
- 2007
-
In: Computer Vision and Image Understanding. - : Elsevier. - 1077-3142 .- 1090-235X. ; 108:3, s. 207-229
-
Journal article (peer-reviewed)abstract
- In this paper, we address the problem of motion recognition using event-based local motion representations. We assume that similar patterns of motion contain similar events with consistent motion across image sequences. Using this assumption, we formulate the problem of motion recognition as a matching of corresponding events in image sequences. To enable the matching, we present and evaluate a set of motion descriptors that exploit the spatial and the temporal coherence of motion measurements between corresponding events in image sequences. As the motion measurements may depend on the relative motion of the camera, we also present a mechanism for local velocity adaptation of events and evaluate its influence when recognizing image sequences subjected to different camera motions. When recognizing motion patterns, we compare the performance of a nearest neighbor (NN) classifier with the performance of a support vector machine (SVM). We also compare event-based motion representations to motion representations in terms of global histograms. A systematic experimental evaluation on a large video database with human actions demonstrates that (i) local spatio-temporal image descriptors can be defined to carry important information of space-time events for subsequent recognition, and that (ii) local velocity adaptation is an important mechanism in situations when the relative motion between the camera and the interesting events in the scene is unknown. The particular advantage of event-based representations and velocity adaptation is further emphasized when recognizing human actions in unconstrained scenes with complex and non-stationary backgrounds.
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| 7. |
- Liang, Xinyue, et al.
(author)
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DeePMOS : Deep Posterior Mean-Opinion-Score of Speech
- 2023
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In: Interspeech 2023. - : International Speech Communication Association. ; , s. 526-530
-
Conference paper (peer-reviewed)abstract
- We propose a deep neural network (DNN) based method that provides a posterior distribution of mean-opinion-score (MOS) for an input speech signal. The DNN outputs parameters of the posterior, mainly the posterior's mean and variance. The proposed method is referred to as deep posterior MOS (DeePMOS). The relevant training data is inherently limited in size (limited number of labeled samples) and noisy due to the subjective nature of human listeners. For robust training of DeePMOS, we use a combination of maximum-likelihood learning, stochastic gradient noise, and a student-teacher learning setup. Using the mean of the posterior as a point estimate, we evaluate standard performance measures of the proposed DeePMOS. The results show comparable performance with existing DNN-based methods that only provide point estimates of the MOS. Then we provide an ablation study showing the importance of various components in DeePMOS.
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| 8. |
- Lindström, Fredric, et al.
(author)
-
A hybrid acoustic echo canceller and suppressor
- 2007
-
In: Signal Processing. - : Elsevier. - 0165-1684. ; 87:4, s. 739-749
-
Journal article (peer-reviewed)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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| 9. |
- Lindström, Fredric, et al.
(author)
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A Method for Reduced Finite Precision Effects in Parallel Filtering Echo Cancellation
- 2007
-
In: IEEE transactions on circuits and systems I-Regular Papers. - : IEEE. - 1057-7122. ; 54:9, s. 2011-2018
-
Journal article (peer-reviewed)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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| 10. |
- Lindström, Fredric, et al.
(author)
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An Improvement of the Two-Path Algorithm Transfer Logic for Acoustic Echo Cancellation
- 2007
-
In: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 15:4, s. 1320-1326
-
Journal article (peer-reviewed)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. |
- Lindström, Fredric, et al.
(author)
-
Efficient Multichannel NLMS Implementation for Acoustic Echo Cancellation
- 2007
-
In: EURASIP Journal on Audio, Speech, and Music Processing. - : Hindawi Publishing Corporation. - 1687-4714 .- 1687-4722. ; 2007
-
Journal article (peer-reviewed)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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| 12. |
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| 13. |
- Lindström, Fredric, et al.
(author)
-
Reusing Data During Speech Pauses in an NLMS-based Acoustic Echo Canceller
- 2006
-
Conference paper (peer-reviewed)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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| 14. |
- Nilsson, John-Olof, et al.
(author)
-
Voice radio communication, pedestrian localization, and the tactical use of 3D audio
- 2013
-
In: 2013 International Conference on Indoor Positioning and Indoor Navigation, IPIN 2013. - : IEEE Computer Society. - 9781479940431 ; , s. 6817918-
-
Conference paper (peer-reviewed)abstract
- The relation between voice radio communication and pedestrian localization is studied. 3D audio is identified as a linking technology which brings strong mutual benefits. Voice communication rendered with 3D audio provides a potential low secondary task interference user interface to the localization information. Vice versa, location information in the 3D audio provides spatial cues in the voice communication, improving speech intelligibility. An experimental setup with voice radio communication, cooperative pedestrian localization, and 3D audio is presented and we discuss high level tactical possibilities that the 3D audio brings. Finally, results of an initial experiment, demonstrating the effectiveness of the setup, are presented.
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| 15. |
- Sai, Hanna, et al.
(author)
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Observations of the very young Type Ia Supernova 2019np with early-excess emission
- 2022
-
In: Monthly notices of the Royal Astronomical Society. - : Oxford University Press (OUP). - 0035-8711 .- 1365-2966. ; 514:3, s. 3541-3558
-
Journal article (peer-reviewed)abstract
- Early-time radiative signals from Type Ia supernovae (SNe Ia) can provide important constraints on the explosion mechanism and the progenitor system. We present observations and analysis of SN 2019np, a nearby SN Ia discovered within 1–2 days after the explosion. Follow-up observations were conducted in optical, ultraviolet, and near-infrared bands, covering the phases from ∼−16.7 d to ∼+ 367.8 d relative to its B-band peak luminosity. The photometric and spectral evolutions of SN 2019np resemble the average behaviour of normal SNe Ia. The absolute B-band peak magnitude and the post-peak decline rate are Mmax(B) = −19.52 ± 0.47 mag and Δm15(B) = 1.04 ± 0.04 mag, respectively. No Hydrogen line has been detected in the nebular-phase spectra of SN 2019np. Assuming that the 56Ni powering the light curve is centrally located, we find that the bolometric light curve of SN 2019np shows a flux excess up to 5.0 per cent in the early phase compared to the radiative diffusion model. Such an extra radiation perhaps suggests the presence of an additional energy source beyond the radioactive decay of central nickel. Comparing the observed colour evolution with that predicted by different models, such as interactions of SN ejecta with circumstellar matter (CSM)/companion star, a double-detonation explosion from a sub-Chandrasekhar mass white dwarf (WD) and surface 56Ni mixing, we propose that the nickel mixing is more favoured for SN 2019np.
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| 16. |
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| 17. |
- Schüldt, Christian, et al.
(author)
-
A Delay-Based Double-Talk Detector
- 2012
-
In: IEEE Transactions on Audio, Speech, and Language Processing. - 1558-7916 .- 1558-7924. ; 20:6, s. 1725-1733
-
Journal article (peer-reviewed)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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| 18. |
- Schüldt, Christian, et al.
(author)
-
A Distortion Reducing Subband Limiter Implementation for Conference Phones
- 2008
-
Conference paper (peer-reviewed)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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| 19. |
- Schüldt, Christian, et al.
(author)
-
A Low-Complexity Delayless Selective Subband Adaptive Filtering Algorithm
- 2008
-
In: IEEE Transactions on Signal Processing. - 1053-587X .- 1941-0476. ; 56:12, s. 5840-5850
-
Journal article (peer-reviewed)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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| 20. |
- Schüldt, Christian, et al.
(author)
-
Adaptive filter length selection for acoustic echo cancellation
- 2009
-
In: Signal Processing. - AMSTERDAM : Elsevier. - 0165-1684 .- 1872-7557. ; 89:6, s. 1185-1194
-
Journal article (peer-reviewed)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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| 21. |
- Schüldt, Christian, et al.
(author)
-
An Improved Deviation Measure for Two-Path Echo Cancellation
- 2010
-
Conference paper (peer-reviewed)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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| 22. |
- Schüldt, Christian, 1978-, et al.
(author)
-
Blind low-complexity estimation of reverberation time
- 2013
-
In: 2013 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA). - : IEEE conference proceedings. - 9781479909728 ; , s. 6701875-
-
Conference paper (peer-reviewed)abstract
- Real-time blind reverberation time estimation is of interest in speech enhancement techniques such as e.g. dereverberation and microphone beamforming. Advances in this field have been made where the diffusive reverberation tail is modeled and the decay rate is estimated using a maximum-likelihood approach. Various methods for reducing the computational complexity have also been presented. This paper proposes a method for even further computational complexity reduction, by more than 60% in some cases, and it is shown through simulations that the results of the proposed method are very similar to that of the original.
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| 23. |
- Schüldt, Christian, 1978-, et al.
(author)
-
Decay Rate Estimators and Their Performance for Blind Reverberation Time Estimation
- 2014
-
In: IEEE/ACM Transactions on Audio, Speech, and Language Processing. - 2329-9290. ; 22:8, s. 1274-1284
-
Journal article (peer-reviewed)abstract
- Several approaches for blind estimation of reverberation time have been presented in the literature and decay rate estimation is an integral part of many, if not all, of such approaches. This paper provides both an analytical and experimental comparison, in terms of the bias and variance of three common decay rate estimators; a straight-forward linear regression approach as well as two maximum-likelihood based methods. Situations with and without interfering additive noise are considered. It is shown that the linear regression based approach is unbiased if no smoothing is applied, and that the estimation variance in the absence of noise is constantly about twice that of the maximum-likelihood based methods. It is shown that the methods that do not take possible noise into account suffer from similar estimation bias in the presence of noise. Further, a hybrid method, combining the noise robustness and low computational complexity advantages of the two different maximum-likelihood based methods, is presented.
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| 24. |
- Schüldt, Christian, et al.
(author)
-
Evaluation of an Improved Deviation Measure for Two-Path Echo Cancellation
- 2010
-
Conference paper (peer-reviewed)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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| 25. |
- Schüldt, Christian
(author)
-
Low-Complexity Adaptive Filtering for Acoustic Echo Cancellation in Audio Conferencing Systems
- 2009
-
Licentiate thesis (other academic/artistic)abstract
- With the globalization of the world’s economy, the demand for effortless, quick and efficient communication is increasing. Modern audio conferencing allows people at different locations to have a conversation as if they were sitting in the same room, without having to travel. This obviously saves time and money, and also lessens the environmental strain caused by travel. Most audio conferencing systems and hands-free systems in particular, suffer from electric and/or acoustic echoes. Electric echoes typically originate from 2-4 wire conversion in hybrid circuits in the telephone network, while acoustic echoes arise due to acoustic coupling between loudspeaker and microphone. In digital audio communication equipment, the echoes are usually removed through digital signal processing methods such as adaptive filtering. Since audio conferencing systems are consumer electronic products, the manufacturing cost is a key issue. In order to accomplish low manufacturing costs, the choice of a low cost digital signal processor (DSP) to perform the signal processing tasks is central. Further, due to the limited resources of low cost DSPs, there is an intrinsic demand for low complexity signal processing algorithms. This thesis presents low complexity algorithms for adaptive filtering in acoustic echo cancellation applications. Both the actual update of the adaptive filter and the update control to prevent divergence and so called howling, are considered. Computer simulations, as well as real time implementations in actual acoustic systems are used to verify the performance of the proposed algorithms.
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| 26. |
- Schüldt, Christian, et al.
(author)
-
Low-Complexity Adaptive Filtering Implementation for Acoustic Echo Cancellation
- 2006
-
Conference paper (peer-reviewed)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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| 27. |
- Schüldt, Christian
(author)
-
Low-Complexity Algorithms for Echo Cancellation in Audio Conferencing Systems
- 2012
-
Doctoral thesis (other academic/artistic)abstract
- Ever since the birth of the telephony system, the problem with echoes, arising from impedance mismatch in 2/4-wire hybrids, or acoustic echoes where a loudspeaker signal is picked up by a closely located microphone, has been ever present. The removal of these echoes is crucial in order to achieve an acceptable audio quality for conversation. Today, the perhaps most common way for echo removal is through cancellation, where an adaptive filter is used to produce an estimated replica of the echo which is then subtracted from the echo-infested signal. Echo cancellation in practice requires extensive control of the filter adaptation process in order to obtain as rapid convergence as possible while also achieving robustness towards disturbances. Moreover, despite the rapid advancement in the computational capabilities of modern digital signal processors there is a constant demand for low-complexity solutions that can be implemented using low power and low cost hardware. This thesis presents low-complexity solutions for echo cancellation related to both the actual filter adaptation process itself as well as for controlling the adaptation process in order to obtain a robust system. Extensive simulations and evaluations using real world recorded signals are used to demonstrate the performance of the proposed solutions.
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| 28. |
- Schüldt, Christian, 1978-, et al.
(author)
-
Noise robust integration for blind and non-blind reverberation time estimation
- 2015
-
In: Acoustics, Speech and Signal Processing (ICASSP), 2015 IEEE International Conference on. - : IEEE Signal Processing Society. ; , s. 56-60
-
Conference paper (peer-reviewed)abstract
- The estimation of the decay rate of a signal section is an integral component of both blind and non-blind reverberation time estimation methods. Several decay rate estimators have previously been proposed, based on, e.g., linear regression and maximum-likelihood estimation. Unfortunately, most approaches are sensitive to background noise, and/or are fairly demanding in terms of computational complexity. This paper presents a low complexity decay rate estimator, robust to stationary noise, for reverberation time estimation. Simulations using artificial signals, and experiments with speech in ventilation noise, demonstrate the performance and noise robustness of the proposed method.
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| 29. |
- Schüldt, Christian, 1978-, et al.
(author)
-
On Implications of the ISO 3382 Backward Integration Method for Automated Decay Rate Estimation
- 2015
-
In: Journal of The Audio Engineering Society. - : Audio Engineering Society. - 1549-4950. ; 63:3, s. 161-173
-
Journal article (peer-reviewed)abstract
- The Schröder backward integration method for estimating the reverberation time of an enclosure, as suggested in the ISO 3382 standard, is analyzed from an estimation theoretic perspective, in a general context that is applicable to both blind and non-blind estimation. Expressions for the estimation bias and variance of the reverberation decay rate are derived and verified using Monte-Carlo simulations. Comparison is made with a straight-forward linear regression method (not using backward integration). It is shown that, even though significantly reducing the estimation variance, the use of backward integration can in many cases mitigate the estimation accuracy due to large bias. This clearly indicates that prudence is called for when using backward integration for automated decay rate estimation problems.
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| 30. |
- Schüldt, Christian, et al.
(author)
-
Recognizing human actions : A local SVM approach
- 2004
-
In: PROCEEDINGS OF THE 17TH INTERNATIONAL CONFERENCE ON PATTERN RECOGNITION, VOL 3. - 0769521282 ; , s. 32-36
-
Conference paper (peer-reviewed)abstract
- Local space-time features capture local events in video and can be adapted to the size, the frequency and the velocity of moving patterns. In this paper we demonstrate how such features can be used for recognizing complex motion patterns. We construct video representations in terms of local space-time features and integrate such representations with SVM classification schemes for recognition. For the purpose of evaluation we introduce a new video database containing 2391 sequences of six human actions performed by 25 people in four different scenarios. The presented results of action recognition justify the proposed method and demonstrate its advantage compared to other relative approaches for action recognition.
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| 31. |
- Schüldt, Christian, et al.
(author)
-
Robust low-complexity transfer logic for two-path echo cancellation
- 2012
-
Conference paper (peer-reviewed)abstract
- A well used approach for echo cancellation is the two-path method, where two adaptive filters in parallel are utilized. Typically, one filter is continuously updated, and when this filter is considered better adjusted to the echo-path than the other filter, the coefficients of the better adjusted filter is transferred to the other filter. When this transfer should occur is controlled by the transfer logic. This paper proposes transfer logic that is both more robust and more simple to tune, owing to fewer parameters, than the conventional approach. Extensive simulations show the advantages of the proposed method.
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| 32. |
- Zug, S., et al.
(author)
-
Technical Evaluation of the Carolo-Cup 2014 - A Competition for Self-Driving Miniature Cars
- 2014
-
In: 12th IEEE International Symposium on RObotic and Sensors Environments (ROSE), Politehnica Univ Timisoara, Timisoara, ROMANIA, OCT 16-18, 2014. - : IEEE. - 9781479949274 ; , s. 100-105
-
Conference paper (peer-reviewed)abstract
- The Carolo-Cup competition conducted for the eighth time this year, is an international student competition focusing on autonomous driving scenarios implemented on 1:10 scale car models. Three practical sub-competitions have to be realized in this context and represent a complex, interdisciplinary challenge. Hence, students have to cope with all core topics like mechanical development, electronic design, and programming as addressed usually by robotic applications. In this paper we introduce the competition challenges in detail and evaluate the results of all 13 participating teams from the 2014 competition. For this purpose, we analyze technical as well as non-technical configurations of each student group and derive best practices, lessons learned, and criteria as a precondition for a successful participation. Due to the comprehensive orientation of the Carolo-Cup, this knowledge can be applied on comparable projects and related competitions as well.
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