Liv Merete Reinar, Louise Forsetlund, Arild Bjørndal, Diana Nj Lockwood
{"title":"WITHDRAWN: Interventions for skin changes caused by nerve damage in leprosy.","authors":"Liv Merete Reinar, Louise Forsetlund, Arild Bjørndal, Diana Nj Lockwood","doi":"10.1002/14651858.CD004833.pub4","DOIUrl":"10.1002/14651858.CD004833.pub4","url":null,"abstract":"<p><strong>Background: </strong>More than three million persons are disabled by leprosy worldwide. The main complication of sensory nerve damage is neuropathic ulceration, particularly of the feet. In this review we explored interventions that can prevent and treat secondary damage to skin and limbs.</p><p><strong>Objectives: </strong>To assess the effects of self-care, dressings and footwear in preventing and healing secondary damage to the skin in persons affected by leprosy.</p><p><strong>Search methods: </strong>We searched the Cochrane Skin Group Specialised Register (April 2008), the Cochrane Central Register of Controlled Trials (The Cochrane Library Issue 1, 2008), MEDLINE (from 2003 to April 2008), EMBASE (from 2005 to April 2008), CINAHL (1982-2006) and LILACS (1982- April 2008 ) as well as online registers of ongoing trials (April 2008).</p><p><strong>Selection criteria: </strong>Randomised controlled trials involving anyone with leprosy and damage to peripheral nerves treated with any measures designed to prevent damage with the aim of healing existing ulcers and preventing development of new ulcers.</p><p><strong>Data collection and analysis: </strong>Two authors assessed trial quality and extracted data.</p><p><strong>Main results: </strong>Eight trials with a total of 557 participants were included. The quality of the trials was generally poor. The interventions and outcome measures were diverse. Although three studies that compared zinc tape to more traditional dressings found some benefit, none of these showed a statistically significant effect. One trial indicated that topical ketanserin had a better effect on wound healing than clioquinol cream or zinc paste, RR was 6.00 (95% CI 1.45 to 24.75). We did not combine the results of the two studies that compared topical phenytoin to saline dressing, but both studies found statistically significant effects in favour of phenytoin for healing of ulcer (SMD -2.34; 95% CI -3.30 to -1.39; and SMD -0.79; 95% CI -1.20 to 0.39). Canvas shoes were not much better than PVC-boots, and double rocker shoes did not promote healing much more than below-knee plasters.</p><p><strong>Authors' conclusions: </strong>One study suggested that topical ketanserin is more effective than clioquinol cream or zinc paste. Topical phenytoin (two studies) may be more effective than saline dressing regarding ulcer healing. For the other dressings the results were equivocal. Canvas shoes were a little better than PVC-boots, but not significantly, and the effect of double rocker shoes compared to below-knee plasters was no different in promoting the healing of ulcers. No side effects were documented.There is a lack of high quality research in the field of ulcer prevention and treatment in leprosy. New trials should follow the current standards for design and reporting of randomised controlled trials.</p>","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"20 1","pages":"CD004833"},"PeriodicalIF":8.4,"publicationDate":"2019-08-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1002/14651858.CD004833.pub4","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"84911376","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"OA","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
J. Valin, Timothy B. Terriberry, Christopher Montgomery, Gregory Maxwell
{"title":"A High-Quality Speech and Audio Codec With Less Than 10-ms Delay","authors":"J. Valin, Timothy B. Terriberry, Christopher Montgomery, Gregory Maxwell","doi":"10.1109/TASL.2009.2023186","DOIUrl":"https://doi.org/10.1109/TASL.2009.2023186","url":null,"abstract":"With increasing quality requirements for multimedia communications, audio codecs must maintain both high quality and low delay. Typically, audio codecs offer either low delay or high quality, but rarely both. We propose a codec that simultaneously addresses both these requirements, with a delay of only 8.7 ms at 44.1 kHz. It uses gain-shape algebraic vector quantization in the frequency domain with time-domain pitch prediction. We demonstrate that the proposed codec operating at 48 kb/s and 64 kb/s out-performs both G.722.1C and MP3 and has quality comparable to AAC-LD, despite having less than one fourth of the algorithmic delay of these codecs.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"18 1","pages":"58-67"},"PeriodicalIF":0.0,"publicationDate":"2016-02-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2009.2023186","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62852437","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Efficient Approximation of Head-Related Transfer Functions in Subbands for Accurate Sound Localization.","authors":"Damián Marelli, Robert Baumgartner, Piotr Majdak","doi":"","DOIUrl":"","url":null,"abstract":"<p><p>Head-related transfer functions (HRTFs) describe the acoustic filtering of incoming sounds by the human morphology and are essential for listeners to localize sound sources in virtual auditory displays. Since rendering complex virtual scenes is computationally demanding, we propose four algorithms for efficiently representing HRTFs in subbands, i.e., as an analysis filterbank (FB) followed by a transfer matrix and a synthesis FB. All four algorithms use sparse approximation procedures to minimize the computational complexity while maintaining perceptually relevant HRTF properties. The first two algorithms separately optimize the complexity of the transfer matrix associated to each HRTF for fixed FBs. The other two algorithms jointly optimize the FBs and transfer matrices for complete HRTF sets by two variants. The first variant aims at minimizing the complexity of the transfer matrices, while the second one does it for the FBs. Numerical experiments investigate the latency-complexity trade-off and show that the proposed methods offer significant computational savings when compared with other available approaches. Psychoacoustic localization experiments were modeled and conducted to find a reasonable approximation tolerance so that no significant localization performance degradation was introduced by the subband representation.</p>","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"23 7","pages":"1130-1143"},"PeriodicalIF":0.0,"publicationDate":"2015-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://www.ncbi.nlm.nih.gov/pmc/articles/PMC4678625/pdf/","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"140208278","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"OA","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
T. Nakatani, S. Araki, Takuya Yoshioka, Marc Delcroix, M. Fujimoto
{"title":"Dominance Based Integration of Spatial and Spectral Features for Speech Enhancement","authors":"T. Nakatani, S. Araki, Takuya Yoshioka, Marc Delcroix, M. Fujimoto","doi":"10.1109/TASL.2013.2277937","DOIUrl":"https://doi.org/10.1109/TASL.2013.2277937","url":null,"abstract":"This paper proposes a versatile technique for integrating two conventional speech enhancement approaches, a spatial clustering approach (SCA) and a factorial model approach (FMA), which are based on two different features of signals, namely spatial and spectral features, respectively. When used separately the conventional approaches simply identify time frequency (TF) bins that are dominated by interference for speech enhancement. Integration of the two approaches makes identification more reliable, and allows us to estimate speech spectra more accurately even in highly nonstationary interference environments. This paper also proposes extensions of the FMA for further elaboration of the proposed technique, including one that uses spectral models based on mel-frequency cepstral coefficients and another to cope with mismatches, such as channel mismatches, between captured signals and the spectral models. Experiments using simulated and real recordings show that the proposed technique can effectively improve audible speech quality and the automatic speech recognition score.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2516-2531"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2277937","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892355","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Linearly-Constrained Minimum-Variance Method for Spherical Microphone Arrays Based on Plane-Wave Decomposition of the Sound Field","authors":"Yotam Peled, B. Rafaely","doi":"10.1109/TASL.2013.2277939","DOIUrl":"https://doi.org/10.1109/TASL.2013.2277939","url":null,"abstract":"Speech signals recorded in real environments may be corrupted by ambient noise and reverberation. Therefore, noise reduction and dereverberation algorithms for speech enhancement are typically employed in speech communication systems. Although microphone arrays are useful in reducing the effect of noise and reverberation, existing methods have limited success in significantly removing both reverberation and noise in real environments. This paper presents a method for noise reduction and dereverberation that overcomes some of the limitations of previous methods. The method uses a spherical microphone array to achieve plane-wave decomposition (PWD) of the sound field, based on direction-of-arrival (DOA) estimation of the desired signal and its reflections. A multi-channel linearly-constrained minimum-variance (LCMV) filter is introduced to achieve further noise reduction. The PWD beamformer achieves dereverberation while the LCMV filter reduces the uncorrelated noise with a controllable dereverberation constraint. In contrast to other methods, the proposed method employs DOA estimation, rather than room impulse response identification, to achieve dereverberation, and relative transfer function (RTF) estimation between the source reflections to achieve noise reduction while avoiding signal cancellation. The paper includes a simulation investigation and an experimental study, comparing the proposed method to currently available methods.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2532-2540"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2277939","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892413","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
J. Jensen, J. Benesty, M. G. Christensen, Jingdong Chen
{"title":"A Class of Optimal Rectangular Filtering Matrices for Single-Channel Signal Enhancement in the Time Domain","authors":"J. Jensen, J. Benesty, M. G. Christensen, Jingdong Chen","doi":"10.1109/TASL.2013.2280215","DOIUrl":"https://doi.org/10.1109/TASL.2013.2280215","url":null,"abstract":"In this paper, we introduce a new class of optimal rectangular filtering matrices for single-channel speech enhancement. The new class of filters exploits the fact that the dimension of the signal subspace is lower than that of the full space. By doing this, extra degrees of freedom in the filters, that are otherwise reserved for preserving the signal subspace, can be used for achieving an improved output signal-to-noise ratio (SNR). Moreover, the filters allow for explicit control of the tradeoff between noise reduction and speech distortion via the chosen rank of the signal subspace. An interesting aspect is that the framework in which the filters are derived unifies the ideas of optimal filtering and subspace methods. A number of different optimal filter designs are derived in this framework, and the properties and performance of these are studied using both synthetic, periodic signals and real signals. The results show a number of interesting things. Firstly, they show how speech distortion can be traded for noise reduction and vice versa in a seamless manner. Moreover, the introduced filter designs are capable of achieving both the upper and lower bounds for the output SNR via the choice of a single parameter.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2595-2606"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2280215","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892558","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Investigations on an EM-Style Optimization Algorithm for Discriminative Training of HMMs","authors":"G. Heigold, H. Ney, R. Schlüter","doi":"10.1109/TASL.2013.2280234","DOIUrl":"https://doi.org/10.1109/TASL.2013.2280234","url":null,"abstract":"Today's speech recognition systems are based on hidden Markov models (HMMs) with Gaussian mixture models whose parameters are estimated using a discriminative training criterion such as Maximum Mutual Information (MMI) or Minimum Phone Error (MPE). Currently, the optimization is almost always done with (empirical variants of) Extended Baum-Welch (EBW). This type of optimization requires sophisticated update schemes for the step sizes and a considerable amount of parameter tuning, and only little is known about its convergence behavior. In this paper, we derive an EM-style algorithm for discriminative training of HMMs. Like Expectation-Maximization (EM) for the generative training of HMMs, the proposed algorithm improves the training criterion on each iteration, converges to a local optimum, and is completely parameter-free. We investigate the feasibility of the proposed EM-style algorithm for discriminative training of two tasks, namely grapheme-to-phoneme conversion and spoken digit string recognition.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2616-2626"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2280234","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892624","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Soundfield Imaging in the Ray Space","authors":"D. Markovic, F. Antonacci, A. Sarti, S. Tubaro","doi":"10.1109/TASL.2013.2274697","DOIUrl":"https://doi.org/10.1109/TASL.2013.2274697","url":null,"abstract":"In this work we propose a general approach to acoustic scene analysis based on a novel data structure (ray-space image) that encodes the directional plenacoustic function over a line segment (Observation Window, OW). We define and describe a system for acquiring a ray-space image using a microphone array and refer to it as ray-space (or “soundfield”) camera. The method consists of acquiring the pseudo-spectra corresponding to a grid of sampling points over the OW, and remapping them onto the ray space, which parameterizes acoustic paths crossing the OW. The resulting ray-space image displays the information gathered by the sensors in such a way that the elements of the acoustic scene (sources and reflectors) will be easy to discern, recognize and extract. The key advantage of this method is that ray-space images, irrespective of the application, are generated by a common (and highly parallelizable) processing layer, and can be processed using methods coming from the extensive literature of pattern analysis. After defining the ideal ray-space image in terms of the directional plenacoustic function, we show how to acquire it using a microphone array. We also discuss resolution and aliasing issues and show two simple examples of applications of ray-space imaging.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"12 1","pages":"2493-2505"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2274697","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892128","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
A. Prathosh, T. Ananthapadmanabha, A. Ramakrishnan
{"title":"Epoch Extraction Based on Integrated Linear Prediction Residual Using Plosion Index","authors":"A. Prathosh, T. Ananthapadmanabha, A. Ramakrishnan","doi":"10.1109/TASL.2013.2273717","DOIUrl":"https://doi.org/10.1109/TASL.2013.2273717","url":null,"abstract":"Epoch is defined as the instant of significant excitation within a pitch period of voiced speech. Epoch extraction continues to attract the interest of researchers because of its significance in speech analysis. Existing high performance epoch extraction algorithms require either dynamic programming techniques or a priori information of the average pitch period. An algorithm without such requirements is proposed based on integrated linear prediction residual (ILPR) which resembles the voice source signal. Half wave rectified and negated ILPR (or Hilbert transform of ILPR) is used as the pre-processed signal. A new non-linear temporal measure named the plosion index (PI) has been proposed for detecting ‘transients’ in speech signal. An extension of PI, called the dynamic plosion index (DPI) is applied on pre-processed signal to estimate the epochs. The proposed DPI algorithm is validated using six large databases which provide simultaneous EGG recordings. Creaky and singing voice samples are also analyzed. The algorithm has been tested for its robustness in the presence of additive white and babble noise and on simulated telephone quality speech. The performance of the DPI algorithm is found to be comparable or better than five state-of-the-art techniques for the experiments considered.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2471-2480"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2273717","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62891683","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Body Conducted Speech Enhancement by Equalization and Signal Fusion","authors":"Tomas Dekens, W. Verhelst","doi":"10.1109/TASL.2013.2274696","DOIUrl":"https://doi.org/10.1109/TASL.2013.2274696","url":null,"abstract":"This paper studies body-conducted speech for noise robust speech processing purposes. As body-conducted speech is typically limited in bandwidth, signal processing is required to obtain a signal that is both high in quality and low in noise. We propose an algorithm that first equalizes the body-conducted speech using filters obtained from a pre-defined filter set and subsequently fuses this equalized signal with a noisy conventional microphone signal using an optimal clean speech amplitude and phase estimator. We evaluated the proposed equalization and fusion technique using a combination of a conventional close-talk and a throat microphone. Subjective listening tests show that the proposed method successfully fuses the speech quality of the conventional signal and the noise robustness of the throat microphone signal. The listening tests also indicate that the inclusion of the body-conducted signal can improve single-channel speech enhancement methods, while a calculated set of objective signal quality measures confirm these observations.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":"21 1","pages":"2481-2492"},"PeriodicalIF":0.0,"publicationDate":"2013-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2274696","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62892043","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}