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1993 IEEE International Conference on Acoustics, Speech, and Signal Processing最新文献

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Digital algorithms for suppression of adjacent channel interference in FM receivers 调频接收机中相邻信道干扰抑制的数字算法
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319421
A. Schlereth
{"title":"Digital algorithms for suppression of adjacent channel interference in FM receivers","authors":"A. Schlereth","doi":"10.1109/ICASSP.1993.319421","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319421","url":null,"abstract":"Two novel digital algorithms are presented, which allow the demodulation of an FM signal in broadcasting even if there is one adjacent channel interference with overlapping spectra which is stronger than the desired signal. The first algorithm is the cross-coupled-baseband digital phase-locked loop (CC-BB-DPLL), which avoids the problem of changing the synchronization. Much better compensation results are possible by using the second algorithm, called the forward compensation structure (FCS). The FCS is a simplified CC-BB-DPLL with internal intermediate-frequency (IF) filtering. The complete signal estimation including an adaptive compensation algorithm is presented. All the filter and estimation structures are optimized using FM model processes generated by the Monte Carlo method.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126165530","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}
引用次数: 3
The symmetric convolution approach to the nonexpansive implementations of FIR filter banks for images 图像FIR滤波器组非扩展实现的对称卷积方法
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319748
S. Martucci, R. Mersereau
{"title":"The symmetric convolution approach to the nonexpansive implementations of FIR filter banks for images","authors":"S. Martucci, R. Mersereau","doi":"10.1109/ICASSP.1993.319748","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319748","url":null,"abstract":"The authors describe symmetric convolution and its use for the nonexpansive implementation of multirate filter banks for images. Symmetric convolution is a formalized approach to convolving symmetric FIR (finite impulse response) filters with symmetrically extended data. It is efficient because the discrete sine and cosine transforms can be used to perform the convolution as a transform-domain multiplication. The authors explain how to use symmetric convolution to implement a multiband filter bank for finite-length data that restricts the number of samples in the subbands but still gives perfect reconstruction.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123667117","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}
引用次数: 23
New inverse filter criteria for identification and deconvolution of nonminimum-phase systems by single cumulant slice 基于单累积量片的非最小相位系统识别与反褶积的新反滤波准则
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319627
Wu-Ton Chen, Chong-Yung Chi
{"title":"New inverse filter criteria for identification and deconvolution of nonminimum-phase systems by single cumulant slice","authors":"Wu-Ton Chen, Chong-Yung Chi","doi":"10.1109/ICASSP.1993.319627","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319627","url":null,"abstract":"The authors propose a family of new cumulant based inverse filter criteria which only require a single slice of cumulants of the inverse filter output for the identification and deconvolution of linear time-invariant (LTI) nonminimum-phase systems with only non-Gaussian output measurements contaminated by Gaussian noise. Some simulation results and an application to speech deconvolution are provided to demonstrate that inverse filtering algorithms based on the proposed new criteria work well.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123671730","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}
引用次数: 4
Signal constellations for non-Gaussian communication problems 非高斯通信问题的信号星座
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319428
A. Dabak, Don H. Johnson
{"title":"Signal constellations for non-Gaussian communication problems","authors":"A. Dabak, Don H. Johnson","doi":"10.1109/ICASSP.1993.319428","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319428","url":null,"abstract":"On the basis of a geometric theory of detection, the authors extend the notion of a signal constellation, a concept deeply rooted in Gaussian problems, to the non-Gaussian case. Significant differences between optimal designs for Gaussian and non-Gaussian situations are shown. In particular, square-wave signals are much more important in heavy-tailed, non-Gaussian noise situations than in Gaussian ones. Furthermore, design guidelines for non-Gaussian problems can vary with the number of signal set members and can depend on SNR. The extent to which suboptimal designs affect performance (using Gaussian-based designs in non-Gaussian situations, for example) can be predicted from calculations of the Kullback information, but only in the sense of determining how the logarithmic error probability rates differ.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"29 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125491973","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}
引用次数: 1
Cross-lingual experiments with phone recognition 手机识别的跨语言实验
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319353
L. Lamel, J. Gauvain
{"title":"Cross-lingual experiments with phone recognition","authors":"L. Lamel, J. Gauvain","doi":"10.1109/ICASSP.1993.319353","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319353","url":null,"abstract":"Research on speaker-independent continuous phone recognition for both French and English is presented. The phone accuracy is assessed on the BREF corpus for French, and on the Wall Street Journal (WSJ) and TIMIT corpora for English. Cross-language differences concerning language properties are presented. It is found that French is easier to recognize at the phone level (the phone error for BREF is 23.6% vs. 30.1% for WSJ), but harder to recognize at the lexical level due to the larger number of homophones. Experiments with signal analysis indicate that a 4 kHz signal bandwidth is sufficient for French, whereas 8 kHz is needed for English. Phone recognition is a powerful technique for language, sex, and speaker identification. With 2 s of speech, the language can be identified with better than 99% accuracy. Sex-identification for BREF and WSJ is error-free. Speaker identification accuracies of 98.2% on TIMIT (462 speakers) and 99.1% on BREF (57 speakers) were obtained with one utterance per speaker. 100% accuracies were obtained with two utterances per speaker.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126702592","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}
引用次数: 59
Wavelet regularity of iterated filter banks with rational sampling changes 具有合理采样变化的迭代滤波器组的小波正则性
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319473
T. Blu, O. Rioul
{"title":"Wavelet regularity of iterated filter banks with rational sampling changes","authors":"T. Blu, O. Rioul","doi":"10.1109/ICASSP.1993.319473","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319473","url":null,"abstract":"The regularity property was first introduced by wavelet theory for octave-band dyadic filter banks. In the present work, the authors provide a detailed theoretical analysis of the regularity property in the more flexible case of filter banks with rational sampling changes. Such filter banks provide a finer analysis of fractions of an octave, and regularity is as important as in the dyadic case. Sharp regularity estimates for any filter bank are given. The major difficulty of the rational case, as compared with the dyadic case, is that one obtains wavelets that are not shifted versions of each other at a given scale. It is shown, however, that, under regularity conditions, shift invariance can almost be obtained. This is a desirable property for, e.g. coding applications and for efficient filter bank implementation of a continuous wavelet transform.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115015628","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}
引用次数: 4
Speaker adaptation using improved speaker Markov models 基于改进的说话人马尔可夫模型的说话人自适应
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319370
G. Rigoll
{"title":"Speaker adaptation using improved speaker Markov models","authors":"G. Rigoll","doi":"10.1109/ICASSP.1993.319370","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319370","url":null,"abstract":"An attempt has been made to develop improved and more sophisticated SMMs (speaker Markov models) capable of modeling the acoustic differences between two speakers in a more accurate way, thus leading to improved recognition rates for the adapted speech recognition system. The original SSM approach has been improved by the introduction of the following three features: the use of fenonic speaker Markov models, the introduction of phoneme-dependent SMM parameters, and the use of special weighting between the short original training data of the new speaker and the adapted training data of the reference speaker. It was found that the phoneme recognition performance of these improved SMMs can be more than twice as high as the performance of the original SMM approach, which has already led to satisfying adaptation results for a large-vocabulary speech recognition task.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"17 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115193343","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}
引用次数: 3
Speaker adaptation based on MAP estimation of HMM parameters 基于HMM参数MAP估计的说话人自适应
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319368
Chin-Hu Lee, J. Gauvain
{"title":"Speaker adaptation based on MAP estimation of HMM parameters","authors":"Chin-Hu Lee, J. Gauvain","doi":"10.1109/ICASSP.1993.319368","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319368","url":null,"abstract":"A number of issues related to the application of Bayesian learning techniques to speaker adaptation are investigated. It is shown that the seed models required to construct prior densities to obtain the MAP (maximum a posteriori) estimate can be a speaker-independent (SI) model, a set of female and male models, or even a task-independent acoustic model. Speaker-adaptive training algorithms are shown to be effective in improving the performance of both speaker-dependent and speaker-independent speech recognition systems. The segmental MAP estimation formulation is used to perform adaptive acoustic modeling for speaker adaptation applications. Tested on an RM (resource management) task, it was found that supervised speaker adaptation based on two gender-dependent models gave a better result than that obtained with a single SI seed. Compared with speaker-dependent training, speaker adaptation achieved an equal or better performance with the same amount of training/adaptation data.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115500803","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}
引用次数: 183
Directionalizing adaptive multi-microphone arrays for hearing aids using cardioid microphones 使用心型麦克风的助听器定向自适应多麦克风阵列
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319084
E. D. McKinney, V. DeBrunner
{"title":"Directionalizing adaptive multi-microphone arrays for hearing aids using cardioid microphones","authors":"E. D. McKinney, V. DeBrunner","doi":"10.1109/ICASSP.1993.319084","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319084","url":null,"abstract":"The authors examine the design and performance of a restricted geometry, adaptive four-element acoustic array using cardioid (directional) microphones for hearing enhancement. The array is portable and lightweight. It allows the hearing aid wearer to focus better in a forward-looking direction, reducing interfering noises from unwanted directions. The array also allows the system to respond to changing environments, nulling strong interference noises which do not impinge on the front of the array. The directionality and the adaptable interference rejection of the multi-element array are superior to that possible using single or two-element devices. Adaptive arrays using directional microphones are shown to enhance the array output SNR above that of an adaptive array consisting of omni-directional sensors. Examples and simulation results are presented.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"41 4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115556148","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}
引用次数: 3
Leak monitoring system for gas pipelines 燃气管道泄漏监测系统
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319424
Igal Brodetsky, M. Savic
{"title":"Leak monitoring system for gas pipelines","authors":"Igal Brodetsky, M. Savic","doi":"10.1109/ICASSP.1993.319424","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319424","url":null,"abstract":"An approach and a solution to the continuous leak monitoring problem in underground gas pipelines are presented. This approach places permanent monitoring units along the pipeline. These units detect acoustic signals in the pipeline and discriminate leak sounds from other man-made or natural nonleak sounds that can occur. The system uses the kNN classifier as the detector with LPC (linear predictive coding) cepstrums as signal features. To increase system performance, pipeline effects on acoustic signals were taken into account during the classifier training phase. Each unit can detect 1/4-in-diameter leaks from a distance of 300 m, yielding 600 m as the maximum distance between units.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"35 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1993-04-27","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122331561","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}
引用次数: 27
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