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

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A spherical subspace based adaptive filter 一种基于球面子空间的自适应滤波器
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319545
E. Dowling, R. DeGroat
{"title":"A spherical subspace based adaptive filter","authors":"E. Dowling, R. DeGroat","doi":"10.1109/ICASSP.1993.319545","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319545","url":null,"abstract":"The authors use the adaptation mechanism of the spherical subspace tracker together with the weighting scheme of total least squares (TLS) to construct an adaptive filter that tracks solutions to time-varying ordinary least squares. TLS, data least squares, and reduced rank problems. To study convergence properties, they relate this filter to Thompson's constrained stochastic gradient eigenfilter. They present a convergence rate acceleration scheme that keeps the filter from being slowed down by saddle points in the performance surface. Simulation results verify the theoretical development. The filter behaves well in the full rank case and is more sensitive and slow to converge in certain reduced rank problems.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"62 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":"117049263","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: 5
Direct modulation on LPC coefficients with application to speech enhancement and improving the performance of speech recognition in noise LPC系数的直接调制及其在语音增强中的应用，提高了噪声环境下语音识别的性能
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319242
Cuntai Guan, Yongbin Chen, Boxiu Wu
{"title":"Direct modulation on LPC coefficients with application to speech enhancement and improving the performance of speech recognition in noise","authors":"Cuntai Guan, Yongbin Chen, Boxiu Wu","doi":"10.1109/ICASSP.1993.319242","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319242","url":null,"abstract":"A novel method of noise reduction of speech based on direct modulation of LPC (linear predictive coding) coefficients is proposed. This method introduces higher-order derivatives of LPC coefficients with respect to the noise-to-signal energy ratio (NSR). With these derivatives, the noisy LPC coefficients are refined flexibly and efficiently to reduce noise contaminations. This method only needs the environmental NSR, and does not require knowledge of the probability distribution of the noise. This enhancement method is incorporated in an HMM (hidden Markov model)-based speech recognition system using LPC-derived cepstral features. A pronounced recognition error rate reduction is obtained after the speech enhancement.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"49 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":"116430942","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: 11
Modified MUSIC in the presence of random phase errors 在随机相位误差存在时修改MUSIC
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319121
G. Zunich
{"title":"Modified MUSIC in the presence of random phase errors","authors":"G. Zunich","doi":"10.1109/ICASSP.1993.319121","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319121","url":null,"abstract":"The high resolution direction-finding technique Multiple Signal Classification (MUSIC) provides excellent direction-of-arrival (DOA) performance under ideal conditions. Its performance, however, is significantly degraded under conditions of element phase errors. Robust constraints that have previously been developed to protect signal-to-noise performance of a desired signal in phase perturbed adaptive beamformers can be used in a modified form of MUSIC to protect performance against phase errors. Such phase perturbations can be caused by channel phase errors, array element placement errors, and frequency errors. Simulation results are presented with random element placement errors.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"142 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":"123462280","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
Signal estimation using the discrete polynominal transform 使用离散多标称变换的信号估计
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319685
S. Peleg, B. Friedlander
{"title":"Signal estimation using the discrete polynominal transform","authors":"S. Peleg, B. Friedlander","doi":"10.1109/ICASSP.1993.319685","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319685","url":null,"abstract":"The authors develop an iterative procedure for estimating the parameters of a signal consisting of multiple polynomial-phase components with time-varying amplitudes. The proposed algorithm is based on the discrete polynomial transform (DPT), and shares its computational simplicity. Preliminary experimentation with the algorithm has shown that it is able to provide accurate parameter estimates. A numerical example is included.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"12 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":"123616082","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: 2
The application of subband coding to improve quality and robustness of the sinusoidal transform coder 应用子带编码提高正弦变换编码器的质量和鲁棒性
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319334
R. McAulay, T. Quatieri
{"title":"The application of subband coding to improve quality and robustness of the sinusoidal transform coder","authors":"R. McAulay, T. Quatieri","doi":"10.1109/ICASSP.1993.319334","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319334","url":null,"abstract":"In an attempt to improve the performance of the sinusoidal transform coder (STC) in the presence of channel errors, a structure evolved that was similar to a subband coder. Using some new ideas for subband coding, such as vector quantization of the subband energies and adaptive bit allocation based on the decoded energies, a new quantization algorithm for STC evolved that was not only more robust in channel errors, but led to improved quality, particularly at the 2400 bit/s data rate. It may be possible further to improve the speech quality at this rate by using vector quantization to code the shape of the subband channel gains.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"87 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":"123672739","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: 18
A new speaker adaptation technique using very short calibration speech 一种新的使用极短校准语音的说话人自适应技术
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319369
Yunxin Zhao
{"title":"A new speaker adaptation technique using very short calibration speech","authors":"Yunxin Zhao","doi":"10.1109/ICASSP.1993.319369","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319369","url":null,"abstract":"A speaker adaptation technique based on the separation of speech spectra variation sources is developed for improving speaker-independent continuous speech recognition. The variation sources include speaker acoustic characteristics, phonologic characteristics, and contextual dependency of allophones. Statistical methods are formulated to normalize speech spectra based on speaker acoustic characteristics and then adapt mixture Gaussian density phone models based on speaker phonologic characteristics. Adaptation experiments using short calibration speech (5 s/speaker) have shown substantial performance improvement over the baseline recognition system. On a TIMIT test set, where the task vocabulary size is 853 and the test set perplexity is 104, the recognition word accuracy has been improved from 86.9% to 90.6% (28.2% error reduction). On a separate test set which contains an additional variation source of recording channel mismatch and with the test set perplexity of 101, the recognition word accuracy has been improved from 65.4% to 85.5% (58.1% error reduction).<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"118 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":"122061610","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: 20
A signal subspace approach for speech enhancement 语音增强的一种信号子空间方法
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319311
Y. Ephraim, H. V. Trees
{"title":"A signal subspace approach for speech enhancement","authors":"Y. Ephraim, H. V. Trees","doi":"10.1109/ICASSP.1993.319311","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319311","url":null,"abstract":"A perceptually based linear signal estimator for enhancing speech signals degraded by uncorrelated additive noise is developed. The estimator is designed by minimizing the signal distortion while maintaining the residual noise level below some given threshold. The estimator is shown to be a Wiener filter with adjustable input noise level. This level is determined by the threshold of the permissible residual noise. The estimator is implemented using the signal subspace approach. The vector space of the noisy signal is decomposed into a signal subspace and complementary orthogonal noise subspace. Estimation is performed from vectors in the signal subspace only, since the orthogonal subspace does not contain signal information. The proposed estimator is shown to be a refinement of a version of the spectral subtraction signal estimator. The latter estimator is shown to be asymptotically optimal for stationary signal and noise in the linear minimum mean square error sense.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"164 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":"120882481","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: 1064
Deconvolution of sparse spike trains accounting for wavelet phase shifts and colored noise 考虑小波相移和彩色噪声的稀疏尖峰串的反卷积
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319532
F. Champagnat, J. Idier, G. Demoment
{"title":"Deconvolution of sparse spike trains accounting for wavelet phase shifts and colored noise","authors":"F. Champagnat, J. Idier, G. Demoment","doi":"10.1109/ICASSP.1993.319532","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319532","url":null,"abstract":"The problem of the restoration of spiky sequences when the usual convolution model is corrupted by nonstationary wavelet phase-shifts is addressed. To this end, an extended convolution model driven by a Bernoulli-Gaussian (BG)-like process is introduced. This setting lends itself to easy extension of algorithms designed for BG deconvolution. A comparison of practical results obtained with this new method and BG deconvolution is provided. Numerical experiments indicate an increased robustness compared with standard BG methods.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"34 8 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":"125719955","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: 18
Halftone to continuous-tone conversion of error-diffusion coded images 误差扩散编码图像的半色调到连续色调转换
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319809
S. Hein, A. Zakhor
{"title":"Halftone to continuous-tone conversion of error-diffusion coded images","authors":"S. Hein, A. Zakhor","doi":"10.1109/ICASSP.1993.319809","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319809","url":null,"abstract":"The problem of reconstructuring a continuous-tone (cotone) image from its error-diffused halftoned version is presented. The authors present a POCS (projection onto convex sets)-based iterative nonlinear algorithm for this problem, and show simulation results that compare the performance of the algorithm with that of conventional linear lowpass filtering. It is found that the proposed technique results in subjectively superior reconstruction. As there is a natural relationship between error diffusion and sigma-delta modulation, the proposed reconstruction algorithm can also be applied to the decoding problem for sigma-delta modulators.<<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":"125799598","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: 119
Wideband speech coding in 7.2 kbit/s 宽带语音编码7.2 kbit/s
1993 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1993-04-27 DOI: 10.1109/ICASSP.1993.319380
C. McElroy, B. Murray, A. Fagan
{"title":"Wideband speech coding in 7.2 kbit/s","authors":"C. McElroy, B. Murray, A. Fagan","doi":"10.1109/ICASSP.1993.319380","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319380","url":null,"abstract":"A novel method for coding wideband speech at medium bit rates is proposed. A subband approach is adopted, using sharp cut-off filters to split the speech into lower and upper bands. This allows the upper-band to be very coarsely quantized without introducing significant distortion at low frequencies in the reconstructed speech, which would normally occur using short QMF (quadrature mirror filter). A standard CELP (code excited linear prediction) coder is used for the low-frequency band, and the upper band is quantized using a second-order predictor and gain shape vector quantization at 0.05 bit sample, yielding an overall bit rate of 7.2 kbit/s. The overall effect is to produce a wideband coder that has almost the same computational requirements as a narrowband CELP coder and only a slight increase in the bit rate.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"14 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":"125951054","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: 9
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