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Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing最新文献

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Subband coding of color images with limited palette size 色板大小有限的彩色图像的子带编码
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.389415
P. Waldemar, T. Ramstad
{"title":"Subband coding of color images with limited palette size","authors":"P. Waldemar, T. Ramstad","doi":"10.1109/ICASSP.1994.389415","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389415","url":null,"abstract":"An approach for compression of color images using a sorted color palette and a subband source coder is presented. When the decoded image is to be displayed on an eight bit monitor this method significantly reduces the decoder computational complexity. To avoid requantization before displaying the decoded image, the encoding and decoding are performed on a pseudo greyscale image which represents the color space defined by a sorted palette. To sort the palette, a measure for color similarity is needed. Here an l/sub 2/ distance in the RGB and luminance-chrominance color space has been studied, and a new color correlation measure is proposed. For comparison a transform source coder (JPEG) is used to encode the pseudo greyscale image. Simulation results show that the subband source coder gives better results than the transform source coder. The quality of the decoded color images is surprisingly good.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"391 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122077998","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
The extended H/sub /spl infin// filter-a robust EKF 扩展的H/sub /spl滤波器-鲁棒EKF
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.389841
G. Einicke, L. White
{"title":"The extended H/sub /spl infin// filter-a robust EKF","authors":"G. Einicke, L. White","doi":"10.1109/ICASSP.1994.389841","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389841","url":null,"abstract":"The use of the extended Kalman filter (EKF) is heavily entrenched in non-linear signal processing applications. However linearisation errors inherent in the specification of an EKF can severely degrade its performance. The paper presents a new approach to the robustification of the EKF by application of robust linear design methods based on the H/sub /spl infin// norm minimisation criterion. The results of simulations are presented to demonstrate an advantage for the demodulation of frequency and phase modulated signals.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122172399","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: 0
High-quality harmonic coding at very low bit rates 高质量的谐波编码在非常低的比特率
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.389325
Gao Yang, H. Leich
{"title":"High-quality harmonic coding at very low bit rates","authors":"Gao Yang, H. Leich","doi":"10.1109/ICASSP.1994.389325","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389325","url":null,"abstract":"The paper presents a harmonic vocoder to produce high-quality speech at very low bit rates (below 2 kb/s). Voiced speech is decomposed into forward and backward signals which consist of interpolated harmonics. Unvoiced speech is reconstructed in the time domain with an approach similar to CELP. To remove the \"buzzy\" quality and avoid the \"hoarse\" quality, three methods are presented: the randomness of the harmonic phases is controlled according to pitch value and the continuity of synthetic speech is maintained; the spectral envelope determined by the LP model is modified; some noise components can be introduced for voiced synthetic speech. The harmonic vocoder produces quite natural, clear speech. Its perceptual quality is much better than that of the LPC-10 vocoder.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116602758","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
Estimation of the position of electrocortical generators via subspace techniques 利用子空间技术估计皮层电发生器的位置
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.389746
D. Klimovski, A. Sergejew, A. Cricenti, G. Egan
{"title":"Estimation of the position of electrocortical generators via subspace techniques","authors":"D. Klimovski, A. Sergejew, A. Cricenti, G. Egan","doi":"10.1109/ICASSP.1994.389746","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389746","url":null,"abstract":"There are a number of approaches to the application of subspace techniques for solving spectral estimation problems. These approaches are derived from the covariance matrix which is constructed from incoming data. The covariance matrix can be broken down through the use of appropriate matrix properties and eigen-decomposition techniques into two subspaces. The performance of three traditional algorithms which incorporate subspace techniques in direction of arrival are evaluated under both white and 1/f noise conditions. 1/f noise is chosen because it is typical of the EEG signals. Simulation results suggest that the Johnson and DeGraaf (1982) direction finding algorithm performs best under both noise environments. A typical sample of EEG data was used to evaluate the performance of the three algorithms. The Johnson and DeGraaf algorithm gives estimates for the direction of the signal which approximately agree with the anatomical locations of possible electrocortical generators.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"23 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129603093","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
Robust recursive spectral estimation based on AR model excited by a t-distribution process 基于t分布激励AR模型的鲁棒递归谱估计
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.389981
J. Sanubari, K. Tokuda, M. Onoda
{"title":"Robust recursive spectral estimation based on AR model excited by a t-distribution process","authors":"J. Sanubari, K. Tokuda, M. Onoda","doi":"10.1109/ICASSP.1994.389981","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389981","url":null,"abstract":"In this paper a new robust spectral estimation method based on an AR model is proposed. The optimal coefficient is selected by assuming that the excitation signal is t-distribution t(/spl alpha/) with /spl alpha/ degrees of freedom. The calculation is done by using a recursive algorithm. When /spl alpha/=/spl infin/, we get the RLS method. Simulation results show that the obtained estimates using the proposed method with small /spl alpha/ are more efficient, the standard deviation (SD) of the estimation results are smaller, and more accurate than that with large /spl alpha/. The proposed estimator with small /spl alpha/ is more efficient and more accurate then the recursive method based on Huber's M-estimate.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"62 6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128526031","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
A noise cancelling filter for the digital Holter monitoring system 数字式动态心电图监测系统的降噪滤波器
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.390051
T. Kohama, S. Nakamura, H. Hoshino
{"title":"A noise cancelling filter for the digital Holter monitoring system","authors":"T. Kohama, S. Nakamura, H. Hoshino","doi":"10.1109/ICASSP.1994.390051","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.390051","url":null,"abstract":"We are developing a complete digital Holter monitoring system (HMS). One important problem required with the digital HMS is the ability of huge data storage and processing. A conventional HMS uses an analog magnetic tape to record long-time data. Although recoding the huge data into a magnetic tape is the easiest way, it is not suitable for realizing a simple compact digital HMS. Our main objective is to realize a simple digital HMS. Therefore we have to recode the huge electrocardiogram (ECG) data into a semiconductor memory (IC memory card) instead of the magnetic tape. In order to record the long-time ECG signal into an IC memory card, it has to be compressed with a high ratio over 1/50. A noisy ECG signal with very low fluctuations cannot be compressed sufficiently. A noise cancelling method which is effective for the data compression, is proposed.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128730169","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
On the use of data-driven clustering technique for identification of poly- and mono-phonemes for four European languages 数据驱动聚类技术在四种欧洲语言多音位和单音位识别中的应用
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.389340
O. Andersen, P. Dalsgaard, W. Barry
{"title":"On the use of data-driven clustering technique for identification of poly- and mono-phonemes for four European languages","authors":"O. Andersen, P. Dalsgaard, W. Barry","doi":"10.1109/ICASSP.1994.389340","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389340","url":null,"abstract":"The research reported in this paper presents a method to identify poly- and mono-phonemes for four European languages. The functionality of the poly-phonemes is tested in two experiments, and a limited set of mono-phonemes is identified for a language-identification experiment. Ten acoustically-similar speech sounds were identified across the four languages British-English, Danish, German, and Italian. These sounds, which constitute a substantial proportion of the phonemes of each language, are designated as (language independent) poly-phonemes, and may serve as a multi-lingual training base for labelling and recognition systems. The remaining sounds of each language, which do not fulfil the similarity conditions, are dubbed mono-phonemes. Two application experiments were conducted. In the first the poly-phonemes are applied in a label alignment task. In the second a small selected of mono-phonemes for each of the four languages is used in a preliminary test of the ability of these sets to serve as language discriminators.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129370333","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: 32
Signal modeling using increments of extended self-similar processes 使用扩展自相似过程增量的信号建模
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.389855
Lance M. Kaplan, C.-C. Jay Kuo
{"title":"Signal modeling using increments of extended self-similar processes","authors":"Lance M. Kaplan, C.-C. Jay Kuo","doi":"10.1109/ICASSP.1994.389855","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389855","url":null,"abstract":"The article expands the fractional Brownian motion (fBm) model by investigating the idea of generalizing self-similarity to create extended self-similar (ESS) processes for which fBm processes are a subset. Properties of ESS processes are discussed, and an ESS increment model parameterized by variables controlling short and long term correlation effects is examined. We propose a fast parameter estimation algorithm for the new model which is based on the decorrelation effect of the Haar transform on the ESS increments, and we demonstrate the performance of this parameter estimation algorithm with numerical simulations.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"8 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124546636","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
Non-uniform unit parsing for SSS-LR continuous speech recognition ss - lr连续语音识别的非统一单元解析
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.389697
H. Singer, J. Takami, S. Matsunaga
{"title":"Non-uniform unit parsing for SSS-LR continuous speech recognition","authors":"H. Singer, J. Takami, S. Matsunaga","doi":"10.1109/ICASSP.1994.389697","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389697","url":null,"abstract":"We describe recent improvements in ATR's experimental speech recognition system ATREUS, which serves as a recognition font end for the speech translation system ASURA. Our next goal is spontaneous speech translation. To constrain the potentially huge search space, better prosodic control, better probabilistic language models and better acoustic models are proposed. The SSS-LR parser was modified to work with non-uniform unit type acoustic and duration models. Experimental results showed, that, for example, use of mora trigram probabilities improved the phrase error rate from 17% to 14%.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"126 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124559681","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
Non-linear input transformations for discriminative HMMs 判别hmm的非线性输入变换
Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing Pub Date : 1994-04-19 DOI: 10.1109/ICASSP.1994.389314
F. Johansen, M. H. Johnsen
{"title":"Non-linear input transformations for discriminative HMMs","authors":"F. Johansen, M. H. Johnsen","doi":"10.1109/ICASSP.1994.389314","DOIUrl":"https://doi.org/10.1109/ICASSP.1994.389314","url":null,"abstract":"This paper deals with speaker-independent continuous speech recognition. Our approach is based on continuous density hidden Markov models with a non-linear input feature transformation performed by a multilayer perceptron. We discuss various optimisation criteria and provide results on a TIMIT phoneme recognition task, using single frame (mutual information or relative entropy) MMI embedded in Viterbi training, and a global MMI criterion. As expected, global MMI is found superior to the frame-based criterion for continuous recognition. We further observe that optimal sentence decoding is essential to achieve maximum recognition rate for models trained by global MMI. Finally, we find that the simple MLP input transformation, with five frames of context information, can increase the recognition rate significantly compared to just using delta parameters.<<ETX>>","PeriodicalId":290798,"journal":{"name":"Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing","volume":"11 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1994-04-19","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129659509","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: 12
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