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[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing最新文献

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Adaptive algorithms for underdetermined active control problems 欠定主动控制问题的自适应算法
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226076
S. Elliott, J. Rex
{"title":"Adaptive algorithms for underdetermined active control problems","authors":"S. Elliott, J. Rex","doi":"10.1109/ICASSP.1992.226076","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226076","url":null,"abstract":"In active sound control, the sound field generated by the operation of a number of secondary actuators is adjusted to destructively interfere, to the greatest possible extent, with the sound field caused by the original, primary, source of sound. An example would be the adjustment of the inputs to structural actuators attached to a panel, also excited by a primary source of vibration, to minimize the acoustic power radiated. In practice the inputs to a number of actuators could be adjusted to minimize the sum of the squared signals from a greater number of sensors-an overdetermined control problem, for which adaptive algorithms have already been developed. An alternative control strategy is to adjust the actuators so that the signal from a single sensor, such as one measuring the net volume velocity of a panel, is driven to zero with the least control effort. Adaptive algorithms for such underdetermined control problems are discussed.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"92 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126187436","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
Subband coding of video using an edge-based vector quantization technique for compression of the upper bands 视频的子带编码使用基于边缘的矢量量化技术压缩上带
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226208
N. Mohsenian, N. Nasrabadi
{"title":"Subband coding of video using an edge-based vector quantization technique for compression of the upper bands","authors":"N. Mohsenian, N. Nasrabadi","doi":"10.1109/ICASSP.1992.226208","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226208","url":null,"abstract":"A new subband video coding technique is introduced which utilizes several memoryless vector quantizing (VQ) schemes for encoding the various layers. A set of quadrature mirror filter banks was applied to the motion compensated frame differences (MCFDs) of the moving sequences to produce seven nonuniform bands. The upper bands displayed a significant amount of information at edge locations, these being the positions where the baseband of each layer was observed to have a similar behavior. Therefore, an edge-detecting operator, e.g., Laplacian or a Gaussian, was incorporated into the video compression model to extract the perceptually important locations of the upper bands from their correspondingly encoded baseband, thus eliminating the need for transmission of their addresses. Promising results were obtained which are suitable for low-bit-rate video applications where videophone and videoconferencing systems may be realized.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123474351","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: 13
Design of stable two-dimensional IIR digital filters with arbitrary magnitude function 任意幅值函数的稳定二维IIR数字滤波器的设计
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226650
Takao Kobayashi, Kazuyoshi Fukushi, K. Tokuda, S. Imai
{"title":"Design of stable two-dimensional IIR digital filters with arbitrary magnitude function","authors":"Takao Kobayashi, Kazuyoshi Fukushi, K. Tokuda, S. Imai","doi":"10.1109/ICASSP.1992.226650","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226650","url":null,"abstract":"A technique for designing two-dimensional (2-D) digital filters which can approximate arbitrary magnitude functions is proposed. A 2-D spectral factorization technique is used to obtain the recursively computable and stable system with nonsymmetric half-plane support from a given 2-D magnitude function. A new class of realizable 2-D digital filters referred to as 2-D log magnitude approximation (2-D LMA) filters are used to approximate the system obtained by the 2-D factorization. The design procedure is straightforward and computationally efficient. A simple stability condition which guarantees the stability of the designed 2-D LMA filter is given. An efficient network structure of 2-D LMA filters for parallel implementation is also discussed.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"154 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116100172","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
Extracting in-phase and quadrature signal components from a bandlimited real signal using a closed form optimal (MSE) halfband multirate filter design and its implementation on the Motorola DSP56001/DSP56ADC16 利用封闭形式最优(MSE)半带多速率滤波器设计从带宽有限的真实信号中提取同相和正交信号分量，并在摩托罗拉DSP56001/DSP56ADC16上实现
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226305
P. D. West, M. D. Austin
{"title":"Extracting in-phase and quadrature signal components from a bandlimited real signal using a closed form optimal (MSE) halfband multirate filter design and its implementation on the Motorola DSP56001/DSP56ADC16","authors":"P. D. West, M. D. Austin","doi":"10.1109/ICASSP.1992.226305","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226305","url":null,"abstract":"Extraction of the in-phase (I) and quadrature (Q) components from an RF signal is an important problem in radar and communication systems. In analog systems, the image spur suppression is a function of the phase and amplitude balance between the mixers and splitters in the two channel of the I/Q demodulation network. In the proposed system, an all digital scheme is described that allows arbitrary image spur rejection to be achieved by increasing the filter coder. An MMSE optimal FIR halfband filter is proposed and results are presented from a real-time implementation on the Motorola DSP56001 digital signal processor and the DSP56ADC16 delta-sigma A/D converter.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123031860","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
An integrated speech-background model for robust speaker identification 一种集成语音背景模型的鲁棒说话人识别
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226089
Douglas A. Reynolds, R. Rose
{"title":"An integrated speech-background model for robust speaker identification","authors":"Douglas A. Reynolds, R. Rose","doi":"10.1109/ICASSP.1992.226089","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226089","url":null,"abstract":"A procedure for text-independent speaker identification in noisy environments where the interfering background signals cannot be characterized using traditional broadband or impulsive noise models is examined. In the procedure, both the speaker and the background processes are modeled using mixtures of Gaussians. Speaker and background models are integrated into a unified statistical framework allowing the decoupling of the underlying speech process from the noise corrupted observations via the expectation-minimization algorithm. Using this formalism, speaker model parameters are estimated in the presence of the background process, and a scoring procedure is implemented for computing the speaker likelihood in the noise corrupted environment. The performance was evaluated using a 16-speaker conversational speech database with both speech babble and white noise background processes.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129828080","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: 10
Systolic array implementations for Chebyshev nonuniform sampling Chebyshev非均匀采样的收缩数组实现
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226457
Y. S. Zhu, S. Leung
{"title":"Systolic array implementations for Chebyshev nonuniform sampling","authors":"Y. S. Zhu, S. Leung","doi":"10.1109/ICASSP.1992.226457","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226457","url":null,"abstract":"A Chebyshev-type interpolation function (CTIF) based on Chebyshev nonuniform sampling is developed, and some important properties of the CTIF are discussed. It has been demonstrated that the concept of signal reconstruction from the nonuniform samples can be described clearly by using the CTIF. Two implementation schemes, a time domain structure and a transformed domain structure using systolic array architecture for this method, are also discussed. The schemes have the advantages of simplicity, regularity, and fast reconstruction, and they are suitable for data compression.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"4 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129869387","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
Performance analysis of a class of transient detection algorithms 一类暂态检测算法的性能分析
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226017
B. Porat, B. Friedlander
{"title":"Performance analysis of a class of transient detection algorithms","authors":"B. Porat, B. Friedlander","doi":"10.1109/ICASSP.1992.226017","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226017","url":null,"abstract":"The detection of transient signals is often performed in the time-frequency transform domain. An analytical framework is developed within which the performance of different detectors based on linear transforms can be easily compared, for different classes of signals. This framework is used to evaluate and compare the performance of detectors based on the Gabor (1946) transform and on the short-time Fourier transform.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128455465","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
Signal approximation via data-adaptive normalized Gaussian functions and its applications for speech processing 基于数据自适应归一化高斯函数的信号逼近及其在语音处理中的应用
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.225952
S. Qian, Dapang Chen, Ke-Shiu Chen
{"title":"Signal approximation via data-adaptive normalized Gaussian functions and its applications for speech processing","authors":"S. Qian, Dapang Chen, Ke-Shiu Chen","doi":"10.1109/ICASSP.1992.225952","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225952","url":null,"abstract":"A signal approximation via data-adaptive normalized Gaussian functions is presented. This approach resembles the traditional Gabor expansion, but it is more precise and efficient. Numerical simulations for the speech signal are included to demonstrate the effectiveness of the new scheme.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"73 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128798770","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: 26
Segmentation and mapping of highly convoluted contours with applications to medical images 高度复杂轮廓的分割和映射及其在医学图像中的应用
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226149
C. Davatzikos, Jerry L Prince
{"title":"Segmentation and mapping of highly convoluted contours with applications to medical images","authors":"C. Davatzikos, Jerry L Prince","doi":"10.1109/ICASSP.1992.226149","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226149","url":null,"abstract":"A method that simultaneously identifies the central layer of the human cortex and maps it onto the interval","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128688690","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
Modeling state durations in hidden Markov models for automatic speech recognition 自动语音识别的隐马尔可夫模型状态持续时间建模
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.225892
P. Ramesh, J. Wilpon
{"title":"Modeling state durations in hidden Markov models for automatic speech recognition","authors":"P. Ramesh, J. Wilpon","doi":"10.1109/ICASSP.1992.225892","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225892","url":null,"abstract":"Hidden Markov modeling (HMM) techniques have been used successfully for connected speech recognition in the last several years. In the traditional HMM algorithms, the probability of duration of a state decreases exponentially with time which is not appropriate for representing the temporal structure of speech. Non-parametric modeling of duration using semi-Markov chains does accomplish the task with a large increase in the computational complexity. Applying a postprocessing state duration penalty after Viterbi decoding adds very little computation but does not affect the forward recognition path. The authors present a way of modeling state durations in HMM using time-dependent state transitions. This inhomogeneous HMM (IHMM) does increase the computation by a small amount but reduces recognition error rates by 14-25%. Also, a suboptimal implementation of this scheme that requires no more computation than the traditional HMM is presented which also has reduced errors by 14-22% on a variety of databases.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1992-03-23","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129007959","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: 93
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