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ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing最新文献

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A methodology for evaluating the performance of dynamic range control algorithms for speech enhancement 一种评估语音增强动态范围控制算法性能的方法
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169753
John T. Lynch
{"title":"A methodology for evaluating the performance of dynamic range control algorithms for speech enhancement","authors":"John T. Lynch","doi":"10.1109/ICASSP.1987.1169753","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169753","url":null,"abstract":"This paper describes a system that has been developed for automatically measuring the enhancement in the peak/rms ratio that a given audio processor provides, and gives some preliminary results on the performance of three commercial audio processors and two rather simple processors. An important outcome of the results is the establishment of an engineering goal for an improved digital signal processing (DSP) speech enhancement algorithm.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"34 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116248241","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
Reconstruction from Fourier transform phase with applications to speech analysis 傅里叶变换相位重构及其在语音分析中的应用
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169708
B. Yegnanarayana, S. Fathima, H. Murthy
{"title":"Reconstruction from Fourier transform phase with applications to speech analysis","authors":"B. Yegnanarayana, S. Fathima, H. Murthy","doi":"10.1109/ICASSP.1987.1169708","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169708","url":null,"abstract":"This paper addresses the problem of signal reconstruction from Fourier transform phase. In particular, we examine two aspects of this problem. First, we discuss signal reconstruction from the phase spectrum of the short-time Fourier transform(STFT). Next, we examine the problem of signal recovery from partial phase information. We present the results of our studies on reconstruction from partial phase and discuss the application of these results in speech analysis and coding.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"5 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125582870","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
Tracking analysis of an ARMA parameter estimation algorithm using weak convergence theory 基于弱收敛理论的ARMA参数估计算法跟踪分析
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169596
B. Rao, R. Peng
{"title":"Tracking analysis of an ARMA parameter estimation algorithm using weak convergence theory","authors":"B. Rao, R. Peng","doi":"10.1109/ICASSP.1987.1169596","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169596","url":null,"abstract":"In this paper we study the problem of adaptively estimating the Autoregressive Moving Average (ARMA) parameters of a time varying ARMA process using a constant step size Gauss-Newton Algorithm. Using weak convergence theory and the concept of prescaling, it is shown that the \"mean\" behavior can be described by an ordinary differential equation (ODE). Computer simulations are provided to substantiate the analysis.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"23 6","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121005634","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
Techniques to increase the computational throughput of bit-serial architectures 提高位串行体系结构计算吞吐量的技术
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169696
Stewart Smith, M. S. McGregor, P. Denyer
{"title":"Techniques to increase the computational throughput of bit-serial architectures","authors":"Stewart Smith, M. S. McGregor, P. Denyer","doi":"10.1109/ICASSP.1987.1169696","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169696","url":null,"abstract":"Three architectural techniques are reported, which accelerate bit-serial computation without compromising its favourable advantages. In essence these techniques rely on multi-wire representations of serial data - a step towards bit-parallelism. Interfacing techniques are developed to support the existence of domains of different throughput within a system, thereby enhancing the range of bandwidth-matching techniques available to the systems designer. These techniques also realise the potential to mix processing wordlengths within a serial-data system.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"67 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131601310","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: 19
Sector-focused stability for high resolution array processing 面向扇区的高分辨率阵列处理稳定性
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169331
C. Byrne, A. Steele
{"title":"Sector-focused stability for high resolution array processing","authors":"C. Byrne, A. Steele","doi":"10.1109/ICASSP.1987.1169331","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169331","url":null,"abstract":"When the ambient noise field is spatially correlated the eigenvectors of the cross spectral matrix associated with the lowest eigenvalues are nearly orthogonal to all planewave arrivals within the region of noise concentration, making their nulls unstable for high resolution bearing estimation. By modifying the quadratic minimization problem that yields these eigenvectors as solutions, to include a subspace constraint dependent on a sector of interest in wavevector space, we obtain high resolution bearing estimators that are stable within the chosen sector, while reducing the size of the matrix on which we must operate.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"10 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130964764","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: 31
Full-span structural compilation of DSP hardware DSP硬件全跨度结构编译
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169715
Stewart Smith, P. Denyer, D. Renshaw, K. Asada, K. Coplan, M. Keightley, J. Mhar
{"title":"Full-span structural compilation of DSP hardware","authors":"Stewart Smith, P. Denyer, D. Renshaw, K. Asada, K. Coplan, M. Keightley, J. Mhar","doi":"10.1109/ICASSP.1987.1169715","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169715","url":null,"abstract":"Most current silicon compilers rely on an underlying hardware function-library, thereby restricting the user to producing structural assemblies of library components. Although this gives the systems designer the power to implement function directly in silicon, it precludes the use of arbitrary cell functions that might be more suited to his application than the existing set. Moreover, should a new process become available, the cell-library must be redesigned and laid-out according to new rules, by a circuit and layout expert. An advance in structural silicon compilation, SECOND, is reported, which promises to impart flexibility and portability to cell-libraries. Through the automatic generation of layout from logical descriptions, cell-libraries may be maintained in process-independent form, and incorporate not only new components, but also new processes with ease. Implementations of user-specified chips may be in custom or semi-custom form - only physical assembly procedures differ.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132792833","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
Discrete signal reconstruction from its autocorrelation function and one sample 从其自相关函数和一个样本重构离散信号
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169525
Zhongze Wu, Yanda Li, Tong Chang
{"title":"Discrete signal reconstruction from its autocorrelation function and one sample","authors":"Zhongze Wu, Yanda Li, Tong Chang","doi":"10.1109/ICASSP.1987.1169525","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169525","url":null,"abstract":"This work follows the 'Discrete Signal Reconstruction from its Spectral Magnitude and Some Samples'[1]. The uniqueness of the discrete signal reconstruction from its autocorrelation function and one sample has been discussed in detail in this paper. Four theorems are presented. In addition, we provide an effective iterative algorithm recovering the discrete signal.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"82 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115096877","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
A new idea of code book design in vector quantization of speech 语音矢量量化中码本设计的新思路
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169568
Ping Zheng, Hong-ji Zhang
{"title":"A new idea of code book design in vector quantization of speech","authors":"Ping Zheng, Hong-ji Zhang","doi":"10.1109/ICASSP.1987.1169568","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169568","url":null,"abstract":"The now available method of code book design in vector quantization coding of speech, based on the minimum distortion criterion and an iterative process, is time-consuming; and the code book's quality measurement, the average distortion, can't be predetermined. This paper suggests a new method of book design which is a one-time process and saves a lot of computer time. And for any given positive value ε we can obtain a satisfactory code book, with its average distortionDleqvarepsilon.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"17 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134462979","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
Design of a multistage decimation-interpolation filter 多级抽取插值滤波器的设计
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169830
V. Hansen
{"title":"Design of a multistage decimation-interpolation filter","authors":"V. Hansen","doi":"10.1109/ICASSP.1987.1169830","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169830","url":null,"abstract":"A 2 chip digital multi-rate FIR filter implemented in 2 micron CMOS performs 10 million multiplications and 20 million accumulations per second. The filter has 48 programmable bandwidths in a 1-2-5 sequence, and can either interpolate or decimate. This paper describes the design and implementation of the filter.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"19 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132964210","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
Image reconstruction from one-bit Fourier phase: Theory, sampling, and coherent image model 从一位傅里叶相位图像重建:理论，采样，和相干图像模型
ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1987-04-06 DOI: 10.1109/ICASSP.1987.1169763
Thomas Huang, J. Sanz, W. Blanz
{"title":"Image reconstruction from one-bit Fourier phase: Theory, sampling, and coherent image model","authors":"Thomas Huang, J. Sanz, W. Blanz","doi":"10.1109/ICASSP.1987.1169763","DOIUrl":"https://doi.org/10.1109/ICASSP.1987.1169763","url":null,"abstract":"In this paper, we tackle the problem of recovering an image from its Fourier transform phase quantized to 1 bit, or, equivalently, from the zero crossings of the real part of the Fourier transform. We first present new theoretical results that set an algebraic condition under which real zero crossings uniquely specify a band-limited image. We then show, however, through a large-scale set of experiments, that sampling in the frequency domain presents a major obstacle to good reconstruction resuits due to the information loss produced by the approximated knowledge of the zero crossing locations. We finally show that, by using a \"coherent\" image model in which the image is complex and the spatial-domain phase is random and highly uncorrelated, we can significantly reduce the effect of this information loss and improve the quality of image reconstruction.","PeriodicalId":140810,"journal":{"name":"ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1987-04-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124667937","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
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