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An exact numerical algorithm for computing the unwrapped phase of a finite-length sequence 一种精确的数值算法，用于计算有限长度序列的展开相位
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196965
David G. Long
{"title":"An exact numerical algorithm for computing the unwrapped phase of a finite-length sequence","authors":"David G. Long","doi":"10.1109/ICASSP.1988.196965","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196965","url":null,"abstract":"A direct relationship between a one-dimensional time series and its unwrapped phase was shown by R. McGowan and R. Kuc (1982). They proposed an algorithm for computing the unwrapped phase by counting the number of sign changes in a Sturm sequence generated from the real and imaginary parts of the DFT (discrete Fourier transform). Their algorithm is limited to relatively short sequences by numerical accuracy. An extension of their algorithm is proposed which, by using all-integer arithmetic, permits exact computation of the number of multiples of pi which must be added to the principal value of the phase to uniquely give the unwrapped phase of a one-dimensional rational-valued finite-length sequence of arbitrary length. This extended algorithm should be of interest when highly accurate phase unwrapping is required.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132344987","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
Word synthesis based on line spectrum pairs 基于谱对的词合成
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196676
S. Everett
{"title":"Word synthesis based on line spectrum pairs","authors":"S. Everett","doi":"10.1109/ICASSP.1988.196676","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196676","url":null,"abstract":"The author described the initial investigation of a synthetic speech system based on line spectrum pair (LSP) analysis of the speech signal. The synthesizer contains a library of stored LSP speech segments extracted from natural speech. These segments are modified as necessary by a small set of context-sensitive rules and then concatenated to generate high-quality natural-sounding speech. Tests of a preliminary system produced MRT and DRT scores of 87.3 and 79.7, respectively. The LSP vocabulary synthesizer is limited to utterances of a single syllable, but future research will expand its capabilities to allow implementation of a full text-to-speech system.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"25 2","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132365033","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
Dynamic adaptation of Hidden Markov models for robust isolated-word speech recognition 基于隐马尔可夫模型的鲁棒孤立词语音识别
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196507
E. A. Martin, R. Lippmann, D. Paul
{"title":"Dynamic adaptation of Hidden Markov models for robust isolated-word speech recognition","authors":"E. A. Martin, R. Lippmann, D. Paul","doi":"10.1109/ICASSP.1988.196507","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196507","url":null,"abstract":"The authors describe an HMM-based isolated-word recognition system that dynamically adapts word model parameters to new speakers and to stress-induced speech variations. During recognition all input tokens presented to the system can be used to augment the current word model parameters. New tokens can be weighted so that adaptation simply increases the size of the training set, or tracks systematic changes by exponentially weighting all previously seen data. This system was tested on the 35-word 10710 token Lincoln stressed speech data base. Speaker adaptation experiments produced error rates equivalent to speaker-trained systems after the presentation of only a single new token per vocabulary word. Stress condition adaptation experiments produced results comparable to multistyle-trained systems after the presentation of several new tokens per vocabulary word.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"13 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132513878","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
Variable bit rate vector quantization of video images for packet-switched networks 分组交换网络中视频图像的可变比特率矢量量化
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196804
E. Daly, T. Hsing
{"title":"Variable bit rate vector quantization of video images for packet-switched networks","authors":"E. Daly, T. Hsing","doi":"10.1109/ICASSP.1988.196804","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196804","url":null,"abstract":"The authors investigate a novel vector-quantized compression strategy that provides guaranteed image quality with variable bit rate and show its applicability to packet-switched networks. This preliminary study indicates that significant improvements in both picture quality and SNR value can be achieved. The technique can be refined by reducing the number of overhead bits for each block and by separating coding schemes and high/low detail thresholds for each block size.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"232 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134123292","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: 17
Discriminant clustering using an HMM isolated-word recognizer 基于HMM孤立词识别器的判别聚类
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196506
R. Lippmann, E. A. Martin
{"title":"Discriminant clustering using an HMM isolated-word recognizer","authors":"R. Lippmann, E. A. Martin","doi":"10.1109/ICASSP.1988.196506","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196506","url":null,"abstract":"One limitation of hidden Markov model (HMM) recognizers is that subword models are not learned but must be prespecified before training. This can lead to excessive computation during recognition and/or poor discrimination between similar sounding words. A training procedure called discriminant clustering is presented that creates subword models automatically. Node sequences from whole-word models are merged using statistical clustering techniques. This procedure reduced the computation required during recognition for a 35-word vocabulary by roughly one-third while maintaining a low error rate. It was also found that five iterations of the forward-backward algorithm are sufficient and that adding nodes to HMM word models improves performance until the minimum word transition time becomes excessive.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"38 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133988734","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
Toeplitz-Hankel decomposition for direction of arrival estimation using transient signals 基于Toeplitz-Hankel分解的瞬态信号到达方向估计
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197137
E. Haberman, L. Griffiths
{"title":"Toeplitz-Hankel decomposition for direction of arrival estimation using transient signals","authors":"E. Haberman, L. Griffiths","doi":"10.1109/ICASSP.1988.197137","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197137","url":null,"abstract":"A method is described which can improve the direction-of-arrival estimation when the signal is nonstationary but of a known form. This is the case, e.g. in an active sonar system, when a known waveform is transmitted and the reflections are measured by an array of hydrophones. It is shown that for a class of signals, the knowledge about the signal can be used to improve the resolution in azimuth estimation.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"37 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134294158","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
Noise background normalization for simultaneous broadband and narrowband detection 同时宽带和窄带检测的噪声背景归一化
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197215
S. Davies, M. Knappe
{"title":"Noise background normalization for simultaneous broadband and narrowband detection","authors":"S. Davies, M. Knappe","doi":"10.1109/ICASSP.1988.197215","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197215","url":null,"abstract":"The authors examine the optimization of a normalizer for broadband detection. They then present a normalizer which allows simultaneous detection of both source-related features, narrowband and broadband, from frequency-time intensity displays of power spectral data. The technique combines the output of a normalizer optimized for narrowband detection with that of a normalizer optimized for broadband by taking the maximum output at each frequency. Its properties are discussed and performance demonstrated using simulated data.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"272 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134305064","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: 6
A new, highly parallel, 32 bit floating point DSP vector signal processor 一种新型的、高度并行的32位浮点DSP矢量信号处理器
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.197049
A. Genusov, Peter Feldman, R. Friedlander, V. Fruchter, R. Jaliff, A. Mohr, R. Shenhav
{"title":"A new, highly parallel, 32 bit floating point DSP vector signal processor","authors":"A. Genusov, Peter Feldman, R. Friedlander, V. Fruchter, R. Jaliff, A. Mohr, R. Shenhav","doi":"10.1109/ICASSP.1988.197049","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.197049","url":null,"abstract":"A new 32-bit floating point (IEEE standard) (digital signal processing) DSP vector signal processor architecture is described. The internal architecture is highly parallel. It is based on six well coordinated, independent machines. The ALU (arithmetic logic unit) has a pipeline structure optimized for the execution of DSP and matrix operations (FFT butterflies in particular). The highly flexible set of vectorized instructions allows for most efficient utilization of the internal assets. Together these features yield a high performance, high throughput processor with 31-Mflops computation power and very minimal overhead. A description is given of the architecture of the device, the different internal units and their coordination. The instruction set basic features are presented, and a few benchmarks of a single processor are given. A simple, minimal system architecture combining two processors sharing a single bus, doubling the throughput of a single processor system, is suggested.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"15 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131529345","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
Application of the Gibbs distribution to hidden Markov modeling in isolated word recognition 吉布斯分布在孤立词识别中隐马尔可夫建模中的应用
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196501
Yunxin Zhao, L. Atlas, X. Zhuang
{"title":"Application of the Gibbs distribution to hidden Markov modeling in isolated word recognition","authors":"Yunxin Zhao, L. Atlas, X. Zhuang","doi":"10.1109/ICASSP.1988.196501","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196501","url":null,"abstract":"A new method of formulating hidden Markov models (HMM) for isolated word recognition is presented. The authors model probabilities of hidden state sequences as Gibbs distributions (GDs) instead of the conventional products of transition probabilities. This formulation is based on the Hammersley-Clifford theorem which establishes the equivalence between Markov random fields (MRF) and GDs. The Markov chains in HMM are equivalent to one-dimensional, first order neighborhood MRFs. The observation sequences are modeled by the usual autoregressive Gaussian densities. The flexibility in the choice of energy functions in GDs makes it possible to use only a few parameters while maintaining a powerful model. The authors have developed a learning algorithm to estimate the parameters using maximum likelihood estimation and an algorithm to efficiently compute 1-D, first order neighborhood GDs using a lattice structure.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"112 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131679172","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 single-channel ROM-based complex digital filter implementation in the quadratic residue number systems 二次余数系统中基于单通道rom的复杂数字滤波器的实现
ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1988-04-11 DOI: 10.1109/ICASSP.1988.196981
R. Krishnan
{"title":"A single-channel ROM-based complex digital filter implementation in the quadratic residue number systems","authors":"R. Krishnan","doi":"10.1109/ICASSP.1988.196981","DOIUrl":"https://doi.org/10.1109/ICASSP.1988.196981","url":null,"abstract":"The implementation of complex digital filters using the quadratic residue number system (QRNs) and modified quadratic residue number system (MQRNS) is considered. These QRNS/MQRNS-based filter architectures are memory-intensive because the lookup-table approach has been used in the filter implementation. If the required number of lookup tables is reduced to a reasonable extent, single-chip VLSI implementation of such architectures may become practically feasible. A dual-clock computational module has been proposed to reduce the number of memory requirements in the QRNS/MQRNS-based complex digital filters. A direct FIR (finite-impulse response) filter architecture has been implemented using the proposed computational module in the QRNS and MQRNS schemes.<<ETX>>","PeriodicalId":448544,"journal":{"name":"ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing","volume":"52 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1988-04-11","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131739746","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
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