{"title":"Self-calibration of linear equi-spaced (LES) arrays","authors":"M. Wylie, Sumit Roy, R. F. Schmitt","doi":"10.1109/ICASSP.1993.319110","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319110","url":null,"abstract":"The problem of self-calibration of phased arrays for unknown gain and phase errors in conjunction with their use in high-resolution direction of arrival (DOA) estimation is addressed. The authors consider the problem for the canonical (and widely used) linear equispaced (LES) array with a constant but unknown gain and phase response and develop an algorithm that does not require any prior knowledge of the direction of arrival of a signal. Analytical results are presented regarding estimator performance and are verified by a representative body of simulations.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"269 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":"122468011","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}
J. Tardelli, P. Gatewood, E. W. Kreamer, P. L. Follette
{"title":"The benefits of multi-speaker conferencing and the design of conference bridge control algorithm","authors":"J. Tardelli, P. Gatewood, E. W. Kreamer, P. L. Follette","doi":"10.1109/ICASSP.1993.319333","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319333","url":null,"abstract":"The authors examine the benefits to be gained with a multispeaker narrowband conference capability and design parameters for such a system. Subjective opinions on 18 conferencing systems by 16 subjects, audio recordings of input and output channels for each subject and system, and data files that detail the speech activity and channel switch settings for each subject and system every 50 ms have been provided. An important conferencing objective or conference type effect has been found: for different types of discussion the speech dynamics and system requirements are substantially different. Various speech/bridge dynamic states are evaluated. The percent time that the system is in these states is provided along with interval distributions.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"106 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":"122603801","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}
{"title":"Normalization and convergence of adaptive IIR filters","authors":"M. Rupp","doi":"10.1109/ICASSP.1993.319551","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319551","url":null,"abstract":"Several algorithms for adaptive IIR (infinite impulse response) filters have been proposed. However, their practical use involves considerations such as finding the global minimum, the possible occurrence of instability, and uncertainty about the speed of convergence. A generalization of these adaptive IIR algorithms is presented here. The algorithms can be classified into two groups: those which do not filter the adaptation error and those which do. For the first group a normalization is presented and convergence is proven. For the second group, ideas for normalizations are presented and conditions for convergence are given. Special constraints for normalizations are also considered.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"7 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":"123025231","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}
{"title":"Tracking of partials for additive sound synthesis using hidden Markov models","authors":"P. Depalle, G. García, X. Rodet","doi":"10.1109/ICASSP.1993.319096","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319096","url":null,"abstract":"The authors present a sinusoidal partial tracking method for additive synthesis of sound. Partials are tracked by identifying time functions of parameters as underlying trajectories in a successive set of spectral peaks. This is done by a purely combinatorial hidden Markov model. A partial trajectory is considered as a time sequence of peaks which satisfies continuity constraints on parameter slopes. The method allows frequency line crossing and can be used for formant tracking.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"234 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":"123037589","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}
{"title":"Hybrid survivor path architectures for Viterbi decoders","authors":"P. Black, T. Meng","doi":"10.1109/ICASSP.1993.319148","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319148","url":null,"abstract":"A new approach to survivor path architectures for Viterbi decoders is proposed based on hybrid architectures that combine the classical register-exchange and trace-back methods. Two classes of hybrid architecture are proposed: the hybrid pretrace-back architecture and the hybrid trace-forward architecture. Pretrace-back is a preprocessing of the add-compare-select (ACS) decisions to increase the effective trace-back recursion rate, while trace-forward is a concurrent processing of the decisions to initialize the trace-back recursion. Both of these architectures can be implemented using a single compact memory, typically twice the survivor path length in size, without any loss in throughput compared with conventional trace-back architectures. Based on area estimates for common decoding problems the hybrid architectures reduce the required chip area by up to 40% compared with the popular k-pointed trace-back architecture.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"23 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":"114034711","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}
{"title":"An efficient prime-factor algorithm for the discrete cosine transform and its hardware implementations","authors":"Peizong Lee, Fang-Yu Huang","doi":"10.1109/ICASSP.1993.319487","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319487","url":null,"abstract":"The input index mapping adopted is the Ruritanian mapping; the output index mapping is the same as B.G. Lee's (IEEE Trans. vol. ASSP.37, no.2, p.237-44, Feb. 1989). Hardware implementations for the prime-factor DCT are also studied. The methodology, which deals with general (N/sub 1/*N/sub 2/)-point DCTs, where N/sub 1/ and N/sub 2/ are mutually prime, is illustrated by converting a 15-point DCT problem into a (3*5)-point 2-D DCT problem.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"33 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":"117031226","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}
{"title":"A new word pre-selection method based on an extended redundant hash addressing for continuous speech recognition","authors":"A. Ito, S. Makino","doi":"10.1109/ICASSP.1993.319296","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319296","url":null,"abstract":"This method is based on the isolated word recognition method using the redundant hash addressing principle proposed by T. Kohonen et al. (Inf. Sci. vol.33, p.3-30 of 1984). The redundant hash addressing method cannot be used for continuous speech recognition because the word boundary in continuous speech is not known. Kohonen's method has been extended to continuous speech recognition. The proposed method spots words from the input speech using information about locations of features in reference patterns. The combination of this method and an ordinary parser makes a speech recognition system faster. Experimental results are shown, comparing this method with the continuous-DP matching method, a traditional word spotting method. While the proposed method gives performance comparable with that of continuous-DP matching, it is five times faster.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"2 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":"129813496","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}
{"title":"Low-delay wideband speech coding using a new frequency domain approach","authors":"V. Sánchez, J. Adoul","doi":"10.1109/ICASSP.1993.319328","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319328","url":null,"abstract":"Working in the context of residual speech coders, the proposed technique performs a decomposition of the DFT (discrete Fourier transform) of the target vector (the input speech after the subtraction of the zero input response signal) as the product of the DFT of the impulse response of the LPC (linear predictive coding) synthesis filter and a modified residual signal. At the receiver, the DFT of the target vector is reconstructed by multiplying the quantized modified residual signal by the DFT of the impulse response of the synthesis filter. The inverse DFT is then calculated, and the resulting time-domain quantized target vector is filtered by the inverse synthesis filter in order to get the excitation to the LPC synthesis filter. Magnitude-phase and real-part-imaginary-part encoding of the modified residual signal has been studied, and several quantization schemes have been used. Specifically, scalar quantization and trellis coded quantization have been used in the development of 32-kbit/s coders with a very high quality of the synthesized speech. A 24-kbit/s coder has also been developed by using vector quantization and trellis-coded quantization in a magnitude-phase representation of the modified residual.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"26 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":"128390833","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}
{"title":"An evaluation of time-frequency localization in transforms and filter banks","authors":"Yipeng Liu, A. Akansu","doi":"10.1109/ICASSP.1993.319485","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319485","url":null,"abstract":"The authors derive the discrete-time uncertainty and evaluate the time-frequency localizations of orthogonal bases. They measure and compare the localization features of known block transforms, filter banks, and wavelet filters. It is emphasized that the time and frequency behaviors of basis functions should be considered simultaneously in the design of block transforms and filter banks for further improvements in their applications.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"35 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":"128794566","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}
{"title":"Adaptive impulse noise filtering","authors":"S. R. Kim, A. Efron","doi":"10.1109/ICASSP.1993.319622","DOIUrl":"https://doi.org/10.1109/ICASSP.1993.319622","url":null,"abstract":"The authors present a preprocessor to mitigate the effects of the impulse noise. In contrast with previous work, the proposed preprocessor requires no information about the correlated background process and can adapt as the background process changes. An adaptive nonlinear filter (preprocessor) introduced mitigates the effects of the impulses when the nominal process (background noise) is a correlated Gaussian process. The preprocessing procedure consists of identifying the impulsive component, eliminating it and replacing the samples with an estimate. Comparison of techniques for suppressing impulse noise is given.<<ETX>>","PeriodicalId":428449,"journal":{"name":"1993 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"20 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":"128308501","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}