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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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A fast VQ codebook design algorithm for a large number of data 一种针对大数据量的快速VQ码本设计算法
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.225960
M. Nakai, H. Shimodaira, Masayuki Kimura
{"title":"A fast VQ codebook design algorithm for a large number of data","authors":"M. Nakai, H. Shimodaira, Masayuki Kimura","doi":"10.1109/ICASSP.1992.225960","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225960","url":null,"abstract":"The authors point out that the LBG algorithm (see Y. Linde et al., (1980)) requires a lot of computation as the training vectors increase, and proposes a fast VQ (vector quantization) algorithm for a large amount of training data. This algorithm consists of three steps: first, divide training vectors into small groups; second, quantize each group into a few codewords by the LBG algorithm; finally, construct a codebook by clustering these codewords using the LBG algorithm again. The authors also report they can reduce the distortion error of the algorithm by adapting an effective data-dividing method. In experiments of quantizing 17500 training vectors into 512 codewords, this algorithm requires only 1/6 computation time compared with the conventional algorithm, while the increase of distortion is only 0.5 dB.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"6 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":"114973421","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
Cramer Rao bounds on direction estimates for closely spaced emitters in multi-dimensional applications 多维应用中近间隔发射体方向估计的Cramer Rao界
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226007
J. Jachner, H. Lee
{"title":"Cramer Rao bounds on direction estimates for closely spaced emitters in multi-dimensional applications","authors":"J. Jachner, H. Lee","doi":"10.1109/ICASSP.1992.226007","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226007","url":null,"abstract":"The main results presented are characterizations of the Cramer-Rao (CR) bound on the variance of direction estimates for closely spaced emitters in multiple parameter (multi-D) scenarios. Specifically, simple analytic expressions for the CR bound are presented for colinear emitter configurations, which show the bound to be very sensitive to the maximum spacing between emitters ( delta omega ). Results also are cited for CR bound sensitivity to delta omega for emitter configurations in which the emitters are not colinear. The latter results exhibit greatly reduced sensitivity to the direction separation factor delta omega . Thus, the results show that degeneracies are present in multi-D parameter estimation scenarios that are not present in 1-D scenarios. Specifically, emitter resolution and direction estimation can be expected to be much more challenging for some emitter configurations than for others. The case of colinear emitters appears to be a particularly stressful one.<<ETX>>","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":"115005874","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
Automatic generation of architectural models for designing dedicted VLIW signal processors 设计专用VLIW信号处理器的自动生成体系结构模型
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226563
G. Menez, M. Auguin, Fernand Boéri, C. Carrière
{"title":"Automatic generation of architectural models for designing dedicted VLIW signal processors","authors":"G. Menez, M. Auguin, Fernand Boéri, C. Carrière","doi":"10.1109/ICASSP.1992.226563","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226563","url":null,"abstract":"Some results of the CAPSYS method are presented. It is a new high-level synthesis method whose objective is to benefit from very long induction word (VLIW) compilation and digital signal processor (DSP) synthesis experiences to generalize the automatic processor design process. The major characteristic of this approach is to synthesize architecture in a global way. Instead of a single operative area, the synthesis considers a complete architecture. It permits one to take into account memory throughput, for instance, and, more generally, exchanges with the environment of the synthesized processor. The first part of the method, the generation of architectural models, is emphasized. This procedure furnishes a full spectrum of architectural decompositions (i.e. the number and the type of the functional units) from maximally parallel ones to fully serial ones. In each one, the synthesis algorithm creates an adequacy between the machine and the microcode. Results showing the influence of memory management on synthesized processors are given.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"20 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":"115479175","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
Tree searched multi-stage vector quantization of LPC parameters for 4 kb/s speech coding 4 kb/s语音编码中LPC参数的树搜索多阶段矢量量化
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.225961
B. Bhattacharya, W. LeBlanc, S. Mahmoud, V. Cuperman
{"title":"Tree searched multi-stage vector quantization of LPC parameters for 4 kb/s speech coding","authors":"B. Bhattacharya, W. LeBlanc, S. Mahmoud, V. Cuperman","doi":"10.1109/ICASSP.1992.225961","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225961","url":null,"abstract":"The authors present a tree searched multi-stage vector quantization (TS-MSVQ) scheme which achieves spectral distortion lower than 1 dB with low complexity and good robustness using 24 b/frame. The M-L search is used and it is shown that it achieves performance close to that of the optimal search for a relatively small M. The best performance/complexity trade-offs are obtained with relatively small size codebooks cascaded in a three-four stage configuration. Results for log-area ratio (LAR) and line spectral pain (LSP) parameters are presented. A training technique which reduces outliers at the expense of a slight average performance degradation is introduced. The robustness across different languages and input spectral shapings is studied. Finally, it is shown that TS-MSVQ significantly outperforms the split-codebook approach.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"107 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":"115491759","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
Hands-free voice communication in an automobile with a microphone array 带麦克风阵列的汽车免提语音通信
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.225916
Stephen Oh, V. Viswanathan, P. Papamichalis
{"title":"Hands-free voice communication in an automobile with a microphone array","authors":"Stephen Oh, V. Viswanathan, P. Papamichalis","doi":"10.1109/ICASSP.1992.225916","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.225916","url":null,"abstract":"The authors present the result of their research on developing a hands-free voice communication system with a microphone array for use in an automobile environment. The goal of this research is to develop a speech acquisition and enhancement system so that a speech recognizer can reliably be used inside a noise automobile environment, for digital cellular phone application. Speech data have been collected using a microphone array and a digital audio tape (DAT) recorder inside a real car for several idling and driving conditions, and processed using delay-and-sum and adaptive beamforming algorithms. Performance criteria including signal-to-noise ratio and speech recognition error rate have been evaluated for the processed data. Detailed performance results presented show that the microphone array is superior to a single microphone.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"58 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":"115710530","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: 68
Silicon compilation of video-rate digital filters 视频速率数字滤波器的硅编译
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226545
T. Yoshino, H. Nishimura
{"title":"Silicon compilation of video-rate digital filters","authors":"T. Yoshino, H. Nishimura","doi":"10.1109/ICASSP.1992.226545","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226545","url":null,"abstract":"The design and implementation of a silicon compiler system for video-rate digital filters are presented. The compiler is capable of mapping arbitrary linear filters, given in the form of a ratio of polynomials in Z/sup -1/ or signal flow graph, into compact VLSI layouts. Filters from actual video processing systems are synthesized to demonstrate the effectiveness of the compiler.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"68 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":"123106312","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
Effects of mutual coupling on super-resolution DF in linear arrays 相互耦合对线性阵列超分辨DF的影响
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226521
C. Roller, W. Wasylkiwskyj
{"title":"Effects of mutual coupling on super-resolution DF in linear arrays","authors":"C. Roller, W. Wasylkiwskyj","doi":"10.1109/ICASSP.1992.226521","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226521","url":null,"abstract":"The traditional narrowband array signal processing model is expanded by incorporating the effects of mutual coupling via a closed-form expression for the array impedance matrix. The mutual coupling matrix affects the incident signal like a set of narrowband beamformers; this interpretation is explored. The array's super-resolution direction finding (SRDF) behavior/performance in the presence of mutual coupling is explored. Simulation results examine the effects of mismatch in antenna termination impedance upon DF performance, using parameters from a typical mobile cellular radio environment scenario. The results indicate that array DF performance can be improved by mismatching the array terminations, thus reducing the effect of mutual coupling.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"44 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":"121827363","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: 33
A hybrid neural network, dynamic programming word spotter 一种混合神经网络，动态规划词识别器
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226116
T. Zeppenfeld, A. Waibel
{"title":"A hybrid neural network, dynamic programming word spotter","authors":"T. Zeppenfeld, A. Waibel","doi":"10.1109/ICASSP.1992.226116","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226116","url":null,"abstract":"A novel keyword-spotting system that combines both neural network and dynamic programming techniques is presented. This system makes use of the strengths of time delay neural networks (TDNNs), which include strong generalization ability, potential for parallel implementations, robustness to noise, and time shift invariant learning. Dynamic programming models are used by this system because they have the useful capability of time warping input speech patterns. This system was trained and tested on the Stonehenge Road Rally database, which is a 20-keyword-vocabulary, speaker-independent, continuous-speech corpus. Currently, this system performs at a figure of merit (FOM) rate of 82.5%. FOM is the detection rate averaged from 0 to 10 false alarms per keyword hour. This measure is explained in detail.<<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":"121028980","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: 40
A compiler for multiprocessor DSP implementation 一个用于多处理器DSP实现的编译器
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226553
P. Hoang, J. Rabaey
{"title":"A compiler for multiprocessor DSP implementation","authors":"P. Hoang, J. Rabaey","doi":"10.1109/ICASSP.1992.226553","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226553","url":null,"abstract":"McDAS, a software environment designed to support the real-time implementation of digital signal processing (DSP) applications onto multiple processors, is described. Users program their algorithms as they would on a single processor, and McDAS automatically schedules and compiles the program onto the target multiprocessor. The scheduler maximizes the computational throughput by simultaneously considering pipelining, retiming, and parallelism while accounting for processor and memory constraints, as well as interprocessor communication delays. If the architecture is scalable or configurable, the scheduler can be invoked with different numbers of processors and multiprocessor topologies to explore various implementations. The code generator is similarly retargetable to different memory architectures and core processors. Data buffers and synchronizations are automatically inserted to ensure correct execution. The code generated can also execute the algorithms with either quasi-infinite precision or bit-true precision, allowing the algorithm designer to assess the effects of quantization and truncation. The results on a set of benchmarks are presented.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"67 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":"127109400","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
Scaling exponents estimation from time-scale energy distributions 基于时间尺度能量分布的标度指数估计
[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing Pub Date : 1992-03-23 DOI: 10.1109/ICASSP.1992.226634
P. Gonçalves, P. Flandrin
{"title":"Scaling exponents estimation from time-scale energy distributions","authors":"P. Gonçalves, P. Flandrin","doi":"10.1109/ICASSP.1992.226634","DOIUrl":"https://doi.org/10.1109/ICASSP.1992.226634","url":null,"abstract":"It is shown using some examples that the problem of estimating the evolution of scaling exponents characterizing locally a self-similar process can be efficiently handled within the general framework of time-scale energy distributions related to the wavelength transform. As is implicit from the structure of the estimators considered, the proposed methodology is dependent on the degree of nonstationarity of such evolutions, with fast changes leading to bias-variance tradeoffs.<<ETX>>","PeriodicalId":163713,"journal":{"name":"[Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing","volume":"5 6 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":"126094748","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
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