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2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)最新文献

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Generalized unequal length lapped orthogonal transform for subband image coding 子带图像编码的广义不等长重叠正交变换
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.862032
T. Nagai, M. Ikehara, M. Kaneko, A. Kurematsu
{"title":"Generalized unequal length lapped orthogonal transform for subband image coding","authors":"T. Nagai, M. Ikehara, M. Kaneko, A. Kurematsu","doi":"10.1109/ICASSP.2000.862032","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.862032","url":null,"abstract":"In this paper, generalized linear phase lapped orthogonal transforms with unequal length basis functions (GULLOT) are considered. The length of each basis of the proposed GULLOT can be different from each other, while all the bases of the conventional GenLOT are of equal length. In order to apply the GULLOT to subband image coding, we also investigate the size-limited structure to process the finite length signal which is important in practice.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"130286468","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: 12
Smooth wavelet frames with application to denoising 平滑小波帧与应用去噪
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.861887
I. Selesnick, L. Sendur
{"title":"Smooth wavelet frames with application to denoising","authors":"I. Selesnick, L. Sendur","doi":"10.1109/ICASSP.2000.861887","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.861887","url":null,"abstract":"This paper considers the design and application of wavelet tight frames based on iterated oversampled filter banks. The greater design freedom available makes possible the construction of wavelets with a high degree of smoothness, in comparison with orthonormal wavelet bases. Grobner bases are used to obtain the solutions to the nonlinear design equations. Following the dual-tree DWT of Kingsbury (see Proceedings of the Eighth IEEE DSP Workshop, Utah, 1998, and Proc. IEEE Int. Conf. Acoust., Speech, Signal Processing (ICASSP), Phoenix, 1999), one goal is to keep the redundancy-factor bounded by 2, instead of allowing it to grow as it does for the undecimated DWT (which is exactly shift-invariant). For the tight frame presented here, optimal-tree based denoising algorithms can be directly applied.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"380 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134076578","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: 21
DSP implementation issues for UMTS-channel coding umts信道编码的DSP实现问题
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.860085
U. Walther, G. Fettweis
{"title":"DSP implementation issues for UMTS-channel coding","authors":"U. Walther, G. Fettweis","doi":"10.1109/ICASSP.2000.860085","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.860085","url":null,"abstract":"The new wireless communication standard UMTS applies an advanced dual-mode channel coding scheme. We investigate the feasibility of implementing the algorithm on a digital signal processor device and the implication upon the processor architecture. Starting with a base architecture which allows for scalability and customization we derive new system parameters and compare the total device to ASIC solutions.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"2 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134127720","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
Low-band extension of telephone-band speech 电话频段语音的低频段扩展
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.862116
G. Miet, A. Gerrits, J. Valière
{"title":"Low-band extension of telephone-band speech","authors":"G. Miet, A. Gerrits, J. Valière","doi":"10.1109/ICASSP.2000.862116","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.862116","url":null,"abstract":"This paper describes a system that generates a low-band signal (100-300 Hz) from a telephone-band (300-3400 Hz) speech signal to obtain an extended-band speech signal (100-3400 Hz). The low-band increases signal naturalness and listening comfort. This system is applied at the receiving end such that compatibility with all current telephone networks is maintained. The described technique splits the telephone-band speech signal into a spectral envelope and a short-term residual. The spectral envelope and the residual are extended separately and recombined to create an extended band signal. This system is evaluated by listening tests and distortion measurement.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134235338","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: 41
Design of blind decision feedback equalizers for Markovian time varying channels 马尔可夫时变信道盲决策反馈均衡器的设计
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.861077
S. Cherif, M. Alouane, Mériem Jaïdane
{"title":"Design of blind decision feedback equalizers for Markovian time varying channels","authors":"S. Cherif, M. Alouane, Mériem Jaïdane","doi":"10.1109/ICASSP.2000.861077","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.861077","url":null,"abstract":"In this paper, a new class of blind algorithms designed for decision feedback equalization of time varying channels, is proposed. We consider Markovian time variations of the impulse response of the channel as in radio mobile communications. The main idea is to modify classical blind algorithms (decision-directed, constant modulus algorithm,...) in order to give them self-adaptive knowledge of the channel non-stationarity. Simulations show that the proposed algorithms non-stationary DD and non-stationary CMA present better tracking capacity than the classical ones. Hence, they are able to improve the bit error rate especially for severe propagation conditions.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"128 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134360519","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
Efficient integration of multiple pronunciations in a large vocabulary decoder 有效整合多个发音在一个大的词汇解码器
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.862068
H. Schramm, X. Aubert
{"title":"Efficient integration of multiple pronunciations in a large vocabulary decoder","authors":"H. Schramm, X. Aubert","doi":"10.1109/ICASSP.2000.862068","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.862068","url":null,"abstract":"The paper describes the improved handling of multiple pronunciations achieved in the Philips research decoder by (1) incorporating some prior information about their distributions and (2) combining the acoustic contributions of concurrent alternate word hypotheses. Starting from a baseline system where multiple pronunciations are treated as word copies without priors, an extension of the usual Viterbi decoding is presented which integrates unigram priors in a weighted sum of acoustic probabilities. Several approximations are discussed leading to new decoding aspects. Experimental results are presented for US broadcast news recordings. It is shown that the use of unigram priors has a clear positive impact on both error rate and decoding cost while the sum over multiple pronunciation contributions brings another small improvement. An overall 4% reduction of the error rate is achieved on the HUB-4 evaluation sets of 97 and 98.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"38 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131686991","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: 20
Tied posteriors: an approach for effective introduction of context dependency in hybrid NN/HMM LVCSR 捆绑后验:在混合神经网络/HMM LVCSR中有效引入上下文依赖的方法
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.861800
J. Rottland, G. Rigoll
{"title":"Tied posteriors: an approach for effective introduction of context dependency in hybrid NN/HMM LVCSR","authors":"J. Rottland, G. Rigoll","doi":"10.1109/ICASSP.2000.861800","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.861800","url":null,"abstract":"This paper presents a method to improve the recognition rate of hybrid connectionist/HMM speech recognition systems. At the same time this approach allows the easy introduction of context dependent models in the hybrid framework. The approach is based on a standard hybrid connectionist/HMM recognizer, in which the neural nets are trained to estimate the a posteriori probabilities for all phones in each input frame. In the approach presented here, the probabilities of the neural nets are used to replace the codebook of a tied-mixture HMM system. Therefore the resulting system is called tied posterior. The advantages of this structure are that an arbitrary HMM-topology can be used, and that all context dependency and all clustering techniques used in tied-mixture systems can be applied to this hybrid speech recognition system. The approach has been evaluated on the Wall Street Journal (WSJ) database, with the result, that it outperforms the standard hybrid approach on this task.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"48 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131711363","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: 34
Integrating dynamic speech modalities into context decision trees 将动态语音模式集成到上下文决策树中
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.861810
C. Fügen, I. Rogina
{"title":"Integrating dynamic speech modalities into context decision trees","authors":"C. Fügen, I. Rogina","doi":"10.1109/ICASSP.2000.861810","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.861810","url":null,"abstract":"Context decision trees are widely used in the speech recognition community. Besides questions about phonetic classes of a phone's context, questions about their position within a word and questions about the gender of the current speaker have been used so far. In this paper we additionally incorporate questions about current modalities of the spoken utterance like the speaker's dialect, the speaking rate, the signal to noise ratio, the latter two of which may change while speaking one utterance. We present a framework that treats all these modalities in a uniform way. Experiments with the Janus speech recognizer have produced error rate reductions of up to 10% when compared to systems that do not use modality questions.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"46 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132600651","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: 22
Non-data-aided frequency offset and symbol timing estimation for binary CPM: performance bounds 二进制CPM的非数据辅助频率偏移和符号时序估计:性能界限
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.860972
J. Riba, G. Vázquez
{"title":"Non-data-aided frequency offset and symbol timing estimation for binary CPM: performance bounds","authors":"J. Riba, G. Vázquez","doi":"10.1109/ICASSP.2000.860972","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.860972","url":null,"abstract":"The use of (spectrally efficient) CPM modulations may lead to a serious performance degradation of the classical non-data-aided (NDA) frequency and timing estimators due to the presence of self noise. The actual performance of these estimators is usually much worse than that predicted by the classical modified Cramer-Rao bound. We apply some well known results in the field of signal processing to these two important problems of synchronization. In particular we propose and explain the meaning of the unconditional CRB in the synchronization task. Simulation results for MSK and GMSK, along with the performance of some classical and previously proposed synchronizers, show that the proposed bound (along with the MCRB) is useful for a better prediction of the ultimate performance of the NDA estimators.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"132635449","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: 7
Maximum likelihood detection for multicarrier systems employing non-orthogonal pulse shapes 采用非正交脉冲形状的多载波系统的最大似然检测
2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100) Pub Date : 2000-06-05 DOI: 10.1109/ICASSP.2000.860943
Wing-Kin Ma, P. Ching, K. M. Wong
{"title":"Maximum likelihood detection for multicarrier systems employing non-orthogonal pulse shapes","authors":"Wing-Kin Ma, P. Ching, K. M. Wong","doi":"10.1109/ICASSP.2000.860943","DOIUrl":"https://doi.org/10.1109/ICASSP.2000.860943","url":null,"abstract":"Investigation of detection schemes for non-orthogonal multicarrier modulation (MCM) is motivated by two reasons. Firstly, non-orthogonal MCM offers a higher degree of freedom in pulse-shaping design. Secondly, the problem of detecting orthogonal MCM under channel distortion can be viewed as a problem of detecting non-orthogonal MCM. In this work, the maximum likelihood detector (MLD) is considered for non-orthogonal multicarrier systems. In the absence of inter-block interference, it is shown that the MLD can be efficiently achieved by a Viterbi algorithm (VA). In contrast to using the VA for channel equalization, the proposed VA has its survivor metrics running in the\"\"frequency domain\". Incorporating this VA with an interference-canceling approach, we also develop a decision feedback MLD for the case of non-zero inter-block interference. Superior bit error performance of the MLDs is demonstrated by simulations.","PeriodicalId":164817,"journal":{"name":"2000 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.00CH37100)","volume":"06 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2000-06-05","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"129377296","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: 7
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