IEEE Transactions on Audio Speech and Language Processing最新文献_第8页

Nonlinear Acoustic Echo Cancellation Based on a Sliding-Window Leaky Kernel Affine Projection Algorithm 基于滑动窗漏核仿射投影算法的非线性声回波消除

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-09-01 DOI: 10.1109/TASL.2013.2260742

Jose Manuel Gil-Cacho, M. Signoretto, T. Waterschoot, M. Moonen, S. H. Jensen

{"title":"Nonlinear Acoustic Echo Cancellation Based on a Sliding-Window Leaky Kernel Affine Projection Algorithm","authors":"Jose Manuel Gil-Cacho, M. Signoretto, T. Waterschoot, M. Moonen, S. H. Jensen","doi":"10.1109/TASL.2013.2260742","DOIUrl":"https://doi.org/10.1109/TASL.2013.2260742","url":null,"abstract":"Acoustic echo cancellation (AEC) is used in speech communication systems where the existence of echoes degrades the speech intelligibility. Standard approaches to AEC rely on the assumption that the echo path to be identified can be modeled by a linear filter. However, some elements introduce nonlinear distortion and must be modeled as nonlinear systems. Several nonlinear models have been used with more or less success. The kernel affine projection algorithm (KAPA) has been successfully applied to many areas in signal processing but not yet to nonlinear AEC (NLAEC). The contribution of this paper is three-fold: (1) to apply KAPA to the NLAEC problem, (2) to develop a sliding-window leaky KAPA (SWL-KAPA) that is well suited for NLAEC applications, and (3) to propose a kernel function, consisting of a weighted sum of a linear and a Gaussian kernel. In our experiment set-up, the proposed SWL-KAPA for NLAEC consistently outperforms the linear APA, resulting in up to 12 dB of improvement in ERLE at a computational cost that is only 4.6 times higher. Moreover, it is shown that the SWL-KAPA outperforms, by 4-6 dB, a Volterra-based NLAEC, which itself has a much higher 413 times computational cost than the linear APA.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2260742","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62889541","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: 58

Sound Source Distance Estimation in Rooms based on Statistical Properties of Binaural Signals 基于双耳信号统计特性的室内声源距离估计

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-08-01 DOI: 10.1109/TASL.2013.2260155

Eleftheria Georganti, T. May, S. Par, J. Mourjopoulos

引用次数: 36

Bayesian Feature Enhancement for Reverberation and Noise Robust Speech Recognition 基于贝叶斯特征增强的混响和噪声鲁棒语音识别

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-08-01 DOI: 10.1109/TASL.2013.2258013

Volker Leutnant, A. Krueger, Reinhold Häb-Umbach

{"title":"Bayesian Feature Enhancement for Reverberation and Noise Robust Speech Recognition","authors":"Volker Leutnant, A. Krueger, Reinhold Häb-Umbach","doi":"10.1109/TASL.2013.2258013","DOIUrl":"https://doi.org/10.1109/TASL.2013.2258013","url":null,"abstract":"In this contribution we extend a previously proposed Bayesian approach for the enhancement of reverberant logarithmic mel power spectral coefficients for robust automatic speech recognition to the additional compensation of background noise. A recently proposed observation model is employed whose time-variant observation error statistics are obtained as a side product of the inference of the a posteriori probability density function of the clean speech feature vectors. Further a reduction of the computational effort and the memory requirements are achieved by using a recursive formulation of the observation model. The performance of the proposed algorithms is first experimentally studied on a connected digits recognition task with artificially created noisy reverberant data. It is shown that the use of the time-variant observation error model leads to a significant error rate reduction at low signal-to-noise ratios compared to a time-invariant model. Further experiments were conducted on a 5000 word task recorded in a reverberant and noisy environment. A significant word error rate reduction was obtained demonstrating the effectiveness of the approach on real-world data.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-08-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2258013","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62888953","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

A General Compression Approach to Multi-Channel Three-Dimensional Audio 多声道三维音频的通用压缩方法

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-08-01 DOI: 10.1109/TASL.2013.2260156

B. Cheng, C. Ritz, I. Burnett, Xiguang Zheng

引用次数: 16

Sparse Classifier Fusion for Speaker Verification 基于稀疏分类器融合的说话人验证

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-08-01 DOI: 10.1109/TASL.2013.2256895

Ville Hautamäki, T. Kinnunen, Filip Sedlak, Kong-Aik Lee, B. Ma, Haizhou Li

引用次数: 48

Syntax-Based Translation With Bilingually Lexicalized Synchronous Tree Substitution Grammars 基于句法的双语词汇化同步树替换语法翻译

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-08-01 DOI: 10.1109/TASL.2013.2255283

Jiajun Zhang, Feifei Zhai, Chengqing Zong

{"title":"Syntax-Based Translation With Bilingually Lexicalized Synchronous Tree Substitution Grammars","authors":"Jiajun Zhang, Feifei Zhai, Chengqing Zong","doi":"10.1109/TASL.2013.2255283","DOIUrl":"https://doi.org/10.1109/TASL.2013.2255283","url":null,"abstract":"Syntax-based models can significantly improve the translation performance due to their grammatical modeling on one or both language side(s). However, the translation rules such as the non-lexical rule “ VP→(x0x1,VP:x1PP:x0)” in string-to-tree models do not consider any lexicalized information on the source or target side. The rule is so generalized that any subtree rooted at VP can substitute for the nonterminal VP:x1. Because rules containing nonterminals are frequently used when generating the target-side tree structures, there is a risk that rules of this type will potentially be severely misused in decoding due to a lack of lexicalization guidance. In this article, inspired by lexicalized PCFG, which is widely used in monolingual parsing, we propose to upgrade the STSG (synchronous tree substitution grammars)-based syntax translation model with bilingually lexicalized STSG. Using the string-to-tree translation model as a case study, we present generative and discriminative models to integrate lexicalized STSG into the translation model. Both small- and large-scale experiments on Chinese-to-English translation demonstrate that the proposed lexicalized STSG can provide superior rule selection in decoding and substantially improve the translation quality.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-08-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2255283","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62888470","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 Class of Algorithms for Time-Frequency Multiplier Estimation 一类时频乘子估计算法

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-08-01 DOI: 10.1109/TASL.2013.2255274

Anaïk Olivero, B. Torrésani, R. Kronland-Martinet

引用次数: 25

The Spectral Nature of Maximum Likelihood Noise Compensated Linear Prediction 最大似然噪声补偿线性预测的频谱性质

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-08-01 DOI: 10.1109/TASL.2013.2255277

L. Weruaga, L. Dimitrov

引用次数: 3

Broadband DOA Estimation Using Sensor Arrays on Complex-Shaped Rigid Bodies 基于复杂形状刚体传感器阵列的宽带DOA估计

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-08-01 DOI: 10.1109/TASL.2013.2255282

Dumidu S. Talagala, Wen Zhang, T. Abhayapala

{"title":"Broadband DOA Estimation Using Sensor Arrays on Complex-Shaped Rigid Bodies","authors":"Dumidu S. Talagala, Wen Zhang, T. Abhayapala","doi":"10.1109/TASL.2013.2255282","DOIUrl":"https://doi.org/10.1109/TASL.2013.2255282","url":null,"abstract":"Sensor arrays mounted on complex-shaped rigid bodies are a common feature in many practical broadband direction of arrival (DOA) estimation applications. The scattering and reflections caused by these rigid bodies introduce complexity and diversity in the frequency domain of the channel transfer function, which presents several challenges to existing broadband DOA estimators. This paper presents a novel high resolution broadband DOA estimation technique based on signal subspace decomposition. We describe how broadband signals can be decomposed into narrow subband components, and combined such that the frequency domain diversity is retained. The DOA estimation performance is compared with existing techniques using a uniform circular array and a sensor array on a hypothetical rigid body. An improvement in closely spaced source resolution of up to 6 dB is observed for the sensor array on the hypothetical rigid body, in comparison to the uniform circular array. The results suggest that frequency domain diversity, introduced by complex-shaped rigid bodies, can provide higher resolution and clearer separation of closely spaced broadband sound sources.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-08-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2255282","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62888330","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: 10

A Free-Source Method (FrSM) for Calibrating a Large-Aperture Microphone Array 一种自由源法校准大口径传声器阵列

IEEE Transactions on Audio Speech and Language Processing Pub Date : 2013-08-01 DOI: 10.1109/TASL.2013.2256896

Sarthak Khanal, H. Silverman, Rahul R. Shakya

{"title":"A Free-Source Method (FrSM) for Calibrating a Large-Aperture Microphone Array","authors":"Sarthak Khanal, H. Silverman, Rahul R. Shakya","doi":"10.1109/TASL.2013.2256896","DOIUrl":"https://doi.org/10.1109/TASL.2013.2256896","url":null,"abstract":"Large-aperture microphone arrays can be used to capture and enhance speech from individual talkers in noisy, multi-talker, and reverberant environments. However, they must be calibrated, often more than once, to obtain accurate 3-dimensional coordinates for all microphones. Direct-measurement techniques, such as using a measuring tape or a laser-based tool are cumbersome and time-consuming. Some previous methods that used acoustic signals for array calibration required bulky hardware and/or fixed, known source locations. Others, which allowed more flexible source placement, often have issues with real data, have reported results for 2D only, or work only for small arrays. This paper describes a complete and robust method for automatic calibration using acoustic signals which is simple, repeatable, accurate, and has been shown to work for a real system. The method requires only a single transducer (speaker) with a microphone attached above its center. The unit is freely moved around the focal volume of the microphone array generating a single long recording from all the microphones. After that, the system is completely automatic. We describe the free source method (FrSM), validate its effectiveness and present accuracy results against measured ground truth. The performance of FrSM is compared to that from several other methods for a real 128-microphone array.","PeriodicalId":55014,"journal":{"name":"IEEE Transactions on Audio Speech and Language Processing","volume":null,"pages":null},"PeriodicalIF":0.0,"publicationDate":"2013-08-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"https://sci-hub-pdf.com/10.1109/TASL.2013.2256896","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"62888985","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: 15