检索结果-内蒙古大学图书馆

Wideband speech coding advances in VMR-WB standard

IEEE TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING 2007年第4期15卷 1167-1179页

作者： Jelinek, Milan Salami, Redwan Univ Sherbrooke Dept Elect & Comp Engn Sherbrooke PQ J1K 2R1 Canada VoiceAge Corp Montreal PQ H3R 2H6 Canada

This paper presents novel techniques for source-controlled variable-rate wideband speech coding. These techniques have been used in the variable-rate multimode wideband (VMR-WB) speech codec recently selected by the Third-Generation Partnership Project 2 (3GPP2) for wideband (WB) speech telephony, streaming, and multimedia messaging services in the cdma2000 third-generation wireless system. The codec utilizes efficient coding modes optimized for different classes of speech signal including generic coding based on AMR-WB for transients and onsets, voiced coding optimized for stable voiced signals, unvoiced coding optimized for unvoiced segments, and comfort noise generation for inactive segments. Several innovations enable very good performance at average bit rates below 8 kb/s for active speech coding. The article presents an overview of the codec and describes in detail some of the codec novel features: Robust pitch tracking algorithm, coding-mode dependent prediction of linear prediction (LP) filter quantization, and novel frame erasure concealment techniques including supplementary information for reconstruction of lost onsets and improving decoder convergence. Selected results from the Selection and Characterization tests of the codec illustrate its performance.

关键词： linear predictive coding standardization variable-rate speech coding wideband speech coding

来源：评论

学校读者我要写书评

暂无评论

A speech-and-speaker identification system: Feature extraction, description, and classification of speech-signal image

引用

IEEE TRANSACTIONS ON INDUSTRIAL ELECTRONICS 2007年第2期54卷 887-897页

作者： Saeed, Khalid Nammous, Mohammad Kheir Bialystok Tech Univ PL-15351 Bialystok Poland

This paper discusses a speech-and-speaker (SAS) identification system based on spoken Arabic digit recognition. The speech signals of the Arabic digits from zero to ten are processed graphically (the signal is treated as an object image for further processing). The identifying and classifying methods are performed with Burg's estimation model and the algorithm of Toeplitz matrix minimal eigenvalues as the main tools for signal-image description and feature extraction. At the stage of classification, both conventional and neural-network-based methods are used. The success rate of the speaker-identifying system obtained in the presented experiments for individually uttered words is excellent and has reached about 98.8% in some cases. The miss rate of about 1.2% was almost only because of false acceptance (13 miss cases in 1100 tested voices). These results have promisingly led to the design of a security system-for SAS identification. The average overall,success rate was then 97.45% in recognizing one uttered word and identifying its speaker, and 92.5% in recognizing a three-digit password (three individual words), which is really a high success rate because, for compound cases, we should successfully test all the three uttered words consecutively in addition to and after identifying their speaker;hence, the probability of making an error is basically higher. The authors' major contribution to this task involves building a system to recognize both the uttered words and their speaker through an innovative graphical algorithm for feature extraction from the voice signal. This Toeplitz-based algorithm reduces the amount of computations from operations on an n x n matrix that contains n(2) different elements to a matrix (of Toeplitz form) that contains only n elements that are different from each other.

关键词： communication humatronics linear predictive coding processing and recognition speaker recognition speech analysis Toeplitz matrix (TM) eigenvalues understanding speech

来源：评论

学校读者我要写书评

暂无评论

Hierarchical particle filter for bearings-only tracking

引用

IEEE TRANSACTIONS ON AEROSPACE AND ELECTRONIC SYSTEMS 2007年第4期43卷 1567-1585页

作者： Brehard, T. Le Cadre, J. -P. IRISA CNRS F-35042 Rennes France

We address here the classical bearings-only tracking problem (BOT) for a single target, an issue that belongs to the general class of nonlinear filtering problems. Recently, algorithm-based sequential Monte-Carlo methods (particle filtering) have been proposed. However, Fearnhead has observed that in practice this algorithm diverges. This problem is investigated further here. We show that this phenomenon is due to the unobservability of the distance between the observer and the target. We propose a new algorithm named hierarchical particle filter which takes into account this aspect of the BOT. We demonstrate that this novel filter architecture largely overperforms the classical one. Moreover, these results are confirmed when considering highly maneuvering target scenarios. Finally, we propose a general architecture based on Monte-Carlo methods for filtering initialization, able to accommodate poor prior and complex constraints.

关键词： Particle filters Particle tracking Target tracking Nonlinear equations linear predictive coding Trajectory Time measurement Observability Filtering algorithms Sonar

来源：评论

学校读者我要写书评

暂无评论

Codebook-based Bayesian speech enhancement for nonstationary environments

引用

IEEE TRANSACTIONS ON AUDIO SPEECH AND LANGUAGE PROCESSING 2007年第2期15卷 441-452页

作者： Srinivasan, Sriram Samuelsson, Jonas Kleijn, W. Bastiaan Royal Inst Technol Dept Signals Sensors & Syst SE-10044 Stockholm Sweden

In this paper, we propose a Bayesian minimum mean squared error approach for the joint estimation of the short-term predictor parameters of speech and noise, from the noisy observation. We use trained codebooks of speech and noise linear predictive coefficients to model the a priori information required by the Bayesian scheme. In contrast to current Bayesian estimation approaches that consider the excitation variances as part of the a priori information, in the proposed method they are computed online for each short-time segment, based on the observation at hand. Consequently, the method performs well in nonstationary noise conditions. The resulting estimates of the speech and noise spectra can be used in a Wiener filter or any state-of-the-art speech enhancement system. We develop both memoryless (using information from the current frame alone) and memory-based (using information from the current and previous frames) estimators. Estimation of functions of the short-term predictor parameters is also addressed, in particular one that leads to the minimum mean squared error estimate of the clean speech signal. Experiments indicate that the scheme proposed in this paper performs significantly better than competing methods.

关键词： Bayesian codebooks linear predictive coding noise estimation speech enhancement speech processing Wiener filtering

来源：评论

学校读者我要写书评

暂无评论

Transmitter precoding with reduced-complexity soft detection for MIMO systems

引用

IEEE TRANSACTIONS ON WIRELESS COMMUNICATIONS 2007年第3期6卷 817-821页

作者： Li, Yong Moon, Jaekyun Univ Minnesota Dept Elect & Comp Engn Minneapolis MN 55455 USA

We present a precoded reduced-complexity soft detection (PRCSD) algorithm for multiple-input multiple-output (MIMO) systems. The linear operations at both transmit and receive sides based on complex Householder transform convert the MIMO channel to be multiple-diagonal, spatially partial-response-like, so that error propagation is alleviated when applying reduced-complexity soft detection at the receiver. The transform results in unitary precoding and feedforward matrices so that neither transmit power boost nor noise enhancement is present. Performance analysis based on pairwise error probability (PEP) shows that PRCSD achieves larger diversity advantage than existing precoding and multiple-beamforming (MB) schemes, which basically attempt to transmit signals through diagonal independent sub-channels and thus may suffer a diversity loss. PRCSD can achieve full diversity as maximum likelihood (ML) detection in some scenarios while reducing complexity significantly.

关键词： linear predictive coding

来源：评论

学校读者我要写书评

暂无评论

Statistical method for sparse coding of speech including a linear predictive model

引用

PHYSICA A-STATISTICAL MECHANICS AND ITS APPLICATIONS 2006年 367卷 231-251页

作者： Rufiner, Hugo L. Goddard, John Rocha, Luis F. Torres, Maria E. Univ Nac Entre Rios Fac Ingn RA-3100 Oro Verde Entre Rios Argentina Univ Autonoma Metropolitana Iztapalapa Dept Ing Elect Mexico City DF Mexico Univ Buenos Aires Fac Ingn RA-1053 Buenos Aires DF Argentina

Recently, different methods for obtaining sparse representations of a signal using dictionaries of waveforms have been studied. They are often motivated by the way the brain seems to process certain sensory signals. Algorithms have been developed using a specific criterion to choose the waveforms occurring in the representation. The waveforms are choosen from a fixed dictionary and some algorithms also construct them as a part of the method. In the case of speech signals, most approaches do not take into consideration the important temporal correlations that are exhibited. It is known that these correlations are well approximated by linear models. Incorporating this a priori knowledge of the signal can facilitate the search for a suitable representation solution and also can help with its interpretation. Lewicki proposed a method to solve the noisy and overcomplete independent component analysis problem. In the present paper we propose a modification of this statistical technique for obtaining a sparse representation using a generative parametric model. The representations obtained with the method proposed here and other techniques are applied to artificial data and real speech signals, and compared using different coding costs and sparsity measures. The results show that the proposed method achieves more efficient representations of these signals compared to the others. A qualitative analysis of these results is also presented, which suggests that the restriction imposed by the parametric model is helpful in discovering meaningful characteristics of the signals. (c) 2005 Elsevier B.V. All rights reserved.

关键词： speech analysis and representation sparse coding linear predictive coding basis pursuit matching pursuit independent component analysis

来源：评论

学校读者我要写书评

暂无评论

Robust speech watermarking algorithm

引用

ELECTRONICS LETTERS 2007年第24期43卷 1393-1395页

作者： Chen, N. Zhu, J. Shanghai Jiao Tong Univ Dept Elect Engn Shanghai 200240 Peoples R China

A zero-watermarking method based on vector quantisation (VQ) of linear prediction coefficients (LPCs) is proposed. The characteristic of natural speech that the VQ indices of LPCs among neighbouring frames tend to be very similar is utilised to generate the secret key Simulation results demonstrate excellent imperceptibility and robustness of our algorithm.

关键词： vector quantisation linear predictive coding linear prediction coefficient speech coding watermarking secret key generation robust speech watermarking algorithm zero-watermarking method

来源：评论

学校读者我要写书评

暂无评论

On-line signature writer identification

引用

AEJ - Alexandria Engineering Journal 2007年第4期46卷 509-518页

作者： Emam, Ashraf M. Information Technology Dept. Institute of Graduate Studies and Research University of Alexandria Alexandria Egypt

This paper presents a set of descriptors for On-line signature writer identification. These descriptors are intended to be used in e-business and e-government to detect signature forgery where it is hard to identify the writer across the Internet. Some descriptors represent global signature features, while the rest are dynamic signature features derived from the pen's linear speed. The forms of forged and the genuine signatures used in this work look identical. Cepstral descriptors showed a higher rejection rate of (98%) for all forged signatures, and 100% acceptance of any genuine signature. linear predictive descriptors derived from de-noised signature data delivered significant results with a rejection rate of (95%). © Faculty of Engineering Alexandria University.

关键词： Biometric identification Cepstral descriptors linear predictive coding On-line signature verification On-line signature writer identification

来源：评论

学校读者我要写书评

暂无评论

DSP restoration techniques for audio

DSP restoration techniques for audio

引用

IEEE International Conference on Image Processing (ICIP 2007)

作者： Moorer, James A. Adobe Syst Inc San Jose CA 95110 USA

ISBN: (纸本)9781424414369

The CD revolution of 1984 unleashed a massive campaign of rerelease of vintage analog recordings. To their collective horror, record labels found that many of the original master recordings had deteriorated. Often the only existing copy of a recording was a vinyl pressing. This rush to release beloved music in the new digital form spawned intense interest in digital methods of restoring these old recordings. A number of DSP techniques have been developed to give new life to these recordings. This paper is a survey of some of those methods.

关键词： music signal restoration signal processing linear predictive coding Wiener filtering

来源：评论

学校读者我要写书评

暂无评论

Wide-band perceptual audio coding based on frequency-domain linear prediction

Wide-band perceptual audio coding based on frequency-domain ...

引用

32nd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Motlicek, Petr Uallal, Vijay Hermansky, Hynek IDIAP Res Inst Martigny Switzerland Int Comp Sci Inst Berkeley CA USA Ecole Polytech Fed Lausanne Lausanne Switzerland

ISBN: (纸本)9781424407286

In this paper we propose an extension of the very low bit-rate speech coding technique, exploiting predictability of the temporal evolution of spectral envelopes, for wide-band audio coding applications. Temporal envelopes in critically band-sized sub-bands are estimated using frequency domain linear prediction applied on relatively long time segments. The sub-band residual signals, which play an important role in acquiring high quality reconstruction, are processed using a heterodyning-based signal analysis technique. For reconstruction, their optimal parameters are estimated using a closed-loop analysis-by-synthesis technique driven by a perceptual model emulating simultaneous masking properties of the human auditory system. We discuss the advantages of the approach and show some properties on challenging audio recordings. The proposed technique is capable of encoding high quality, variable rate audio signals on bit-rates below 1bit/sample.

关键词： audio signal processing data compression linear predictive coding Hilbert transform

来源：评论

学校读者我要写书评

暂无评论

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案：

请选择收藏分类：

通借通还

建议与咨询 留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案： 新增检索档案 确定 取消

请选择收藏分类： 新增自定义分类 确定 取消

通借通还

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

请选择保存的检索档案：

请选择收藏分类：