检索结果-内蒙古大学图书馆

European Signal processing Conference (EUSIPCO)

作者： Matthias Dörbecker Stefan Ernst Institute of Communication Systems and Data Processing Aachen University of Technology Templergraben 55 52056 Aachen Germany

In this contribution a novel structure for the enhancement of speech signals disturbed by acoustic noise is presented which is based on Spectral Subtraction. The Spectral Subtraction technique is combined with a novel estimator for the noise power spectrum which takes advantage of the employment of a second microphone. Due to the extension to a two-microphone system the Spectral Subtraction can be used to reduce realistic, non-stationary noise sources. Additionally, the performance of the system is further improved by the application of a post filter adapted according to Wiener filter techniques. As a result, the proposed speech enhancement system provides a significant suppression of noise in realistic situations as well as a reduction of room reverberation.

关键词： Noise Speech Microphones Speech enhancement Reverberation Spectral analysis Noise reduction

来源：评论

学校读者我要写书评

暂无评论

Very low complexity CELP speech coding with adaptive phase codebooks

Very low complexity CELP speech coding with adaptive phase c...

引用

1995 IEEE Workshop on Speech Coding for Telecommunications, SCFT 1995

作者： Gerlach, Christian G. Institute of Communication Systems and Data Processing Aachen University of Technology Aachen52056 Germany

In analysis-by-synthesis speech coders the computational complexity of the search for an optimum innovation is still high though transformations were proposed to decrease the complexity e. g. [1]. This limits practical codebook sizes and vector dimensions (block lengths). In [2] a transform coding interpretation of analysis-by-synthesis was stated. Further it has been shown that unity magnitude codebooks in the frequency domain with the correct phase information allow near original speech quality. So unity magnitude codebooks with structured phases were proposed, that permit a computational complexity which increases only proportional to the bit rate and not to the code-book size. But in this simple approach slight perceptual degradations remained. In continuation to these ideas, an adaptive product codebook is proposed in this contribution, which allows more natural phase contours to eliminate the degradations, while keeping the complexity extreme low. Essential for the new concept, is the dynamic bit allocation to frequency intervals with variable bandwidth. Considerable improvements of both objective and subjective quality were achieved. © 1995 Proceedings - 1995 IEEE Workshop on Speech Coding for Telecommunications: Speech Coding for Interoperable Global communications, SCFT 1995. All rights reserved.

关键词： Computational complexity

来源：评论

学校读者我要写书评

暂无评论

Noise estimation techniques for robust speech recognition

Noise estimation techniques for robust speech recognition

引用

International Conference on Acoustics, Speech, and Signal processing (ICASSP)

作者： H.G. Hirsch C. Ehrlicher Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

Two new techniques are presented to estimate the noise spectra or the noise characteristics for noisy speech signals. No explicit speech pause detection is required. Past noisy segments of just about 400 ms duration are needed for the estimation. Thus the algorithm is able to quickly adapt to slowly varying noise levels or slowly changing noise spectra. This techniques can be combined with a nonlinear spectral subtraction scheme. The ability can be shown to enhance noisy speech and to improve the performance of speech recognition systems. Another application is the realization of a robust voice activity detection.

关键词： Noise robustness Speech recognition Speech enhancement Noise reduction Noise level Background noise Statistics Signal to noise ratio Recursive estimation data processing

来源：评论

学校读者我要写书评

暂无评论

Very Low Complexity CELP Speech Coding with Adaptive Phase Codebooks

Very Low Complexity CELP Speech Coding with Adaptive Phase C...

引用

IEEE Workshop on Speech Coding For Telecommunications

作者： C.G. Gerlach Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

来源：评论

学校读者我要写书评

暂无评论

Uplink performance of a new macro-diversity cellular mobile radio architecture

Uplink performance of a new macro-diversity cellular mobile ...

引用

IEEE International Symposium on Personal, Indoor and Mobile Radio communications (PIMRC)

作者： W. Papen Institute of Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany

来源：评论

学校读者我要写书评

暂无评论

Coupled adaptive filters for acoustic echo control and noise reduction

Coupled adaptive filters for acoustic echo control and noise...

引用

International Conference on Acoustics, Speech, and Signal processing (ICASSP)

作者： R. Martin J. Altenhoner Institute of Communication Systems and Data Processing (IND) RWTH Aachen University of Technology Aachen Germany

Presents new adaptive algorithms for acoustic echo control and noise reduction which employ one, two, or possibly more microphone signals. The new algorithms accommodate high echo attenuation and lead to implementations with reduced complexity. These algorithms combine a conventional FIR echo canceller with a second NLMS-adapted FIR filter which attenuates residual echoes. The paper presents a one-microphone system with improved echo attenuation and a two-microphone system which attenuates acoustic echoes as well as ambient noise and near end speech reverberation. The algorithms can be interpreted as a frequency selective generalization of the well known voice controlled switch. The paper explains the algorithms and presents experimental results in real acoustic environments.

关键词： Adaptive filters Acoustic noise Noise cancellation Attenuation Finite impulse response filter Switches Adaptive algorithm Noise reduction Microphones Echo cancellers

来源：评论

学校读者我要写书评

暂无评论

Simulation of time variant room impulse responses

Simulation of time variant room impulse responses

引用

International Conference on Acoustics, Speech, and Signal processing (ICASSP)

作者： C. Antweiler H.-G. Symanzik Institute for Communication Systems and Data Processing RWTH Aachen University of Technology Aachen Germany Rheinisch-Westfalische Technische Hochschule Aachen Aachen Nordrhein-Westfalen DE Inst. for Commun. Syst. & Data Process. Tech. Hochschule Aachen Germany

For many applications such as acoustic echo compensation, adaptive noise reduction or acoustic feedback control it is of great interest to simulate reproducibly a real, time variant room. One approach to describe the transient behavior of a room is the generation of a physical room model. To identify the variation of a room impulse response with time an alternative concept is presented, which uses the normalized least mean square (NLMS) algorithm excited by perfect sequences. The proposed general concept is capable to efficiently simulate the fluctuations of the room impulse response. The time variant simulation can be performed without the necessity to store large amounts of data by storing only the reaction of the unknown system instead of all sets of filter coefficients. The practical aspects of the new concept are pointed out for an acoustic echo control application.

关键词： White noise Filters Signal processing Autocorrelation Delay Computational complexity Proposals Fluctuations Iterative algorithms

来源：评论

学校读者我要写书评

暂无评论

PERFECT SEQUENCE EXCITATION OF THE NLMS ALGORITHM AND ITS APPLICATION TO ACOUSTIC ECHO CONTROL

引用

ANNALES DES TELEcommunicationS-ANNALS OF TELEcommunicationS 1994年第7-8期49卷 386-397页

作者： ANTWEILER, C DORBECKER, M 1. Institute for communication systems and data processing (Ind) Aachen University of Technology Templergraben 55 D-52056 Aachen Allemagne

Adaptive echo cancellers are currently being studied for applications such as audio teleconference systems or hands-free telephone sets with high speech quality. The purpose of the echo control is to eliminate the acoustic feedback from the loudspeaker to the microphone. One problem of echo cancellers using e.g. the normalized least mean square algorithm (NLMS) for the adaptation of the coefficients is that the convergence properties degrade with colored signal input such as speech signals [8, 9, 17]. One approach to accelerate the convergence speed is to introduce linear prediction filters in order to decorrelate the speech signal [1, 2, 12, 15, 20]. This paper presents a new approach, named the excited LMS algorithm or ELMs algorithm, which prewhitens the input signal applying perfect sequences. Coincidently, the proposed algorithm can be interpreted as a combination of the conventional NLMS algorithm and a system identification approach using m-sequences or related sequences.

关键词： ECHO CANCELER ACOUSTIC SIGNAL TELEPHONE TELECONFERENCE ADAPTIVE ALGORITHM WHITENING PSEUDORANDOM SEQUENCE SYSTEM IDENTIFICATION LEAST SQUARE METHOD

来源：评论

学校读者我要写书评

暂无评论

COMBINED ACOUSTIC ECHO CANCELLATION, DEREVERBERATION AND NOISE-REDUCTION - A 2 MICROPHONE APPROACH

引用

ANNALES DES TELEcommunicationS-ANNALS OF TELEcommunicationS 1994年第7-8期49卷 429-438页

作者： MARTIN, R VARY, P 1. Institute for Communication Systems and Data Processing (IND) Aachen University of Technology Templergraben 55 D-52056 Aachen Germany

This paper presents new algorithms for acoustic echo cancellation and noise reduction which use two (or possibly more) microphone signals. In contrast to the single microphone method the multimicrophone approach can exploit the spatial coherence properties of sound fields which arise from noise and reverberated speech. Besides the standard FIR echo canceller the proposed algorithms comprise an adaptive filter to eliminate non coherent signal components. The combined system achieves better Erle than the FIR echo canceller alone, attenuates ambient noise, dereverberates near end speech, and possibly leads to implementations with reduced complexity. The paper analyzes the acoustical properties of typical environments, presents the algorithms and experimental results.

关键词： TELEPHONE ECHO ACOUSTIC SIGNAL ECHO CANCELER ACOUSTIC NOISE AMBIENT NOISE NOISE REDUCTION SPATIAL COHERENCE DIFFUSE REFLECTION

来源：评论

学校读者我要写书评

暂无评论

Celp speech coding with almost no codebook search

Celp speech coding with almost no codebook search

引用

Proceedings of the 1994 IEEE International Conference on Acoustics, Speech and Signal processing. Part 2 (of 6)

作者： Gerlach, Christian G. Institute for Communication Systems and Data Processing Aachen University of Technology Templergraben 55 Aachen52056 Germany

In analysis-by-synthesis speech coders the computational complexity of the searchfor an optimum innovation is dill high although transformations were proposed to decrease the complex-iv. This limits practical codebook sizes and vector dimensions (blocklengths). In this contribution two new structured freqruency domain codebooks are proposed, Thefirst one is a pulse shaped codebook with a reversed search order for the gain and the shape, the second one is a unity magnitude codebook with structured phase. The corresponding algorithms which are based on new insights, result in a drastically reduced search in the transformed domain. The computational complexity increases only proportional to the bit rate and not to the codebooksize. © 1994 IEEE

关键词： Computational complexity

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：