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作者： Xia, Bing-Yin Bao, Chang-Chun Liang, Yan Speech and Audio Signal Processing Lab. School of Electronic Information and Control Engineering Beijing University of Technology Beijing 100124 China

A fast convergence speech enhancement method is proposed in this paper. The noise estimation acceleration technique is applied to the conventional statistical model based algorithm to shorten the convergence time after the sudden change of noise intensity. First, the burst detection of power spectrum is performed on the noisy spectrum. Next, the loglikelihood ratio (LLR) based VAD is used in the period when the noise power is stationary, and the spectral entropy based VAD is implemented in the hang-over frames after the burst of noisy spectrum. Then a flag is set to control the update of noise estimation. Finally, an attenuated version of the noisy spectrum will be used directly as noise estimation if the update flag is set to one. The performance of the proposed method is evaluated under ITU-T G.160. In comparison with the conventional method, the convergence time is reduced evidently, while the abilities of noise reduction and SNR improvement are preserved, and the impact on the objective speech quality is constrained to a low level.

关键词： signal to noise ratio

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Characteristic waveform decomposition algorithm based on bi-orthogonal lifting wavelet transform

Characteristic waveform decomposition algorithm based on bi-...

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作者： Li, Ru-Wei Bao, Chang-Chun Speech and Audio Signal Processing Lab. School of Electronic Information and Control Engineering Beijing University of Technology Beijing 100124 China

A novel characteristic waveform (CW) decomposition method based on bi-orthogonal lifting wavelet transform (BLWT) is proposed for CWI speech codec in this paper. Firstly, the complicated CW alignment operations are not necessary by using this method, and as a result the computational complexity can be greatly reduced. Secondly, the additional delay of CW decomposition based on traditional wavelet transform is cancelled by using boundary treatment. Thirdly, by omitting filter operations and adopting the situ calculation technique, the memory consumption of the proposed algorithm will be much smaller than the traditional one. In addition, the multi-resolution analysis capability of wavelet transform is preserved, which will make the parameter quantization more flexible and the scalable coding much easier to be realized. The performance evaluation shows that the proposed decomposition algorithm could fulfill the requirements of high quality, multi-rate speech compression, and is suitable for the real-time communication systems. The results of MOS and A/B listening test show that the performance of 1.84kb/s CWI coder based on this CW decomposition is very close to that of 2.4kbit/s MELP coder.

关键词： Quality control

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Study of high-resolution range profile for monostatic MIMO radar based on SIAR

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Xi Tong Gong Cheng Yu Dian Zi Ji Shu/Systems Engineering and Electronics 2010年第4期32卷 734-739+763页

作者： Zhao, Guang-Hui Chen, Bai-Xiao Gao, Zhao-Zhao Intelligent Perception and Image Understanding Key Lab of Ministry of Education of China Xidian Univ. Xi'an 710071 China National Key Lab for Radar Signal Processing Xidian Univ. Xi'an 710071 China National Information Control Laboratory Chengdu 610036 China

Conventional pulse compression use a periodical echo of single receive antenna, which is modulated by a certain carrier-frequency, in other words, single spectrum is exploited. But for MIMO radar, as the multi-carrier-frequency signals are transmitted simultaneously, if the spectrum of the target echo after channel separation can be combined to form the whole band spectrum echo, the corresponding range resolution can improve several times as compared with the conventional method, and it will be more convenient for follow-up detection and tracking. Considering the difference between the frequency modulation band and the interval between the adjacent frequencies, the spectrum joint after channel separation will be overlapped or spaced. The methods of spectrum moving of each echo and the spectrum extrapolation with Root-MUSIC algorithm are proposed, by which high-resolution range profile of the target is obtained. Simulation results verify the validity of these methods.

关键词： Multiple signal classification

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A novel super-wideband embedded speech and audio codec based on ITU-T Recommendation G.729.1

A novel super-wideband embedded speech and audio codec based...

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Asia-Pacific signal and Information processing Association 2009 Annual Summit and Conference, APSIPA ASC 2009

作者： Jia, Mao-Shen Bao, Chang-Chun Liu, Xin Li, Rui Speech and Audio Signal Processing Lab School of Electronic Information and Control Engineering Beijing University of Technology Beijing 100124 China

This paper proposes a multi-layer super -widebandembedded speech and audio coding algorithm extending bit rates from 36 to 64 kb/s on the basis of ITU-T Recommendation G.729.1 with a multi-stage coding structure. This codec consists of three embedded stages: G.729.1 wideband coding operating in the range from 8 to 32 kb/s, modified Modulated LappedTransform (MLT) coding of the band (7-14 kHz) at 36, 40 & 48 kb/s and MDCT transform coding for wideband residual signal at 56 and 64 kb/s. In addition, some methods are proposed in transform coding according to perception significance. The objective and subjective listening tests show that this codec has good performance compared with reference codec.

关键词： Codes (symbols)

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Application of beamspace ML algorithm based on array interpolation in height measuring using VHF radar

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Dianbo Kexue Xuebao/Chinese Journal of Radio Science 2009年第4期24卷 660-666页

作者： Hu, Tie-Jun Yang, Xue-Ya Chen, Bai-Xiao National Lab. of Radar Signal Processing Xidian Univ. Xi'an Shaanxi 710071 China Air Traffic Control Central Workshop Beijing 100076 China

A new height measuring scheme for VHF radar is proposed to reduce the computational complexity due to multi-dimensional search in direction of arrival (DOA) estimation using the conventional maximum likelihood (ML) algorithm. By interpolating antennas between any two adjacent elements of the real array, the virtual array can solve the ambiguity in beamspace transformation caused by large element spacing of the original uniform linear array (ULA). Then unambiguous beamspace transformation is carried out to reduce the dimension of the original array. Finally, elevation and height estimates of targets can be obtained followed by beamsapce ML. Simulation and real data processing results show that the algorithm has much lower complexity, while nearly the same estimation precision compared with the conventional ML algorithm. Copyright ©2009 by Editorial Department of Chinese Journal of Radio Science.

关键词： Linear transformations

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A voice activity detection using cyclic statistics based on sinusoidal speech model

A voice activity detection using cyclic statistics based on ...

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2008 9th International Conference on signal processing, ICSP 2008

作者： Hui-Jing, Dou Chang-Chun, Bao Ru-Wei, Li Speech and Audio Signal Processing Lab. School of Electronic Information and Control Engineering Beijing University of Technology Beijing 100124 China

ISBN: (纸本)9781424421794

In this paper, by using the cyclostationary properties of speech signal, a voice activity detection (VAD) algorithm based on cyclic cumulant is proposed. The proposed scheme employs the thirdorder cyclic cumulant of the LPC residual of a speech signal. Analytical expressions for the third-order cyclic cumulant of the LPC residual of short-term speech are derived assuming a sinusoidal model. Matrix pencil method (MP) are adopted to estimate the frequencies of harmonic signal contained in LPC residual of speech signal, which are used as the cyclic frequencies of cyclic cumulant. Then the third-order cyclic cumulant is defined and used to construct the VAD detection variation. The test results show that the proposed algorithm gives better results than G.729B VAD. © 2008 IEEE.

关键词： Speech recognition

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An Improvement of Frame Erasure Concealment for G.722.1 Coding Algorithm

An Improvement of Frame Erasure Concealment for G.722.1 Codi...

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9th International Conference on signal processing(第九届国际信号处理学术会议)(ICSP'08)

作者： Ji-hua Niu Chang-chun Bao Speech and Audio Signal Processing Lab School of Electronic Information and Control EngineeringBeijing University of TechnologyBeijing 100124China

This paper proposes an efficient method for frame erasure concealment in G.722.1 coding algorithm,which can mitigate the adverse impact of frame erasure on the reconstruction *** lost frame is reconstructed in the modulated lapped transform (MLT) domain in terms of magnitude and sign of the *** method of interpolation is employed for the magnitude *** sign information is recovered by extra information from *** test results indicate that improved quality is produced by the proposed algorithm.

关键词： signal processing algorithms Quantization Speech coding Decoding Bit rate Huffman coding Discrete cosine transforms Speech processing Interpolation Testing

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A Novel Embedded Speech and Audio Codec Based on ITU-T Recommendation G.722.1

A Novel Embedded Speech and Audio Codec Based on ITU-T Recom...

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9th International Conference on signal processing(第九届国际信号处理学术会议)(ICSP'08)

作者： Mao-shen JIA Chang-chun BAO Rui LI Speech and Audio Signal Processing Lab School of Electronic Information and Control EngineeringBeijing University of TechnologyBeijing 100124China

This paper describes an embedded speech and audio codec which is based on ITU-T Recommendation G.722.1;it can process 7 kHz bandwidth speech and audio signal at scalable bit *** on the G.722.1 of ITU-T,this algorithm adds two modules: the energy ordering of sub-band and the processing of bit-stream ***,it does some modification on the categorization and noise-fill *** makes sure that the codec could produce embedded bit-stream,so this codec had more robustness in the *** test results by ITUT PESQ show that this codec has good performance as G.722.1 at the same bit-rates.

关键词： Speech codecs Speech processing Decoding signal processing Bandwidth Bit rate Testing Mobile communication control engineering signal processing algorithms

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1kb/s Waveform Interpolation Speech Coding Based on Non-negative Matrix Factorization

1kb/s Waveform Interpolation Speech Coding Based on Non-nega...

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9th International Conference on signal processing(第九届国际信号处理学术会议)(ICSP'08)

作者： Er-juan XUE Chang-chun BAO Speech and Audio Signal Processing Lab School of Electronic Information and Control EngineeringBeijing University of TechnologyBeijing 100124China

A new 1kb/s waveform interpolation (WI) speech coding algorithm based on non-negative matrix factorization (NMF) is proposed and implemented in this ***-frame parameter joint vector or matrix quantization,parameter prediction and discrete cosine transform (DCT) are used to reduce the bit rate and to improve the quality of *** results of informal subjective listening test shows that the intelligibility and articulation of the algorithm are close to that of 2kb/s NMF-based WI coder (called “NMF-WI” as convenience).

关键词： Interpolation Speech coding Bit rate signal processing algorithms Quantization Finite impulse response filter Discrete cosine transforms Testing Humans Low pass filters

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Pitch Detection Method for Noisy Speech signals Based on Pre-Filter and Weighted Wavelet coefficients

Pitch Detection Method for Noisy Speech Signals Based on Pre...

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9th International Conference on signal processing(第九届国际信号处理学术会议)(ICSP'08)

作者： Ru-wei LI Chang-chun BAO Hui-jing DOU Speech and Audio Signal Processing Lab School of Electronic Information and Control EngineeringBeijing University of TechnologyBeijing 100124China

Most of the current pitch detection algorithms can not work well under the high noise *** this reason,a pitch detection algorithm for noisy speech signal based on pre-filtering and weighted wavelet coefficients is ***,the noisy speech signals are ***,the speech pre-filtered is decomposed by the quadratic spline ***,the wavelet coefficients of three consecutive scales are weighted to emphasize the sharp change ***,three candidate pitch periods are extracted from the weighted ***,the pitch period is calculated by autocorrelation *** show that this algorithm can increase the performance of pitch detection in noisy environment and decreases computational complexity compared with DWT-NCCF method.

关键词： Wavelet coefficients Working environment noise Wavelet transforms signal processing algorithms Autocorrelation Speech analysis Filters Speech enhancement Detection algorithms Spline

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