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Investigations on the role of analysis window shape parameter in speech enhancement using hybrid method
1.
978-1-5386-4304-4/18/$31.00 ©2018 IEEE. Investigations
on the Role of Analysis Window Shape Parameter in Speech Enhancement using Hybrid Method *S.China Venkateswarlu,Research Scholar,ECE- JNTUK ,Professor-ECE,MRIET-JNTUH Secunderabad, Telangana State, India sonagiricv@gmail.com ****N.Karthik, Research Scholar from VelTech, Chennai, Asst.Professor-ECE, NREC-Sec-Bad, karthik011190@gmail.com **A.Subba Rami Reddy,Principal & Professor-ECE, GKCE, Sullurpet, JNTUA, INDIA skitprincipal@gmail.com *** K.Satya Prasad, Professor-ECE, JNTUK-Kakinada, India. Prasad_kodati@yahoo.co.in Abstract— This paper investigates the effect of Window shape on the improvement of Speech quality by reducing the noise using Analysis Window with optimum shape. In Speech Enhancement process, signal corrupted by noise is segmented into frames and each segment is Windowed using Analysis Window with variation in the shape parameter. The Windowed Speech segments are applied to the Hybrid Methods and the Enhanced Speech signal is reconstructed in its time domain. The focus is to study the effect of analysis Window shape on the Speech Enhancement process. For different shapes of the analysis Window, It is observed that the Window shape plays an important role on the enhancement process. The quality of the Enhanced Speech was measured using six objective measures. The results are compared with the measures of analysis window and an optimum shape constant for the analysis Window is proposed for better speech quality. This paper investigates the improvement of Speech quality in terms of six objective quality measures using Discrete Wavelet Transform and proposes two Hybrid methods Keywords— discrete wavelet transform, hamming window, Gaussian ,Kaiser, Dolph Chebyshev and Polynomial Windows, , Speech Enhancement, Shape parameter alpha, objective quality measures, Thresholding. I. INTRODUCTION Speech is the most primary human communication. For that reason, it exists a big trend to increase and improve telecommunications [1]. Now-a-days, all the people use the communication devices such as telephones, mobiles, internet etc., as a primary goal and the customers demand a high coverage and quality. But a speech signal is often degraded by additive background noise. Listening task is very difficult at the end user, in such noisy environment. Therefore, it is necessary to develop speech enhancement algorithms. Speech enhancement is the most important field of speech processing. Speech enhancement refers to methods aiming at recovering speech signal from a noisy observation. During the last decades, many algorithms and various approaches have been proposed to the problem such as spectral subtraction [2], wavelet based methods [3], hidden Markovmodelling [4] and signal subspace methods [5] to improve the perceptual quality of the speech signals from the corrupted input signal. The wavelet based de-noising algorithm is one of the ways for speech enhancement. Telugu speech sentences are applied to this algorithm for enhancement. Wavelets have been found to be a powerful tool for removing noise. The paper is organized as follows: Chapter I –Introduction, Chapter II explains the fixed and Analysis windows. Chapter III explains objective measures; Chapter IV describes the Algorithms, Chapter V presents the Implementation and Simulation Results and chapter VI describes the Conclusion. A. Fixed and analysis windows In literature many windows have been proposed [7-10]. They are known as suboptimal solutions, and the best window is depending on the applications. Windows can be categorized as fixed or adjustable [11]. The Dolph-Chebyshev Window is constructed in the frequency domain by taking uniformly spaced samples of the window with Discrete Fourier Transform (DFT) and =Length of the side lobe attenuation. Discrete time domain Dolph-Chebyshev Window function can be written as 1N 0k N j2π2π W(k)ew(n) ------ (1.1) B. Kaiser-Bessel Windows The Kaiser-Bessel window was formulated to be a simple-to- compute approximation to the zero th order discrete prolate
2.
978-1-5386-4304-4/18/$31.00 ©2018 IEEE. spheroidal
sequences in the main lobe [8]. In discrete time domain, Kaiser Window is defined by [9] (1.2) Where “α” is the adjustable shape parameter, and I0(x) is the modified Bessel function of the first kind of order zero and it is described by the power series expansion as (1.3) where I0 (・) is the zero th-order modified Bessel function of the first kind. As ” increases, the main lobe width widens and the side lobe attenuation increases. For ” = 0, the Kaiser- Bessel window is a rectangular window. For ” = 5.3, the Kaiser-Bessel window is close to a Hamming window. The two parameters useful to obtain the desired amplitude response pattern of the above defined windows are the length of the sequence N and a shape parameter. In this study the number of FFT points considered in the Windowed speech frame is fixed to 160 or 240 hence the window length N is fixed. Therefore the shape parameter” can only be varied to achieve the desired pattern of the magnitude response of the Window used. As the shape parameter “” increases, the side lobe level of the magnitude response decreases at the cost of main lobe width. II. Objective Measures i) Signal-To-Noise Ratio (SNR): The Signal-to-Noise Ratio (SNR) is the ratio of signal energy to noise energy and is expressed in decibels dB given by [2-4] as (2.1) ii) Segmental Signal-to-noise Ratio: The Seg-SNR is the frame-based SNR and is estimated as It is defined [2-4] as = (2.2) iii) Weighted Spectral Slope Distance: WSS distance measure computes the weighted difference between the spectral slopes in each frequency band. The spectral slope is obtained as the difference between adjacent spectral magnitudes in decibels. The WSS measure is defined and evaluated [18] as (2.3) iv) Log Likelihood Ratio: The LLR measure is defined [19] as LLR= (2.4) v) Cepstrum distance: The Cepstrum distance [19] provides an estimate of the log spectral distance between two spectra. It is defined as + c(k) for 1≤m≤p (2.5) vi) Frequency weighted segmental SNR: It is computed [4, 20] using the following equation (2.6) III. Algorithms A. Weiner Filter Method Noise corrupted speech signals are processed by the Weiner Filter speech enhancement algorithm based on a priori SNR estimation [21]. Let s(t) and b(t) denote the speech and the additive noise processes, respectively. The observed signal x(t) is given by x(t)=s(t)+b(t). (3.1) (3.2) (3.3) GW (k, t) = (3.4) B. Spectral Subtraction
3.
978-1-5386-4304-4/18/$31.00 ©2018 IEEE. The
noise corrupted speech is processed by the Spectral Subtraction method to get processed or enhanced speech. Spectral Subtraction [21-24] is a popular frequency domain method to reduce the effect of additive uncorrelated noise in a signal. y(n) = s(n) + v(n) (3.5) = (3.6) Here Y(w, k) is the DFT of y(n, k) given by = (3.7) In this paper, all the measures were evaluated with a = 1.7. (3.8) Where 0 ≤ µ ≤ 1 is exponential averaging constant. In this work “µ” is selected as 0.85. C. Speech Enhancement Using Wavelet Transform The wavelet based speech signal enhancement technique was proposed by Donoho and Johnstone [8]. This method is based on thresholding the wavelet coefficients of noisy speech signal. The fundamental idea behind wavelets is to analyse according to scale. D. Hybrid Thresholding In this method the authors are proposed two new thresholding schemes by combining with modified improved thresholding scheme with soft thresholding and modified improved thresholding with improved thresholding and are defined in EQ.11 and EQ.12 given below. (4.13) (3.9) IV. Implementation and Simulation Results A. Speech Materials The aim of this section is to acquire the speech samples. The experimental part consists of recording each of the well- known Telugu Speech proverbs at a normal speaking rate three times in a quiet room by three male and three female native Telugu speakers (age around 23 years) at a sampling rate of 48 kHz and 16 bit value. These digitized speech sounds are then down sampled to 8 kHz and then normalized for the purpose of analysis. The Gaussian white noise is added to the speech signal in four particular SNRs: (15 dB, 10 dB, 5 dB, 0 dB). DWT is used to obtain the Enhanced Speech Signal from noisy Speech Signal. The so produced pairs of reference and Enhanced Signals are used for evaluating the objective measures of speech quality. Figure 4: Block Diagram of Wavelet De- noising of Speech Signal Noisy speec h Enhance Speech DWT Level Dependent Thresholding Framing &Windowing Windowing IDWT Overlap &Add
4.
978-1-5386-4304-4/18/$31.00 ©2018 IEEE. V. VI. VII. VIII. IX. X. XI. XII.
Phonetically balanced clean speech signals and noise Figure 4(a): LLR Variation of enhanced speech as a function of β Figure 4(b): WSS Variation of enhanced speech as a function of β Figure 4(c): CEP Variation of enhanced speech as a function of β Figure 4(d): SEG-SNR Variation of enhanced speech as a function of β Figure 4(e): FWSEG Variation of enhanced speech as a function of β Figure 4(f): SNR Variation of enhanced speech as a function of β
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978-1-5386-4304-4/18/$31.00 ©2018 IEEE. Corrupted
signals at different SNR levels have been taken from a speech corpus called NOIZEUS. Noise corrupted speech signal is segmented into frames containing 160 samples of 20ms length (at 8 KHz Sampling rate).160- point Discrete Fourier Transform (DFT) of each segment is obtained after applying the Kaiser window with variable shape parameter “α”. Spectral Subtraction Algorithm is applied to the spectral components of each segment using mathematical equations. The signal is reconstructed in its time domain with the help of IDFT and overlap adds method (with 40% overlap between frames). The signal thus obtained is the enhanced signal. The performance of the enhanced signal is analyzed by using six objective measures for speech enhancement. The measures are WSS, LLR, fwseg-SNR, Cep, Seg-SNR, and SNR. All the measures are computed by segmenting the sentences using 30- ms duration Hamming windows with 75% overlap between adjacent frames. A tenth order LPC analysis was used in the computation of LPC- based objective measure LLR. V. Conclusion In this paper, we proposed an improved wavelet thresholding speech enhancement system, which uses the proposed Super-Soft thresholding algorithm improves the noisy speech wavelet coefficients in a way that avoids sharp time-frequency discontinuities in the speech spectrogram that can decrease the quality of the enhanced speech signal. This system also uses the estimation of the clean speech signal energy for each frame to select the threshold for the thresholding algorithm of the current frame. A further advantage of this algorithm is that unlike most of the other wavelet-based algorithms in which the detection of unvoiced segments highly affects their performances, the proposed method does not require any voiced/unvoiced detection method. The results confirmed the improvement in performance and achievements of our system. Acknowledgements The work is carried out through the research facility at the Department of Electronics & Communication Engineering GKCE, Sullurpet, Nellor-Dt,. Andhra Pradesh, as part of Research Work. The Authors also would like to thank the authorities of JNTUK, Kakinada, A.P, INDIA, encouraging this research work. Our thanks to the experts who have contributed towards development of this Research work. VI. References [1]. P. Loizou, Speech Enhancement Theory and Practice, CRC Press, 2007. [2]. Milner, B.P., Speech enhancement: In Speech Technology for Telecommunications,London. Chapman and Hall, London, 1998. [3]. Hansen, J.H.L., “Speech enhancement. Encyclopaedia of electrical and electronics engineering, Wiley, 20, 159-175, 1999. [4]. Y. Ephraim, H. L. Ari, and W. Roberts, A brief survey of Speech Enhancement, in The Electronic Handbook, 2nd ed. CRC Press, Apr. 2005. [5]. Y. Ephraim and I. Cohen, Recent advancements in Speech enhancement, in The Electrical Engineering Handbook. 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Saram¨ki,Finite impulse response filter design, in Hand book for Digital digital Processing, S. K. Mitra and J. F. Kaiser, Eds., Wiley, New York, NY, USA, 1993. [12]. Rabiner, L. R., McClellan, J. H. and Parks, T., FIR Digital Filter Design Techniques Using Weighted Chebyshev Approximation, Proc. IEEE, Vol. 63, pp. 595 610, April 1975 [13]. Puneet Singla and Tamnaraj Singh Desired order of continuous polynomial Time window functions for Harmonic Analysis, IEEE Transactions on Instrumentation and Measurement, Vol.59, No.9, 2475-2481 September 2010. [14]. S.Boll, Suppression of acoustic noise in speech using spectral subtraction, IEEE Tans Acoust., Speech, Signal Processing, vol. ASSP- 27, pp 113-120, April 1979. [15]. Kamath, S. and Loizou, P. “A multi-band spectral subtraction method for enhancing Speech corrupted by colored noise, Proc. Of International Conference on Acoustics, Speech and Signal Processing, Vol. 4, pp. 4160- 4164, 2002. [16]. Vaseghi, Saeed V., Advanced Digital Signal Processing and Noise Reduction, Second Edition, John Wiley & Sons Ltd, July 2000. [17]. P.Scalart and J. Filho, speech enhancement based on a priory signal to noise estimation",in proc. IEEE Int. Conf. Acoust, speech, Signal Process., 1996 pp.629-632. [18]. J.S.Lim, A.V.Oppenheim, Enhancement and bandwidth compression of noisy speech, in Proc. IEEE, Vol. 67, pp.-1586-1604, December 1979. [19]. Ephraim, Y., and Malah, M., Speech enhancement using a minimum mean-square error short-time spectral amplitude estimator. IEEE Trans. Acoustics, Speech and Signal Process. 32(6), 1109-1121, 1984. [20]. Y. Ephraim and D. Malah, “Speech enhancement using a minimum mean square error log-spectral amplitude estimator, IEEE Trans. Acoust., Speech, Signal process., vol.ASSP-33,No.2, pp. 443,445, Apr. 1985. [21]. David L. Donoho, “De-noising by Soft Thresholding”, IEEE Transactions on Information Theory, vol. 41, No. 3, 1995. [22]. J.I. Agbinya, Discrete Wavelet Transform Techniques in Speech Processing,IEEE Tencon Digital Signal Processing Applications Proceedings, IEEE, New York, NY, 1996, pp 514-519. [23]. Y. Ghanbari , M. Karami, A Modified Speech Enhancement System Based on the Thresholding of the Wavelet Packets, 13th ICEE-2005, Vol. 3, Zanjan, Iran, May 10-12, 2005. [24]. Y.Hu & P.Loizou, “Evaluation of objective measures for speech enhancement.IEEE Trans. Audio speech Lang. process, Vol .16. No.1, pp.229- 238, Jan-2008. [25]. P. Krishnamoorthy, An Overview of Subjective and Objective Quality Measures for Noisy Speech Enhancement Algorithms, IETE technical review, vol 28, issue 4, jul-aug 2011, pp 292-301. [26]. S. China Venkateswarlu, ASR Reddy, K. S. Prasad, Speech Enhancement using Bolls Spectral Subtraction Method based on Gaussian Window, Global Journal of Researches in Engineering: F, E and E Engineering, Volume 14 Issue 6, Version 1.0, pp.9-20, Year 2014.
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978-1-5386-4304-4/18/$31.00 ©2018 IEEE. Biography S.China
Venkateswarlu born in Mittathmakur, a nearby Village of Gudur, Nellore District, Andhra Pradesh, India in 1969. He obtained his B.Tech (Electronics & Communication Engineering) from Nagarjuna University (Acharya Nagarjuna University), A.P, INDIA. He earned his DPH & S.Tech from the College of Medical Technology, Ongole, A.P, India. M.Tech (Digital Systems & Computer Electronics) from JNTU College of Engineering, Kukatpally, Hyderabad, A.P, INDIA. And Pursuing PhD degree from JNTU College of Engineering, Kakinada, A.P, INDIA. All in Electronics & Communication Engineering. He Worked as Associate Lecturer, Assistant Professor, Associate Professor, Professor, and Head of the Department in different Engineering Colleges of Andhra Pradesh, INDIA. Working as an Professor in Dept. of ECE of MRIET-Hyderabad, TG, INDIA. S.China Venkateswarlu is having more than 16 years of teaching experience and published more than 25 papers in referred National and International Journals. He has presented more than15 papers in National and International Journals. He has reviewed one Book on Digital Signal Processing from M/S Persons Educations. He is a Life member of ISTE, CSI, and IAENG. He is working as a Reviewer in Different International Journals. His areas of interest Digital Signal Processing, Speech Processing, Embedded Systems, Digital Image Processing, E-mail IDs: sonagiricv@gmail.com,& cvenkateswarlus@gmail.com Dr. A. Subbarami Reddy born in Anjimedu, a nearby village of Tirupati, Andhra Pradesh, India in 1958. He obtained his M.Sc Physics (Electronics) from S.V University, Tirupati, India. He earned his AMIE from the Institution of Engineers (India), Kolkata, India, M.Tech from now NIT, Kurukshetra, India and PhD degree from Andhra University, Waltair, India all in Electronics and Communication Engineering. He worked as Laboratory Assistant, Associate Lecturer, Lecturer, Assistant Professor, Associate Professor, Professor, Sr.Professor and Head of the Department and Dean in different Engineering colleges of Andhra Pradesh, India. Presently he is the Principal of GKCE, Andhra Pradesh, India. Dr. A. Subbarami Reddy is having more than 30 years of experience and published more than 25 papers in referred International and National Journals. He is a life member of ISTE (India), Fellow of IETE. His areas of interest include Signal Processing and its Sub fields. Email ID: skitprincipal@gmail.com K.SatyaPrasad3 , B.Tech., ME., Ph.D., L.M.I.S.T.E., MIEEE, F.I.E., F.I.E.T.E. is a Ex-RECTOR & PROFESSOR in ECE, in JNT University, Kakinada, A.P, INDIA, Worked in various Engineering College and have 38 years of Teaching, R&D and Industrial experience. He has published 28 research papers in National and International Journals. He has presented 20 National and 2 International Conference Papers. His research interest includes Signal Processing, Speech Processing, Wireless Communications and Embedded Systems. He has guided 04 PhDs, 15 students pursuing PhD under his guidance. He has guided 15 M.Techs and 13 Projects. He has authored or Co-authored book entitled: Electronic Devices and Circuits. He is the Member Curriculum development: Chairman, Board of Studies, JNTU, Hyderabad and GPR College of Engineering, Kurnool. He officiated as : Head of ECE dept. from January 1997 to September 2003, Vice-Principal, JNTUCEK from October 2003 to May 2005 Principal, JNTUCEK from May 2005 to June 2007. He has completed R & D Project sanctioned by Ministry of Human Resources and Development, Government of India. Email ID: prasad_kodati@yahoo.co.in.
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