Abstract
Filter bank theory is used to identify the approximation and detail coefficients of the wavelet filter that are used to identify the scaling and wavelet function of the wavelet. If the filter bank coefficients of the sounds of musical instruments are calculated then it is possible to identify the signature wavelet of the sound signal, which can be used to reconstruct the original signal with negligible error. The filter bank coefficients can be identified with adaptive algorithms viz. LMS, NLMS and RLS. Among the three algorithms, RLS algorithm perform better in all regard and the algorithm converges very fast i.e. number of iterations are less. Hence an algorithm based on RLS algorithm is developed to find out scaling and wavelet functions of the sounds of musical instruments with better accuracy and speed of convergence.
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References
Ramchandran K, Vetterli M, Herley C (1996) Wavelets sub band coding and best bases. Proc IEEE 84(4):541–560
Vetterli M, Harley C (1992) Wavelets and filter banks theory and design. IEEE Trans Signal Process 40(9)
Chapa JO, Rao RM (2000) Algorithms for designing wavelets to match a specified signal. IEEE Trans Signal Process 48(12):3395–3406
Sinith MS, Nair MN, Nair NP, Parvathy S (2010) Identification of wavelets and filter bank coefficients in musical instruments. In: IEEE conference on audio language and image processing, pp 727–731
Tewfik A, Sinha D, Jorgensen P (1992) On the optimal choice of a wavelet for signal representation. IEEE Trans Inf Theory 38(2):747–766
Gianpolo E (1993) Pitch synchronous wavelet representation of speech and music signals. IEEE Trans Signal Process 41(12):3313–3330
Soria E, Calpe J, Chambers J, Martinez M, Camps G, Guerrero JDM (2008) A novel approach to introducing adaptive filters based on the LMS algorithm and its variants. IEEE Trans 47:127–133
Markus R (1993) The behavior of LMS and NLMS algorithms in the presence of spherically invariant process. IEEE Trans Signal Process 41(3):1149–1160
Razzaq MS, Khan NM (2010) Performance comparison of adaptive beam forming algorithms for smart antenna systems. World Appl Sci J 11(7):775–785
Chen T, Vaidyanathan PP (1994) Vector space framework for unification of one and multidimensional filter bank theory. IEEE Trans Signal Process 42(8):2006–2021
Douglas SC, Markus R (2000) Convergence issue in the LMS adaptive filter. CRC Press LLC
Sharma R (2015) Denoising, coding and identification of sounds of some musical instruments, PhD Thesis
Sharma R, Pyara VP (2012) Denoising of sounds of musical instruments by RLS adaptive algorithm. Int J Adv Res Comput Sci 3(5):1–4
Sharma R, Pyara VP (2012) Comparative study of adaptive algorithms for identification of filter bank coefficients of wavelets. In: Proceedings, international journal of computer applications® (IJCA), pp 21–25
Sharma R, Pyara VP (2012) A comparative analysis of mean square error adaptive filter algorithms for generation of modified scaling and wavelet function. Int J Eng Sci Technol 4(4):1402–1407
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Sharma, R., Pyara, V.P. (2017). Signature Wavelet Identification of Sounds of Musical Instruments Using RLS Algorithm. In: Attele, K., Kumar, A., Sankar, V., Rao, N., Sarma, T. (eds) Emerging Trends in Electrical, Communications and Information Technologies. Lecture Notes in Electrical Engineering, vol 394. Springer, Singapore. https://doi.org/10.1007/978-981-10-1540-3_27
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DOI: https://doi.org/10.1007/978-981-10-1540-3_27
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