DESIGN OF AN EFFECTIVE DIGITAL RECONFIGURABLE FILTER BANK

Transcription

1 696 DESIGN OF AN EFFECTIVE DIGITAL RECONFIGURABLE FILTER BANK UMAR.M 1, SUBBULAKSHMI.N 2 1ME VLSI DESIGN Student, SRI RAMAKRISHNA ENGINEERING COLLEGE, COIMBATORE, India (Member of IEEE) 2Assistant professor, Department of ECE (PG-ME VLSI DESIGN), SRI RAMAKRISHNA ENGINEERING COLLEGE, COIMBATORE, India 1 er.umar@yahoo.com, 2 lakshu.125@gmail.com ABSTRACT Fixed filter bank cannot provide flexible decomposition plans for the compensation of different hearing impairment. This paper proposes a reconfigurable filter bank that consists of a multiband-generation block and a sub band-selection block. According to some control parameters different sub bands can be produced without affecting the structure of the filter bank. The interpolation, decimation, and Frequency-response masking techniques are used to reduce the computational complexity. It concentrates on the hardware complexity by realizing the entire system with only three prototype filters. Various optimization techniques are used to achieve low power and area utilization.these techniques are discussed by means of examples, the proposed filter bank can achieve a better matching to the audiogram and has smaller complexity compared with the fixed filter bank. Keywords: Decimation, interpolation, frequency response masking, reconfigurable filter bank [1] INTRODUCTION HEARING impairment is one of the most common sensory disturbances in the world. An effective treatment for the problem is hearing assistive devices. The main function of a hearing-aid device is to amplify the sound selectively, and then transfer the processed signal to the ear. Much study has been invested into the design of digital filter banks for selective amplification. Most of the current designs use filter banks with fixed sub bands. One approach is to use uniform filter-banks. While the previous research aims at realizing the band decomposition, recent research focuses on reducing the complexity of the algorithms besides the accomplishment of the decomposition. Lattice wave digital filter banks (LWDFBs) were used to process the sound waves in LWDFBs have lower complexity than finite-impulse response (FIR) filter banks and are not sensitive to the coefficients. In, a DFT filter bank with the multidimensional logarithmic number system was realized to reduce the complexity. In, some well-known simple methods for critically sampled filter banks were extended to the oversampled case. All the filter banks mentioned earlier have fixed sub bands, thus they cannot provide flexible sound decomposition plans ac-cording to the characteristics of different types of hearing loss. It is attractive to design filter banks with adjustable sub bands that can be customized for an individual hearing-loss case. Little work has been done in this area. 2. ADAPTIVE FILTER BANK Adaptive filter bank is the existing one. Which has the drawback of fixed sub band, thus cannot provide the flexible sound decomposition. The existing block diagram is shown in Fig 1. Fig. 1. Structure of the Adaptive filter block 2.1 Output Response The output response of the existing adaptive filter bank is shown in the Fig 2.the output response consist of the

2 697 original and the attenuated signal coming from the fixed sub band via audio signal, the response is plot between the time index and the signal value. Fig. 2 Output response of the Adaptive filter block 3. PROPOSED FILTER BANK Reconfigurable means that the subbands are adjustable ac-cording to some control parameters without changing the structure of the filterbank. The proposed filterbank consists of two blocks. One is the multibandgeneration block whose function is to produce magnitude responses having multiple passbands. The other block is the subband-selection block whose function is to extract the desired subbands. Control information is sent to both blocks so that the blocks can be reconfigured accordingly. The proposed structure for the FIR reconfigurable block is shown in Fig. 3. Fig. 3. Structure of the FIR reconfigurable block 3.1 Multiband generation block To make the multiband-generation block reconfigurable, interpolation and decimation techniques. In order to improve the individuality of digital hearing aids, in this paper, an FIR reconfigurable filter bank is proposed. The proposed filter bank is expected to provide multiple band decomposition plans and has small computational complexity. To make the filter bank reconfigurable, interpolation and decimation techniques are used. To lower the complexity, frequency-response masking technique is employed. Multi band generation block consist of interpolation and decimation section, it will helps to improve output effectively. Interpolation is the process of up sampling and the decimation is the process of down sampling which helps to compress and expand the input sample given to the pulse generator. 3.2 Sub band selection block Sub band selection block consist of three branches, its takes the input from the multiband generation block. The input is followed by Ai,Bi,Ci,which is given to the storage block. Depending upon the switch selection, response is given to the masking filter bank and the final response given to the output side. 3.3 Design procedure The design procedure of the proposed filter bank can be de-scribed as follows Determine the control parameters decimation(d1&d2) and interpolation. Design the masking filter bank for each interpolation.the complementary filter and the highpass symmetric filter of the prototype filter can be used to simplify the design. The prototype filter is dependent on the output of the multiband generation block thus should be discussed

3 case by case. 3.4 Output response 698 Fig. 4. Output response of the FIR reconfigurable block 4. DESIGN EXAMPLE The idea of the proposed filterbank can be illustrated by an following block diagram and that ouput response.the structure of the masking filter bank is shown in the Fig 5 masking filter bank is used to lower the complexity of the filter. 4.1 Design of masking filter bank The above block diagram consist of two subfilter,two transfer function and three switches.the input frequency is given to the chirp block and that frequency is travel through the fist order and second order transfer function.the

4 699 output respone is shown in the scope block.the pupose of the masking filter bank is to be improves the efficient of the output response and also lower the complexity 4.2 Output response Fig.5. Structure of the masking filterbank Fig. 6. Output response of the masking filter bank The output response of the masking filter bank shown in the fig.6.depends upon the different frequencies the output may varies. The fig.6 shows the output response of the 25 hz frequency.the input given to the chirp signal block,which is sown in the fig.5 that frequency is travel through the different order transfer function ant reach the scope block, from that block we get the required response. CONCLUSIONS Finally conclude that efficient way to design a reconfigurable digital FIR filterbank is proposed for hearing aids. The system consists of a multiband-generation block and a subband selection block. By changing the value of the control signals, the transfer function of the filterbank can be changed, thus providing different sound decomposition methods. The output of the first stage is manipulated to separate adjacent bands, thus the complexity of the masking stage is reduced greatly. Only three prototype filters are needed for the whole system.. It should be noted that the delay of the proposed structure is relatively large for a real hearing-aid system REFERENCES [1] M. A. Hersh,M. A. Johnson, C. Andersson, D. Campbell, A. Farquharson,S. Furner, J. Gill, A. Jackson, and J. Lucker, Assistive Technology for the Hearing-Impaired, Deaf and Deaf-Blind. London, U.K.: Springer- Verlag, [2] A. M. Engebretson, Benefits of digital hearing aids, IEEE Eng. Med.Biol. Mag., vol. 13, no. 2, pp , Apr./May [3] A. R. Moller, Hearing: Anatomy, Physiology and Disorders of the Auditory System, 2nd ed. Orlando, FL, USA: Academic Press, Sep. 11, [4] Y. C. Lim, A digital filterbank for digital audio systems, IEEE Trans.Circuits Syst., vol. 33, no. 8, pp , Aug [5] T. Lunner and J. Hellgren, A digital filterbank hearing aid design, implementation and evaluation, in Proc. Int. Conf. Acoust., Speech Signal Process., Apr. 1991, vol. 5, pp [6] T. Schneider and R. Brennan, A multichannel compression strategy for a digital hearing aid, in Proc. IEEE Int. Conf. Acoust., Speech Signal Process., 1997, vol. 1, pp

5 700 [7] H. Li, G. A. Jullien, V. S. Dimitrov, M. Ahmadi, and W. Miller, A 2- digit multidimensional logarithmic number system filterbank for a digital hearing aid architecture, in Proc. IEEE Int. Symp. Circuits Syst., AZ, pp [8] R. Cassidy and J. O. Smith, A tunable, nonsubsampled, nonuniform filterbank for multi-band audition and level modification of audio signals, in Proc. 38th Asilomar Conf. Signals, Syst. Comput., Nov. 7 10, 2004, vol. 2, pp [9] Y. Lian and Y.Wei, A computationally efficient non-uniform FIR digital Filterbank for hearing aid, IEEE Trans. Circuits Syst. I: Reg. Papers, vol. 52, pp , Dec [10] K. S. Chong, B. H. Gwee, and J. S. Chang, A 16-channel low-power nonuniform spaced filterbank core for digital hearing aid, IEEE Trans. Circuits Syst., vol. 53, no. 9, pp , Sep [11] Y.-T. Kuo, T.-J. Lin, Y.-T. Li, and C.-W. Liu, Design and implementation of low-power ANSI S1.11 filterbank for digital hearing aids, IEEE Trans. Circuits Syst. I: Reg. Papers, vol. 57, no. 7, pp , [12] A. B. Hamida, An adjustable filter-bank based algorithm for hearing aid systems, in Proc. Int. Conf. Ind. Electron., Control Instrum., 1999, vol. 3, pp [13] Y. C. Lim and Y. Lian, The optimal design of one- and two-dimensional FIR filters using the frequency response masking technique, IEEE Trans. Circuits Syst., II Analog Digit. Signal Process., vol. 40, pp , Feb [14] Y. Lian and Y. C. Lim, Reducing the complexity of frequency-response masking filters using half band filters, Signal Process., vol. 42, no. 3, pp , Mar [15] Y. Lian, Complexity reduction for FRM-based FIR filters using the prefilter- equalizer technique, Circuits, Syst. Signal Process., vol. 22, no. 2, pp , Mar [16] M. A. Stone and B. C. J. Moore, Tolerable hearing-aid delays: III. Effects on speech production and perception of across-frequency variation in delay, Ear and Hearing, vol. 24, no. 2, pp , Apr [17] B. C. J. Moore, Perceptual consequences of cochlear hearing loss and their implications for the design of hearing aids, Ear and Hearing, vol. 17, no. 2, pp , Apr [18] T. B. Deng, Three-channel variable filter-bank for digital hearing aids, IET Signal Process., vol. 4, no. 2, pp , Apr ] N. Ito and T.-L. Deng, Variable-bandwidth filter-bank for low-power hearing aids, in Proc. 3rd Int. Congr. Image Signal Process., 2010, pp [19] Y.Wei and D. Liu, A design of digital FIR filterbanks with adjustable subband distribution for hearing aids, in Proc. 8th Int. Conf. Inf., Commun. Signal Process., Singapore, Dec. 2011, pp [20] R. Mahesh anda. P.Vinod, Coefficient decimation approach for realizing reconfigurable finite impulse response filters, in Proc. IEEE Int. Symp. Circuits Syst., Seattle, USA, May 2008, pp [21] Y. C. Lim, Frequency-response masking approach for the synthesis of sharp linear phase digital filters, IEEE Trans. Circuits. Syst., vol. 33, no. 4, pp , Apr [22] B. C. J. Moore, Perceptual consequences of cochlear hearing loss and their implications for the design of hearing aids, Ear Hearing, vol. 17, pp , Apr [23] A. Boothroyd, K. Fitz, J. Kindred, S. Kochkin, H. Levitt, B.C. Moore, and J. Yanz, Hearing aids and wireless technology, The Hearing Review, June [24] N.S. Jayant and P. Noll, Digital Coding ofwaveforms, Prentice Hall, [25] R. Brennan and T. Schneider, Flexible filterbank structure for extensive signal manipulations in digital hearing aids, in Proceedings of the IEEE International Symposium on Circuits and Systems (ISCAS 1998), Monterey, CA, USA, May 1998, vol. 6, pp [26] D. Hermann, R. L.Brennan, H. Sheikhzadeh, and E. Cornu, Low-power implementation of the Bluetooth subband audio codec, in Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2004), Montreal, Canada, May [27] J. Johnson, D. Hermann, M. Witter, E. Cornu, R. Brennan, and A. Dufaux, An ultra-low power subbandbased electronic stethoscope, in Proceedings of the IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP 2006), Toulouse, France, May [28 E. Cornu, A. Dufaux, and D. Hermann, A high performance, low latency, low power audio processing system for wideband speech over wireless links, in Proceedings of the European Signal Processing Conference (EUSIPCO 2006), Florence, Italy, September [29] R.E. Crochiere and L.R. Rabiner, Multirate Digital Signal Processing, Prentice Hall, [30] P.P. Vaidyanathan, Multirate Systems and Filter Banks, Prentice Hall, [31] H. S. Malvar, Signal Processing with Lapped Transforms, Artech House, [32] T. Karp and N. J. Fliege, Modified DFT filter banks withperfect reconstruction, IEEE Transactions on Circuits and Systems II:Analog and Digital Signal Processing, vol. 46, no.11, pp , November 1999.