Design of an Optimized Twin Mode Reconfigurable Adaptive FIR Filter Architecture for Speech Signal Processing

被引:1
作者
Padmapriya, S. [1 ]
Jagadeeswari, M. [1 ]
Prabha, Lakshmi, V [2 ]
机构
[1] Sri Ramakrishna Engn Coll, Dept ECE, Coimbatore, Tamil Nadu, India
[2] PSG Coll Technol, Dept ECE, Coimbatore, Tamil Nadu, India
来源
INFORMACIJE MIDEM-JOURNAL OF MICROELECTRONICS ELECTRONIC COMPONENTS AND MATERIALS | 2019年 / 49卷 / 04期
关键词
Adaptive Alter; Least Mean Square Algorithm; Reconfigurable Filtering; Speech Signal Processing; IMPLEMENTATION; POWER;
D O I
10.33180/InfMIDEM2019.406
中图分类号
TM [电工技术]; TN [电子技术、通信技术];
学科分类号
0808 ; 0809 ;
摘要
Reconfigurability, low complexity and low power are the key requirements of FIR filters employed in multi-standard wireless communication systems. Digital Alters are used to filter the audio data stream and increase the reliability of speech signal. Therefore, it is imperative to design an area optimized and low power based reconfigurable FIR filter architectures. The reconfigurable architecture designed in this research is capable of achieving lower adaptation-delay and area-delay-power efficient implementation of a Delayed Least Mean Square (DLMS) adaptive filter with reversible logic gates. The Optimized Adaptive Reconfigurable (OAR) FIR filter architectures are proposed. The optimized architectures are implemented across the combinational blocks by reducing the pipeline delays, sampling period, energy consumption and area, to increase the Power-Delay Product (PDP) and Energy Per Sample (EPS).The noisy speech signals are used for verifying the efficiency of the proposed architectures. By implementing the proposed scheme in signal corrupted by various real-time noises at different Signal to Noise Ratios (SNRs), the efficiency of the architecture is verified.
引用
收藏
页码:241 / 254
页数:14
相关论文
共 31 条
  • [1] [Anonymous], 2009, SPEAR NOIS SPEACH DA
  • [2] [Anonymous], 2008, SPEECH PROCESSING LA
  • [3] Baghel S., 2011, 2011 IEEE Students' Technology Symposium (TechSym), P214, DOI 10.1109/TECHSYM.2011.5783848
  • [4] Baghel S., 2011, 2011 International Conference on Communications and Signal Processing (ICCSP), P443, DOI 10.1109/ICCSP.2011.5739356
  • [5] Becchetti C., 1999, Speech Recognition, Theory and C++ Implementation
  • [6] BLOCK IMPLEMENTATION OF ADAPTIVE DIGITAL-FILTERS
    CLARK, GA
    MITRA, SK
    PARKER, SR
    [J]. IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING, 1981, 29 (03): : 744 - 752
  • [7] Garcia A.L., 2009, STATICS RANDOM PROCE
  • [8] HAIMICOHEN R, 1990, INT CONF ACOUST SPEE, P1273, DOI 10.1109/ICASSP.1990.115604
  • [9] Haykin S., 2003, Least-Mean-Square Adaptive Filters
  • [10] Jayashri R., 2011, 2011 International Conference on Communications and Signal Processing (ICCSP), P179, DOI 10.1109/ICCSP.2011.5739296