IJERT-EMS
IJERT-EMS

Sub Band Coding of Speech Signal by using Multi-Rate Signal Processing


Sub Band Coding of Speech Signal by using Multi-Rate Signal Processing
Authors : Vijayakumar Majjagi
Publication Date: 31-08-2013

Authors

Author(s):  Vijayakumar Majjagi

Published in:   International Journal of Engineering Research & Technology

License:  This work is licensed under a Creative Commons Attribution 4.0 International License.

Website: www.ijert.org

Volume/Issue:   Vol.2 - Issue 9 (September - 2013)

e-ISSN:   2278-0181

Abstract

Abstract: Interest in signal processing long predates computers. As long as people have tried to send or receive information through electronic media, such as telegraphs, telephones, television, radar, etc., there has been the realization that these signals may be affected by the system used to acquire, transmit, or process them. Sometimes these systems are imperfect and introduce noise, distortion, or other artifacts. Understanding the effects these systems have and finding ways to correct them is the foundation of signal processing. There are many types of signal processing. Among that Digital signal processing is more efficient and widely used. Multirate systems are building blocks commonly used in digital signal processing In conventional speech processing applications, speech signal is encoded using fixed number of bits over the entire speech signal band. During the process, the bandwidth requirement for speech transmission is relatively high which is of concern. The QMF (Quadrature Mirror Filter) banks are the fundamental building blocks for spectral splitting. The technique is developed to design the so-called perfect reconstruction QMF bank, which allows complete elimination of amplitude and phase distortion of the reconstructed signal. The low pass filtered signal is decimated and encoded with more number of bits and high pass filtered signal is also decimated and encoded with less number of bits. These two bit streams are multiplexed and transmitted. In receiver side the received signal is de-multiplexed and decoded. The signal is passed through the interpolators and then through the synthesis filters so as to reconstruct the speech signal. The reconstructed signal is compared with the original speech signal.

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