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Audio De-noising by Spectral Subtraction Technique


Kajol Bayeti , shrinathji institute of technology & engineering; Prof. Pankaj Rathi, shrinathji institute of technology & engineering


Audio De-noising, Spectral Subtraction Technique


Hearing aids are now used to alleviate hearing impairments. However, more than 60% of impaired persons feel uncomfortable when using their hearing-aids because of the worse intelligibility resulting from bad speech comprehensibility. We believe that hearing impairments can be alleviated by a system with characteristics closer to being body’s ones. Research has been deeply involved in developing new algorithms to improve speech intelligibility. Although, lot of researches was held to enhance speech for impaired people, few authors deal with the problem of power consuming. No numerical results are given, making the comparison only with the hearing aids of the market. Most closely related to our approach is the work of, who provide a method for power reducing based on algorithm and hardware optimizations along with the architecture uses the odd/even data lifting. Our work is improved by avoiding the access to the memory for data storage. Instead, we formulate our algorithm under which a given data of the speech signal is segmented at the input and each segment is processed individually. In order to avoid losing of data information, the segments are overlapped and an overlap technique is then used for treatment. This approach provides perfect analysis and efficient computation. Consequently, the framework devised here is made generic and requires simulations and empirical evaluation of the routing scheme in order to be applied. Traditionally, Digital Signal Processing (DSP) algorithms are implemented using General Purpose Processors (GPP) for low rate applications. These devices showed limited capabilities for processing high volume data efficiently in real time. The trends had then been shifted to Special Purpose DSP (SPDSP) and Application Specific Integrated Circuits (ASICs) in order to meet the increased complexity and to gain in performance requirements of these algorithms but with high cost functions. Today, FPGAs are highly preferred for their relatively high capacity, low cost, short design cycle and short time to market. FPGA affords the capability of constant reconfiguration to meet application performances. Dealing with digital speech processing as it pertains to the hearing impaired persons especially for miniaturized system applications; FPGA allows increasing sophisticated features to be built for better sound reproduction while keeping small size and low power consumption of the devices. Fortunately, simulation tools provide us a rapid design and basic information. Similarly, a high-level programming language is an efficient comparison tool for the final output results and system evaluation. In practice, the implementations are often subject to lot of limitations. Using DWT at multi-resolution over disjoint bands remains up to now a practical necessity for perfect design for digital speech processing in particular and herein some references from our work, where the goal is to investigate noise reduction and hardware implementation. This work extends previous research described in. In this we present the implementation of a multi-level one dimension DWT combined to an OLA on FPGA for a bio-inspired medical hearing aid application. The methodology aims to improve in one side better speech quality and in the other side, an efficient flexible reconfiguration and reduced cost functions. The scheme represents architecture for de-noising and frequency shifting. It is realized targeting a DE2 development kit board of Altera (EP2C70F896) and results are compared to that obtained in Matlab. The system provides a generic framework allowing the use of DWT analysis / synthesis with frequency shaping of the speech signal to improved speech intelligibility. We present some simulation results under VHDL and Matlab. Hence, a comparative study is done based on the Mean Square Error (MSE) and the Signal to Noise Ratio (SNR). MOS evaluations are presented for speech intelligibility and the gain obtained by the proposed architecture.

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Paper ID: IJSRDV5I80128
Published in: Volume : 5, Issue : 8
Publication Date: 01/11/2017
Page(s): 32-34

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