Performance Analysis of Postprocessing Algorithm and Implementation on ARM7TDMI
Manoj Gupta, Brajesh Kumar Kaushik, Laxmi Chand · 2009
Due to rapid advancement in multimedia communication a growing need emerges for the implementation of audio algorithm in real time. Filters are often used in digital signal processing applications. The filters fast processing is usually done on DSP chips. The calculation of the DSP operations like multiplication and accumulation (MAC) and convolution should be fast enough so that the audio can be heard in real time. In this paper we take up a post processing algorithm for audio that is used to control the bass and treble. For this purpose first order IIR filter is employed. This filter is used due to low computational burden that it places on DSP resources. The method that is presented here is based on transforming a prototype analog filter to the desired digital filter using the technique of bilinear transformation for mapping the poles and zeros in digital domain. The calculated coefficients are used for implementation of audio algorithm for ARM7TDMI. Implemented code is optimized for the real time operations. The analysis is done in MATLAB for showing the effect of treble and bass for different frequencies. The filter mapped in digital domain is stable.