978-1-5090-5170-0/16/$31.00 ©2016 IEEE Implementation of Finite Impulse Response Digital Filter in Digital Signal Processor Kit for Voice Signal Application Endang Djuana, Suhartati Agoes, R. Deiny Mardian, Revi Noviananda Nurmalasari Department of Electrical Engineering, Faculty of Industrial Technology Trisakti University, Jakarta, Indonesia {edjuana, sagoes, deiny_wp}@trisakti.ac.id, revi.noviananda@gmail.com Abstract— Transmission and receiving of voice signal must be clear and understandable. To maintain the voice quality, filtering is needed. Filtering is a part of signal processing to improve the output signal quality. Digital filter is being used for filtering. Finite Impulse Response (FIR) filter has many advantages such as linear phase, stability, and minimum error caused by quantization. In this research, a filter implementation using digital signal processor (DSP) kit device with voice signal as input has been done. Voice signal will experience Analog to Digital Conversion (ADC), filtering and Digital to Analog (DAC) processes. Filtering outputs are heard on the loudspeaker and the graphs are shown on Matlab scope. Keywords— voice signal, FIR digital filter, DSP kit I. INTRODUCTION Rapid development of technology includes digital processing of voice signal. Voice is human common media for communication [1]. Received voice signal must be clear. However, signal processing stages are frequently influenced by noise. To maintain the voice quality, filtering is needed. Filtering is a part of signal processing which improves the output quality [2] [3]. The most common filters being used in telecommunication area are analog filter and digital filter. But digital filter is more dominant [4]. This is caused by the Very Large Scale Integration (VLSI) technology which makes smaller sized digital filter possible, which can operate at very low frequency, have linear phase response, insensitive to temperature and have flexibility to change its frequency response [4]. Digital filter is divided into two categories: Infinite Impulse Response (IIR) and Finite Impulse Response (FIR). The division is based on its impulse response [5]. FIR has impulse response with finite length. FIR has several advantages such as linear phase, good stability, and less error caused by quantization [6]. FIR digital filter consists of four types: low pass, high pass, band pass, and band stop. Several investigations have been done in relation to FIR application for voice signal processing but they are limited to software simulation [7] [8]. While FIR digital filter using DSP kit for voice signal as the input has not been investigated [9]. Therefore, in this research, voice signals of male and female with age range of 22 – 28 years old are investigated. The voices have been recorded and saved in .wav format. This research uses Matlab and Code Composer Studio (CCS) as the software platform. FIR digital filter are developed for four types of filter and uses Hamming method as the windowing method. Filtering process is implemented on DSP kit device. The output signals are heard on the speaker and the graphs are shown on Matlab scope. Finally, filtering outputs of four filter types are analyzed. II. VOICE SIGNAL Voice signal is a signal representative of the sound. The signal is formed from a combination of frequency, amplitude, and phase. In time domain, voice signal is represented in the form of voltage or current in function of time. Voice signal in the time domain is shown in Figure 1. Figure 1. Voice signal in time domain While in frequency domain, voice signal is represented in the form of amplitude and phase in frequency function. Voice signals in frequency domain is shown in Figure 2. Figure 2. Voice signal in frequency domain