Equalizer is fundamentally audio consequence device to stress or deemphasize certain specific frequence constituents of a signal. The chief use of equaliser is to flatten the system spectral response [ 1 ] . In order to hold our desired response of system for given input audio signal we used filtrating device. Using filtering devices different methods had been developed. Graphic Equalizer is one of method used which consist of audio equipment for flattening the system spectral response in the audio signal set or bring forth other desirable effects. The study demonstrates the execution of in writing equaliser by utilizing DSK ( DSP Starter Kit ) C6713 kit interfaced with lab position package tool to analyse the response of sound and frequence in clip sphere every bit good as in frequence sphere.
Key Words – Equalizers, Band base on balls filters, low base on balls filters, high base on balls filters, band stop filters, and Lab position.
Equalizers are fundamentally electrical filter webs which generate their end product behavior alteration with frequences. The most good known equalisers which are available normally are amplitude equalisers, their basic property is that end product signal alterations with regard to amplitude [ 2 ] .Equalizers are of many types for case parametric equaliser, postponing equaliser, active etc, but here we used in writing equaliser. Equalizers are used in many topographic points: in broadcast senders and receiving systems, sound systems, tape recording and reproduction etc. Equalizers have batch of applications, they are used to get the better of loss in storage systems and many more applications [ 2 ] . The forte of in writing equaliser is to extinguish the noise with great end product of audio sound quality. In this undertaking we focus on in writing equaliser. The chief intent of the undertaking is to deploy digital signal processor kit C613 for processing and analysing different frequence signals in the sound, and so eventually DSP kit is interfaced with lab position to look into the feature of signals in both the scenarios in clip sphere and frequence sphere.
The tool which was used to plan in writing equaliser is code composer studio. This tool is used for debug and develops embedded applications.
Lab position is fundamentally designed to imitate different communicating systems, here in this undertaking the motivation was to complect codification composer studio with lab position by utilizing C codification programming to bring forth the end product frequence response in existent clip informations exchange environment.lab position is the environment to make suited scenario for programming linguistic communications by utilizing different instruments. One of the chief advantages of lab position over other scenarios is to assist the users to entree hardware easy. Lab position has the compiler that generates codification for CPU. Lab position has batch of applications, for case informations geting, signal generating, mathematical operations, statistical operations and signal analysis etc [ 3 ] .
2. PRE STUDY
Before get downing the execution of in writing equaliser by utilizing different techniques, it would be necessary to hold better apprehension of digital signal processing techniques. The most common techniques used are: FIR ( Finite Impulse Response ) , IIR ( Infinite Impulse Response ) , FFT ( Fast Fourier Transform ) and IFFT ( Inverse Fast Fourier Transform ) .Different illustrations illustrate the execution of these digital signal processing techniques. Every technique has its ain properties and effects. For case IIR is more executable in footings of algorithms, requires less clip to be executed in footings of rhythms but needs complex computation for filter coefficients. On the other manus the demand of FFT is high with regard to try points and requires complex executing to extinguish deformation from input signal. The basic thought which works behind FFT is the algorithm that is responsible for change overing clip domain signal into frequence domain signal that depends upon distinct Fourier transform. From acute survey and the execution of these digital signal processing techniques we came to cognize that FIR technique is the most suited and dependable technique for the execution of equaliser. Because FIR filters are additive stage and they are more compatible with digital signal processors. This technique is the simplest and executable with hardware as compared to IIR and FFT [ 4 ] .
Furthermore there should be sufficient cognition about matlab FDA tool execution on proposed filters like high base on balls, low base on balls, set base on balls and set halt. Additionally the concluding stage of the undertaking demands precise survey of interfacing DSP kit with lab position. It is necessary to cognize about the operation and functionality of LAB View tool [ 3 ] .
3. Experimental apparatus
The architecture TMS320C6713 ( C6713 ) consist of the C6713 floating-point digital signal processor and a 32-bit stereo codec TLV320AIC23 ( AIC23 ) for input and end product. The onboard codec AIC23 [ 1 ] uses a sigma-delta engineering that provides ADC and DAC. It connects to a 12-MHz system clock. Variable trying rates from 8 to 96 kilohertzs can be set readily.
The DSK board includes 16MB ( Ms ) of synchronal dynamic random entree memory ( SDRAM ) and 256kB ( Ks ) of brassy memory.
4. Undertaking Description
A. Filter Design
The FIR filtering method is used for execution of assorted of DSP Design circuits as being the most efficient. For our class undertaking we use 4 separate FIR filters with different frequence specification are designed. The item specification can be found in C codification of Equalizer. We can set the addition of the end product of every filter such that it provides elaboration for the specific frequence constituent of the audio signal. This manner the signal can be compensated which is distorted by some noise hence doing it sound better. The equaliser designed here covers the frequence scope from 0 to 4.4 KHz. Most of physical resonances of aduio instruments normally fall between, say 1 or 2 KHz these frequences are likely get downing points. Therefore different frequence constituents can be altered efficaciously. We used four different types of filter which are Low base on balls, High Pass, Band Pass and Band Stop.
Equalizer filter sets
Band Frequency ( KHz )
Low Pass filter
High Pass filter
1 to 3.3
Band Pass Filter
3.3 to 4
Band Stop Filter
4 to 4.4
The input voice is split into 4 different frequence constituents by go throughing it through 4 FIR filters with different frequence features. Such that we have had level end product. The constituents are multiplied with their several user configurable addition to increase or diminish their magnitude. They are added all together and we had modified end product signal. We connect all the FIR filters of equal length in analogue, multiply them with their several addition and sum them into a individual filter. In this manner we increasing the efficiency and the work load of the equaliser is reduced work load to a greater extent.
MATLAB FDATool was used to plan the FIR filters and we can besides utilize SPTool. It can be accessed by typing the ‘fdatool ‘ bid in the MATLAB bid window. The coefficients obtained with matlab are in drifting point format so we changed it to the whole number format by unit of ammunition bid of Matlab. The coefficients obtained are in drifting point format. We change it to fixed-point figure representation improved the public presentation and truth of the Microprocessor. The Hamming Window method is used to deduce the filter coefficients. Such that we used filter order of 40 The filter size is kept at 40 which was the minimal size at which end product was of good quality. When I used higher order filters the end product was distort. Due to ground that higher order filters utilize great resources of DSP Processor and calculation take long clip. So we had distorted end product.
We had showed in Figure 1 how we can used FDATool of Matlab. We besides so compose C codification for FIR filters such that we connect it decently.
In order to alter Gain value with slider map in C codification Composer multiplied with their several filter coefficients and added together to do a filter to cut down computational power. But in our undertaking we used Lab View for implementing slider map. Such that we place skidders on front panel of Lab View and name it with volume 1, volume 2, volume 3, and volume 4.
We know that two subdivision of our undertaking assembly code running in DSP starter Kit or DSK and Lab View codification, supplying user interface must pass on each other.
So both can pass on onboard Joint Test Action Group ( JTAG ) interface through Real-time informations exchange ( RTDX ) channels.
We know that lab position is data acquisition, control systems, frequence responses and mathematical graphs. So we place Spectral measuring Express VI ( practical instrument ) for measuring of FFT RMS magnitude Figure 2.we had end product in clip sphere.
From image it appear that we have no end product in wave form graph instrument but as treating velocity of DSP is really fast so we have no graphical show. But when we used Probe option on wire linking Spectral measuring with wave form graph in Block Diagram of Project we can hold FFT RMS value shown in Figure 3.
Though end product quality is good from our equaliser we designed and implemented on DSP Kit. But we can better our end product in different ways. If use filter of less order the computational efficiency of DSP starter Kit can be increased. We can better quality of filtering of sound and have pleasant by utilizing Lab View Embedded Edition. It is a particular edition of Lab VIEW that installs in a separate directory and does non interfere with Lab VIEW Base, Full, or Professional development systems. The DSP Module is illustration of Labview Embedded edition which when installed. We can utilize graphical scheduling methods to larn DSP basicss and to develop applications for DSP hardware without holding to compose any C, assembly, or book beginning. So we will had to non compose the RTDX codification for pass oning with DSP starter Kit of C6713.Thus Lab View will pass on straight and we will hold optimized end product.
We appreciate the counsel of Mr. Kerstin Nilsson of School Engineering Karlskorna in transporting out this undertaking. Indeed it was invaluable chance to work in Digital Signals Processor field.
Digital Signal Processing and Applications with the C6713 and C6416 DSK By Rulph Chassaing ISBN 0-471-69007-4 Copyright A© 2005 by John Wiley & A ; Sons, Inc
TMS320C6000 CPU and Instruction Set Reference Guide, SPRU189F, Texas Instruments, Dallas, TX, 2000.
[ 3 ] www.Labivew.com