Anyone give me Media players Graphic equalizer Source code and Algorithm?

Plz can anyone give me Media players(WMP,Winamp,etc) Graphic equalizer Source code and Algorithm?





I don't need exactly the same Media players code but just a basic code for starting the design for Graphic Equalizers


(Visual C/c++ Language or VB or anyone)





What r the prerequisite knowledge to be had before goin to design graphic equalizer





Plz do also give a brief algorithm for it if possible





Thanks in advance

Anyone give me Media players Graphic equalizer Source code and Algorithm?
Google for IIR biquad filter (which is normally a good choice for this sort of thing).





You might also wish to look into pole/zero analysis in the S domain and the the Z transform for the gory detail of how to get a given filter behaviour, and the Parks/Mclaren algorithm for how to go from the Z plane equation to the recurrence relation you need to implement the filter.





Spending a minute in the music dsp mailing list archive turned the following 3 band C code up (converting to C++ should be trivial):





Code courtesy of : Neil C / Etanza Systems, 2006 :)


--------------------------------------...


Simple 3 band equaliser with adjustable low and high frequencies ...





Fairly fast algo, good quality output (seems to be accoustically transparent with all gains set to 1.0)





How to use ...





1. First you need to declare a state for your eq





EQSTATE eq;





2. Now initialise the state (we'll assume your output frequency is 48Khz)





set_3band_state(eq,880,5000,480000);





Your EQ bands are now as follows (approximatley!)





low band = 0Hz to 880Hz


mid band = 880Hz to 5000Hz


high band = 5000Hz to 24000Hz





3. Set the gains to some values ...





eq.lg = 1.5; // Boost bass by 50%


eq.mg = 0.75; // Cut mid by 25%


eq.hg = 1.0; // Leave high band alone





4. You can now EQ some samples





out_sample = do_3band(eq,in_sample)








Have fun and mail me if any problems ... etanza at lycos dot co dot uk








Neil C / Etanza Systems, 2006 :)











Code :


First the header file ....


//------------------------------------...


//


// 3 Band EQ :)


//


// EQ.H - Header file for 3 band EQ


//


// (c) Neil C / Etanza Systems / 2K6


//


// Shouts / Loves / Moans = etanza at lycos dot co dot uk


//


// This work is hereby placed in the public domain for all purposes, including


// use in commercial applications.


//


// The author assumes NO RESPONSIBILITY for any problems caused by the use of


// this software.


//


//------------------------------------...





#ifndef __EQ3BAND__


#define __EQ3BAND__








// ------------


//| Structures |


// ------------





typedef struct


{


// Filter #1 (Low band)





double lf; // Frequency


double f1p0; // Poles ...


double f1p1;


double f1p2;


double f1p3;





// Filter #2 (High band)





double hf; // Frequency


double f2p0; // Poles ...


double f2p1;


double f2p2;


double f2p3;





// Sample history buffer





double sdm1; // Sample data minus 1


double sdm2; // 2


double sdm3; // 3





// Gain Controls





double lg; // low gain


double mg; // mid gain


double hg; // high gain





} EQSTATE;








// ---------


//| Exports |


// ---------





extern void init_3band_state(EQSTATE* es, int lowfreq, int highfreq, int mixfreq);


extern double do_3band(EQSTATE* es, double sample);








#endif // #ifndef __EQ3BAND__


//------------------------------------...





Now the source ...


//------------------------------------...


//


// 3 Band EQ :)


//


// EQ.C - Main Source file for 3 band EQ


//


// (c) Neil C / Etanza Systems / 2K6


//


// Shouts / Loves / Moans = etanza at lycos dot co dot uk


//


// This work is hereby placed in the public domain for all purposes, including


// use in commercial applications.


//


// The author assumes NO RESPONSIBILITY for any problems caused by the use of


// this software.


//


//------------------------------------...





// NOTES :


//


// - Original filter code by Paul Kellet (musicdsp.pdf)


//


// - Uses 4 first order filters in series, should give 24dB per octave


//


// - Now with P4 Denormal fix :)








//------------------------------------...





// ----------


//| Includes |


// ----------





#include %26lt;math.h%26gt;


#include "eq.h"








// -----------


//| Constants |


// -----------





static double vsa = (1.0 / 4294967295.0); // Very small amount (Denormal Fix)








// ---------------


//| Initialise EQ |


// ---------------





// Recommended frequencies are ...


//


// lowfreq = 880 Hz


// highfreq = 5000 Hz


//


// Set mixfreq to whatever rate your system is using (eg 48Khz)





void init_3band_state(EQSTATE* es, int lowfreq, int highfreq, int mixfreq)


{


// Clear state





memset(es,0,sizeof(EQSTATE));





// Set Low/Mid/High gains to unity





es-%26gt;lg = 1.0;


es-%26gt;mg = 1.0;


es-%26gt;hg = 1.0;





// Calculate filter cutoff frequencies





es-%26gt;lf = 2 * sin(M_PI * ((double)lowfreq / (double)mixfreq));


es-%26gt;hf = 2 * sin(M_PI * ((double)highfreq / (double)mixfreq));


}








// ---------------


//| EQ one sample |


// ---------------





// - sample can be any range you like :)


//


// Note that the output will depend on the gain settings for each band


// (especially the bass) so may require clipping before output, but you


// knew that anyway :)





double do_3band(EQSTATE* es, double sample)


{


// Locals





double l,m,h; // Low / Mid / High - Sample Values





// Filter #1 (lowpass)





es-%26gt;f1p0 += (es-%26gt;lf * (sample - es-%26gt;f1p0)) + vsa;


es-%26gt;f1p1 += (es-%26gt;lf * (es-%26gt;f1p0 - es-%26gt;f1p1));


es-%26gt;f1p2 += (es-%26gt;lf * (es-%26gt;f1p1 - es-%26gt;f1p2));


es-%26gt;f1p3 += (es-%26gt;lf * (es-%26gt;f1p2 - es-%26gt;f1p3));





l = es-%26gt;f1p3;





// Filter #2 (highpass)





es-%26gt;f2p0 += (es-%26gt;hf * (sample - es-%26gt;f2p0)) + vsa;


es-%26gt;f2p1 += (es-%26gt;hf * (es-%26gt;f2p0 - es-%26gt;f2p1));


es-%26gt;f2p2 += (es-%26gt;hf * (es-%26gt;f2p1 - es-%26gt;f2p2));


es-%26gt;f2p3 += (es-%26gt;hf * (es-%26gt;f2p2 - es-%26gt;f2p3));





h = es-%26gt;sdm3 - es-%26gt;f2p3;





// Calculate midrange (signal - (low + high))





m = es-%26gt;sdm3 - (h + l);





// Scale, Combine and store





l *= es-%26gt;lg;


m *= es-%26gt;mg;


h *= es-%26gt;hg;





// Shuffle history buffer





es-%26gt;sdm3 = es-%26gt;sdm2;


es-%26gt;sdm2 = es-%26gt;sdm1;


es-%26gt;sdm1 = sample;





// Return result





return(l + m + h);


}





Hope that gives the idea.





Regards, Dan.
Reply:google it
Reply:you can start by learning recursive algorithms. You tell by looking at the repeating and mirrored shapes and lines. Winamp and wmv graphics are recursive graphic algorithms, but i doubt if anyone will throw their code in your face. That stuff takes time and effort.