//---------------------------------------------------------------------------- // // 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 #include #include #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->lg = 1.0; es->mg = 1.0; es->hg = 1.0; // Calculate filter cutoff frequencies es->lf = 2 * sin(M_PI * ((double) lowfreq / (double) mixfreq)); es->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, int sample) { // Locals double l, m, h; // Low / Mid / High - Sample Values // Filter #1 (lowpass) es->f1p0 += (es->lf * ((double) sample - es->f1p0)) + vsa; es->f1p1 += (es->lf * (es->f1p0 - es->f1p1)); es->f1p2 += (es->lf * (es->f1p1 - es->f1p2)); es->f1p3 += (es->lf * (es->f1p2 - es->f1p3)); l = es->f1p3; // Filter #2 (highpass) es->f2p0 += (es->hf * ((double) sample - es->f2p0)) + vsa; es->f2p1 += (es->hf * (es->f2p0 - es->f2p1)); es->f2p2 += (es->hf * (es->f2p1 - es->f2p2)); es->f2p3 += (es->hf * (es->f2p2 - es->f2p3)); h = es->sdm3 - es->f2p3; // Calculate midrange (signal - (low + high)) //m = es->sdm3 - (h + l); // fix from http://www.musicdsp.org/showArchiveComment.php?ArchiveID=236 ? m = sample - (h + l); // Scale, Combine and store l *= es->lg; m *= es->mg; h *= es->hg; // Shuffle history buffer es->sdm3 = es->sdm2; es->sdm2 = es->sdm1; es->sdm1 = sample; // Return result return (int) (l + m + h); }