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Latest musicdsp comments
Added to : 3 Band Equaliser
Date : 25/06/09
By : bechar[ DOT ]ce[ AT ]ygamil[ DOT ]com
Comment :
Hi Great Stuff,
how to create 6 band equalizer, is any algorithm for 6 band same like 3 band, please help me if any one
thanks in advance
Added to : Quick & Dirty Sine
Date : 22/06/09
By : by toast[ AT ]somewhereyoucantfind[ DOT ]com
Comment :
Juuso,
Absolutely!
Toast
Added to : Band Limited waveforms my way
Date : 22/06/09
By : antto [AT] mail [DOT] bg
Comment :
excuse me, there is a typo
amp = scale / h;
Added to : One pole filter, LP and HP
Date : 18/06/09
By : bob[ AT ]yahoob[ DOT ]com
Comment :
Or...
out += a0 * (in - out);
:)
Added to : Beat Detector Class
Date : 01/06/09
By : kwolff[ AT ]intra-team[ DOT ]de
Comment :
Can you tell me how to implement the variable bpm, to get the beats per minute?
Thanx,
Kris
Added to : Simple Tilt equalizer
Date : 29/05/09
By : neolit123[ AT ]gmail[ DOT ]com
Comment :
correction:
(ex: +6db -> hp; -6db -> lp)
Added to : 4th order Linkwitz-Riley filters
Date : 29/05/09
By : neolit123[ AT ]gmail[ DOT ]com
Comment :
LR2 with DFII:
//------------------------------
// LR2
// fc -> cutoff frequency
// pi -> 3.14285714285714
// srate -> sample rate
//------------------------------
fpi = pi*fc;
wc = 2*fpi;
wc2 = wc*wc;
wc22 = 2*wc2;
k = wc/tan(fpi/srate);
k2 = k*k;
k22 = 2*k2;
wck2 = 2*wc*k;
tmpk = (k2+wc2+wck2);
//b shared
b1 = (-k22+wc22)/tmpk;
b2 = (-wck2+k2+wc2)/tmpk;
//---------------
// low-pass
//---------------
a0_lp = (wc2)/tmpk;
a1_lp = (wc22)/tmpk;
a2_lp = (wc2)/tmpk;
//----------------
// high-pass
//----------------
a0_hp = (k2)/tmpk;
a1_hp = (-k22)/tmpk;
a2_hp = (k2)/tmpk;
//=========================
// sample loop, in -> input
//=========================
//---lp
lp_out = a0_lp*in + lp_xm0;
lp_xm0 = a1_lp*in - b1*lp_out + lp_xm1;
lp_xm1 = a2_lp*in - b2*lp_out;
//---hp
hp_out = a0_hp*in + hp_xm0;
hp_xm0 = a1_hp*in - b1*hp_out + hp_xm1;
hp_xm1 = a2_hp*in - b2*hp_out;
// the two are with 180 degrees phase shift,
// so you need to invert the phase of one.
out = lp_out + hp_out*(-1);
//result is allpass at Fc
Added to : 3 Band Equaliser
Date : 24/05/09
By : bob[ AT ]yahoob[ DOT ]com
Comment :
What problems were you getting? Doesn't removing the delay cause phase problems?
Added to : 3 Band Equaliser
Date : 23/05/09
By : philip[ AT ]blastbay[ DOT ]com
Comment :
This is a great little filter, I am using it in an application but when I first started playing with it I noticed some problems. The mid range didn't seem to be calculated properly, a friend of mine who knows more about dsp than I do took a quick look at it and suggested the following change:
m = es->sdm3 - (h + l);
Should be:
m = sample - (h + l);
I've tested it with this small fix and everything works perfectly now. Just thought I'd bring this to your attention... Thanks for a great code snippet!
Added to : 3 Band Equaliser
Date : 22/05/09
By : bob[ AT ]yahoob[ DOT ]com
Comment :
For more bands, you could take the low-pass and repeat the process on that.
Added to : Polynominal Waveshaper
Date : 19/05/09
By : flavien[ DOT ]volken[ AT ]gmail[ DOT ]com
Comment :
According to another post, the tube is simply a non linear function, for example a tan(x). Actually by saturating any signal you will get harmonics (any but a pure square which cannot be more saturated of couse...). As tan(x) is not linear, you should get harmonics… that's all. Now if you want to pass only the high frequencies,just split the signal into 2 frequencies using a lowpass vs highpass = signal - lowpass and process the frequencies you want to.
Added to : Fast in-place Walsh-Hadamard Transform
Date : 19/05/09
By : bob[ AT ]yahoob[ DOT ]com
Comment :
If you can't convert a few lines of 'c' into to another language, then you must be mad, or lazy, or both.
Added to : Envelope detector
Date : 19/05/09
By : bob[ AT ]yahoob[ DOT ]com
Comment :
Should use "fabsf" really. :)
Added to : 3 Band Equaliser
Date : 05/05/09
By : vhain6512[ AT ]gmail[ DOT ]com
Comment :
How can I expand this 3 Band EQ into X Band EQ..?!
Anybody answer me, or email me..
Added to : Fast Downsampling With Antialiasing
Date : 02/05/09
By : mumart[ AT ]gmail[ DOT ]com
Comment :
filter_state is the previous input sample * 0.25, so zero is a good starting value for a non-periodic waveform.
Added to : Reverberation techniques
Date : 24/04/09
By : krs[ AT ]ms15[ DOT ]hinet[ DOT ]net
Comment :
matlab reverb ..thanks
Added to : Notch filter
Date : 24/04/09
By : efishocean[ AT ]gmail[ DOT ]com
Comment :
does this code work with float output?
I really need a notch filter, i will try the code.
Added to : Simple biquad filter from apple.com
Date : 20/04/09
By : neolit123[ AT ]gmail[ DOT ]com
Comment :
i've missed a line:
===========================
mem[4*i+3] = output;
//>>> insert here
input = output;
//>>> insert here
);
//----sample loop
===========================
lubomir
Added to : Stereo Enhancer
Date : 17/04/09
By : claytonhotson[ AT ]gmail[ DOT ]com
Comment :
You mean to use MonoSign variable somewhere - as in:
#define StereoEnhanca(SamplL,SamplR,MonoSign, \
stereo,WideCoeff ) \
MonoSign = (SamplL + SamplR)/2.0; \
stereo = SamplL - Sampl1R; \
stereo = stereo * WideCoeff; \
SamplL = MonoSign + stereo; // R+Stereo = L
SamplR = MonoSign - stereo; // L-Stereo = R
Or some variation?
Clayton
Added to : Waveshaper
Date : 16/04/09
By : a[ AT ]b[ DOT ]com
Comment :
This is one of my faves...
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