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Latest musicdsp comments
Added to : Beat Detector Class
Date : 05/03/10
By : ata_n[ AT ]hotmail[ DOT ]com
Comment :
Im having a hard time setting the comparison level. My audio data is signed 16 bit integers, so I have set the levels at (0.3*32768) and (0.15*32768)...
ive tried different levels, nothing responds correctly..
any ideas anyone?
Thanks,
Ata
Added to : Butterworth
Date : 05/03/10
By : jdmcox[ AT ]jdmcox[ DOT ]com
Comment :
State2 = B4*Stage1 + A4/A3*Output + State2;
should read
State2 = B4*Stage1 + A4/A3*Output + State3;
Added to : Peak/Notch filter
Date : 01/03/10
By : slo77y (at) yahoo DOT de
Comment :
this code sounds bitcrushed like hell translated to c++, any suggestions ?
float pi = 3.141592654;
float r = dQFactor*0.99609375;
float f = cos(pi*iFreq);
float a0 = (1-r) * sqrt ( r * ( r-4 * ( f * f ) + 2 ) + 1 );
float b1 = 2 * f * r;
float b2 = - ( r * r );
float outp = 0.0, outp1 = 0.0, outp2 = 0.0;
for (i = 0; i < iSamples; i++)
{
float inp = fInput[i];
outp = a0 * inp + b1 * outp1 + b2 * outp2 + p4;
outp2 = outp1;
outp1 = outp;
fOutput[i] = (inp-outp); //notch
}
Added to : Fast Float Random Numbers
Date : 27/02/10
By : aojudy[ AT ]gmail[ DOT ]com
Comment :
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Added to : Band Limited PWM Generator
Date : 16/02/10
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Added to : Band Limited PWM Generator
Date : 16/02/10
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Added to : Band Limited PWM Generator
Date : 16/02/10
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Added to : Band Limited PWM Generator
Date : 15/02/10
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Added to : Band Limited PWM Generator
Date : 12/02/10
By : kasper756[ AT ]unique-papers[ DOT ]com
Comment :
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Added to : Band Limited PWM Generator
Date : 04/02/10
By : penrose[ AT ]alumni[ DOT ]princeton[ DOT ]edu
Comment :
The implementation certainly produces aliased waveforms -- they are glaring on a spectrogram at -60dB and faint at -30dB. But it is a remarkably efficient algorithm. The aliasing can be mitigated somewhat by using a smaller feedback coefficient.
Added to : Fast Float Random Numbers
Date : 24/01/10
By : bob[ AT ]yahoob[ DOT ]com
Comment :
I don't understand Judahmenter's comment about 3 not limiting the amplitude. As it stands it returns a value -1 to 1, so just multiply by your 'amp' value.
This turns into a handy 0-1 random number if you take off the sign bit:
(float)(RandSeed & 0x7FFFFFFF) * 4.6566129e-010f;
Added to : Polyphase Filters
Date : 21/01/10
By : bobby
Comment :
Is the cutoff at 20kHz? What sample rate are these coefficients for? How would I calculate suitable coefficients for arbitrary sample rates?
Added to : Direct Form II biquad
Date : 13/01/10
By : ano[ AT ]nymous[ DOT ]com
Comment :
true, this structure is faster. but it is also (even) more sensitive to coefficients changes, so it becomes unstable quite fast compaerd to the DF I form. I'd really like to know if there's a way to change coefficients and at the same time time changing the history of the filter for avoiding unstability.
Added to : Moog VCF, variation 2
Date : 08/01/10
By : http://www[ DOT ]myspace[ DOT ]com/paradoxuncreated
Comment :
You have to subract each filter, from the input in the cascade.
Check also the Karlsen filters, which I made a few years ago, when going through this stage in DSP.
Added to : Simple peak follower
Date : 07/01/10
By : marianomazzeo[ AT ]hotmail[ DOT ]com
Comment :
Totally newby question. What apis are you using for this code?
Added to : Fast Float Random Numbers
Date : 29/12/09
By : earlevel [] earlevel [] com
Comment :
It should be noted in the code that for method #3, you must initialize the seed to non-zero before using it.
Added to : Fast Float Random Numbers
Date : 20/11/09
By : judahmenter at gee mail dot com
Comment :
There is no doubt that implementation 3 is fast, but the problem I had with it is that there's no obvious way to limit the amplitude of the generated signal.
So instead I tried implementation 2 and ran into a different problem. The code is written such that it assumes that RAND_MAX is equal to 0x7FFF, which was not true on my system (it was 0x7FFFFFFF). Fortunately, this was easy to fix. I simply removed the >> 16 and worked fine for me. My final implementation was:
return (float)(RandSeed = RandSeed * 214013L + 2531011L) / 0x7FFFFFFF * 2.0f * amp - amp;
where "amp" is the desired amplitude.
Added to : DIRAC - Free C/C++ Library for Time and Pitch Manipulation of Audio Based on Time-Frequency Transforms
Date : 11/11/09
By : tizwah[ AT ]gmail[ DOT ]com
Comment :
...there is source code included, see http://www.dspdimension.com/download/
Added to : Moog VCF, variation 2
Date : 10/11/09
By : bardiclug[ AT ]gmail[ DOT ]com
Comment :
YAND (Yet Another Newbie Developer) here -
This filter sounds good, and with the addition of a 2nd harmonic waveshaper in the feedback loop, it sounds VERY good.
I was hoping I could make it into a HP filter through the normal return in-out4 - but that strategy doesn't work for this method. I'm afraid I'm at a loss as to what to try next - anyone have a suggestion?
--Coz
Added to : Moog VCF, variation 1
Date : 23/10/09
By : thisguy[ AT ]nowhere[ DOT ]com
Comment :
The domain seems to be 0-nyquest (samplerate/2.0), but the range is 0-1
A better way to get smoother non-linear mapping of frequency would be this:
(give you a range of 20Hz to 20kHz)
freqinhz = 20.f * 1000.f ^ range;
then
frequency = freqinhz * (1.f/(samplerate/2.0f));
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