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Latest musicdsp comments
Added to : Fast Float Random Numbers
Date : 26/07/10
By : mengchuanbond[ AT ]gmail[ DOT ]com
Comment :
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Added to : Gaussian White noise
Date : 17/07/10
By : foo[ AT ]bar[ DOT ]de
Comment :
@nick: Way to late, but y2 will always be initialized as in the first run "pass" is 0 (i.e. false). The C# compiler just can't prove it.
Added to : Karlsen Fast Ladder
Date : 17/07/10
By : checkpageforemail[ AT ]checkpageforemail[ DOT ]com
Comment :
I also did a 9th order gaussian filter (minimal phase), using only 5 orders, for my limiter, which is released under the GPL LICENCE. http://www.paradoxuncreated.com
Added to : Lookahead Limiter
Date : 17/07/10
By : checkpageforemail[ AT ]checkpageforemail[ DOT ]com
Comment :
There is a very sophisticated GPL open-source limiter, over at http://www.paradoxuncreated.com.
Added to : (Allmost) Ready-to-use oscillators
Date : 14/07/10
By : kathygray[ AT ]mail333[ DOT ]com
Comment :
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Added to : Biquad, Butterworth, Chebyshev N-order, M-channel optimized filters
Date : 18/06/10
By : LSun[ AT ]verizon[ DOT ]net
Comment :
These codes are just what I am looking for. Too bad they are incomplete as posted here. Could someone direct me to a complete version?
Added to : Tone detection with Goertzel (x86 ASM)
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Added to : Tone detection with Goertzel (x86 ASM)
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Added to : Tone detection with Goertzel (x86 ASM)
Date : 12/06/10
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Added to : Tone detection with Goertzel (x86 ASM)
Date : 09/06/10
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Added to : Tone detection with Goertzel (x86 ASM)
Date : 06/06/10
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Added to : 3 Band Equaliser
Date : 27/05/10
By : marian_sabianaa[ AT ]hotmail[ DOT ]com
Comment :
hello! thanks for your code!!
i tried to use the code in my project of guitar distortions in real time (in C) and i could'nt, i'm in linux using jack audio server, and it starts to have x-runs everytime i turn on the equalizer. do you any idea of how solving this? (from 5 to 50 milliseconds o x-runs)
i was thinking of coding it in assembler but i don't know if that would be the solution.
excuse me for my english, i'm from argentina and it's been a while since i last wrote in this language!
thanks in advance, hoping to see any answer!
mariano
Added to : Tone detection with Goertzel (x86 ASM)
Date : 27/05/10
By : jennifermartinez[ AT ]mail15[ DOT ]com
Comment :
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Added to : Delphi Class implementation of the RBJ filters
Date : 25/05/10
By : crlab[ AT ]gatech[ DOT ]edu
Comment :
I have created an open source graphical project that allows you to interactively apply and adjust these filters:
http://www.cabiatl.com/CABI/biquad-filters/
I also include a terminal application that compiles with Delphi, Lazarus and free pascal. Thanks for this useful code.
Added to : 3 Band Equaliser
Date : 06/05/10
By : wijesena[ AT ]gmail[ DOT ]com
Comment :
How to extend this to 6 band equalizer?
Added to : Compressor
Date : 28/04/10
By : pak[ DOT ]nine[ AT ]gmail[ DOT ]com
Comment :
This is a useful reference, however the RMS calculations are pretty dodgy. Firstly there is a bug where is calculates the number of samples to use:
int sr, // sample rate (smp/sec)
...
double twnd, // window time (ms)
...
// samples count in lookahead window
int nrms = (int) (sr * twnd);
The units are mixed when calculating the number of samples in the RMS window. The window time needs to be converted to seconds before multiplying by the sample rate.
As others have mentioned the RMS calculation is also really expensive, and in my tests I found it was pretty innacurate unless you use a LOT of samples (you basically need a (sample rate)/2 window of samples in your RMS calculation to accurately measure the power of all frequencies).
I ended up using the 1 pole low pass filter approach suggested here, and it is a good cheap approximation of power. I did, however, end up mulitplying it by root(2) (the RMS of a sine wave, which seemed like a reasonable normalisation factor) in order to get it between 0 and 1, which is a more useful range.
Another slightly more accurate way to caculate the RMS without iterating over and entire window for each sample is to keep a running total of the squared sums of samples.
for( i = 0; i < NumSamples; ++i )
{
NewSample = Sample[i];
OldSample = Sample[i - RMSWindowSize];
SquaredSum = SquaredSum + NewSample * NewSample;
SquaredSum = SquaredSum - OldSample * OldSample;
RMS = sqrt( SquaredSum / RMSWindowSize );
// etc...
}
Calculating the power in the signal is definately the awkward part of this DSP!
Added to : Millimeter to DB (faders...)
Date : 11/03/10
By : wouter[ DOT ]hisschemoller[ AT ]gmail[ DOT ]com
Comment :
Flash ActionScript translation:
/**
* Maps normalized value between 0 and 1 to decibel from -200 to 10.
* @param normalizedValue: Value between 0 and 1.
* @return Number: Value in decibel from -200 to 10.
*/
public function normalizedToDecibel(value : Number) : Number
{
value = (1 - value) * 100;
if(value <= 0.0) var db : Number = 10.0;
else if(value < 48.0) db = 10.0 - 5.0 / 12.0 * value;
else if(value < 84.0) db = -10.0 - 10.0 / 12.0 * (value - 48.0);
else if(value < 96.0) db = -40.0 - 20.0 / 12.0 * (value - 84.0);
else if(value < 100.0) db = -60.0 - 35.0 * (value - 96.0);
else db = -200.0;
return db;
}
/**
* Maps decibel from -200 to 10 to normalized value between 0 and 1.
* @param decibel: Value in decibel from -200 to 10.
* @return Number:
*/
public function decibelToNormalized(decibel : Number) : Number
{
if(decibel >= 10.0) var normalizedValue : Number = 0.0;
else if (decibel > -10.0) normalizedValue = -12.0 / 5.0 * (decibel - 10.0);
else if (decibel > -40.0) normalizedValue = 48.0 - 12.0 / 10.0 * (decibel + 10.0);
else if (decibel > -60.0) normalizedValue = 84.0 - 12.0 / 20.0 * (decibel + 40.0);
else if (decibel > -200.0) normalizedValue = 96.0 - 1.0 / 35.0 * (decibel + 60.0);
else normalizedValue = 100.0;
return (100.0 - normalizedValue) / 100.0;
}
Added to : Denormalization preventer
Date : 10/03/10
By : antiprosynthesis[ AT ]gmail[ DOT ]com
Comment :
You could also add input noise and assure output samples are reset to 0 if they're below a certain treshold, slightly higher than your noise volume. That ensures hosts can do proper tail detection and it's cheap.
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;
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