VST equalizer demo.

[tor]

I'm sorry I wrote Q.. I meant Slope witch is the best term in a shelving filter. It is not locked to 6, 12, 18, 24, 48 dB per octave as I understand you claim.

Correct me if I misunderstood

[Dell]

Once you alter the shelf to create a steep slope, you are asking it to deliver a peak response. Going from - 1dB @ 20 Hertz to - 27 dB @ 10 Hertz is a steep roll-off which is common with peaks.

If you want to tackle the problem, create your own spectrum analyser that can measure 0.5 Hz with Synthmaker and move away from the VST Analyser.

I only use VST Analyser to show the frequency response from a 0 dB perspective. Upon viewing my Blog, you will notice all real-time measurements are conducted using other means of test equipment.

Here are the Narrow Q tests…

http://www.youtube.com/watch?v=5LSo4efZTRc

http://www.youtube.com/watch?v=W7pOiTbTpW0


If you want to get rid of the fluctuations, follow the dB per octave rule and don’t try to create your own customised roll-off slopes that will add more artefacts in the design.

I understand you want everything to be designed using your parts. However, there are many developers that have already created filters that will not offer any artefacts. Here is one of them

viewtopic.php?f=10&t=5823



Bare in mind I do not make clean plug-ins. All my designs are supposed to add their own tonal characteristics upon usage… Just like hardware equipment.

Cheers!

[tor]

Any way.

The biquads I use is the same as the ones you link to. And mine is perfectly stable too. But... when I cascade 4 of then I get this fluctuation. I do NOT think this has any thing to do with the VSTP.A..

I think I need some more in depth competence here. It is not magic... it is DSP technology and I'm quite sure there is some one in here who can help me out.

I repeat my problem atm:

Fluctuation in the lowest frequencies on the highpass filter when measuring in VSTpluginanalyzer. (Curve jumping increasingly from about 100hz down to 0hz. Significant below 25hz.)

Bootsy talked about some warping in the high frequencies. I have not been able to understand/ identify what he means.

[tor]

Another quest for mr. Mo

[Dell]

You may need to start building various types of equalisers to have a better understanding how they work. Nothing beats practice.

Equalisers are frequency driven. If you alter one frequency the frequencies surrounding the alterd frequency are going to react. That is how all equalisers respond whether it is a DSP, transistor or valve equaliser.

Your slope is too steep and is peaking.

The Q is not wide enough to control the frequencies lower than 10 Hz and they are reacting to the sine wave signal being fed from VST Analyser.

Since there is no way to turn off the signal and adjust the measurements in VST anaylser you are getting a bouncing signal. The rise is due the frequency below 10 cycles returning back to 0 dB in which VST Analyser cannot display for it is limited to 5.3 cycles.

Slopes continue to decline per octave until it reaches DC or 0 cycles.

Peaks will not follow the slope method unless the Q is wide enough to cover various octaves, are set at a very low frequency and, offers enough dB loss to mimic the response of a slope.


MyCo offers an OSM with optimise filters which do not offer any artefacts in the link I offered.

Cheers!

[Disco_Steve]

The problem is math error in the tail of the response due to the resolution of the floats used.
Low frequency filters have long tails and are thus more susceptible to these problems, you need to use a different form of biquad, (there are about ten) it's to do with the order in which the coefficients are added together, I canny remember which one is best when chaining filters (sorry) but the is a particular one that has been shown to give the best response.

The asymmetry in the response in the high frequencies is due to the nyquist limit. It's just a fact of the filters used, infect all IIR's suffer from the effect. The solution is to use higher sample rates. As far as I can tell this is the only reason why higher sample rates sound better, because the filters on the audio sound more symmetric which sounds more analogue because analogue filters don't have the same problem.

The problem with higher sample rates is that it pushes this low frequencies cause math problems error up to higher frequencies as well, so you need to have a greater bit depth in the maths. 24bit mentesa floats should be just enough if your cairful.

Helpful?

Oww just remembered, the best place to see all the different forms of biquad is in the help file in matlab (I think it's online) search freqz and it should come up. I think it also talks through the advantage and disadvantage. You might have to sacrifice some CPU though, as as far as I remember th best one for chaining isn't best for CPU use

Steve
Steve

[Disco_Steve]

Reading mo closely.

The best way to see the frequency and phase response of a filter is to get it's impulse response and fft that.

This will get rid of any widowing problems that is currently causing the wobbling at low frequencies you are experiencing, as some times the wave is in a different place and the square window is causing different results.

That should be more helpful

Steve

[tor]

Now that was very helpful!

Very nice explained.

THANK YOU!

[tor]

Ok I now have two small code chunks that I would love to get optimized in asm. Any one up to the task?

Please.... with sugar and candy

@ Mo?
@ Anyone?

[tor]

@ Disco Steve:

I know about direct form 1 and 2 for the biquads. I will investigate more on this

[Andrew J]

The problem with higher sample rates is that it pushes this low frequencies cause math problems error up to higher frequencies as well, so you need to have a greater bit depth in the maths. 24bit mentesa floats should be just enough if your cairful

Note also that there are techniques for rearranging higher order filters as a series of second order filters for improved performance with single precision floats. Check Matlab (or Octave) documentation for zp2sos.

This might be a bit tricky to implement in green code, but you can get around it by generating tables of coefficients offline. Though I should warn that you'll sometimes get discontinuities in the values of one or more coefficients - and this introduces some risk if you're interpolating the coefficients.

[Mo]

Quick optimizing..

Here you are:

[tor]

Quick optimizing..

Here you are:

You're tha man!

Thank you again

Andrew and Disco: I will see what I can manage to do with this after some more reading. Thanks a bunch for all replies

[tor]

Hey... so I finally managed to finish the optimizing process. There are still some tweaking to do. But I'm getting there soon. The biggest issue atm is the fluctuation problem in the low end I think. Have not had the time to look a the different biquad forms. I have just one peak filter in this test EQ. The finished thing will have more. I added 2*oversampling and a noise knob for eliminating denormals and otherwise creative use. The CPU consumption is very acceptable now.

Please test it.

[tor]

Ah... I forgot to mention. It should now also work correct in all samplerates.