VST equalizer demo.

[Dell]

Hi.

The CPU usage is much better than the previous version.

I would suggest having a master volume (input or output) to regulate the overall levels.

Cheers!

[tor]

Hi.

The CPU usage is much better than the previous version.

I would suggest having a master volume (input or output) to regulate the overall levels.

Cheers!

That's was not a part of my plan. But I might consider it. Anyone had the time to test it yet? Please report what platform, host and samplerates it has been tested on.

Thanks in advance.
Tor

[tor]

By the way... I posted this under examples. I will upload the osm when I have finished the EQ and or if I get totally stuck with something.

Cheers!

[Dell]

Hi.

If you are planning to offer your equaliser to the public, having a control to regulate the overall level will be necessary.

I tried a quick test using a 192 kHz wav file, and it appears to be working fine. Upon engaging the low-pass filter everything above 20 kHz is rolled off. I used a spectrum analyser that can detect beyond 40 kHz.

Cheers!

[tor]

I actually did that on purpose. I really don't see the point of keeping info over 20k. The number of people that can hear anything up there are so very very few. But I'm going to implement a gain knob

[Dell]

Most humans cannot hear beyond 16 kHz. The ones who can hear 20 kHz, will just keep the filters off.

The difference is noticable using 88.2,96, 176.4, in addition to, 192 kHz files. The majority uses 44.1 and at times, 48 kHz. Under those conditions 20 kHz is already being severed by the limited bandwidth.

Cheers!

[bootsy]

The curve warping near Nyquist is still there. So, whats the oversampling good for?

[MyCo]

I haven't look deeply into all posted OSMs, but in the optimized versions of filtercode2 & 3 from Mo, there is some room. Stage 0 can be completely removed by the well known ABS-ASM hack.
BTW: in the ASMs there are some undeclared variables.

[tor]

Thanks for the feedback guys.

@ Bootsy: I dont know what causes this. The oversampling should help.. but not apparently. Can you point me in the right direction?

@ MyCo: I really have no asm experience or knowledge. Could you please fix these issues this for me? You will be credited

I now start thinking that I have to start learning Octave to wrap my head around the maths in IIR's..

[Mo]

I haven't look deeply into all posted OSMs, but in the optimized versions of filtercode2 & 3 from Mo, there is some room. Stage 0 can be completely removed by the well known ABS-ASM hack.
BTW: in the ASMs there are some undeclared variables.

Seems hardly worth it?

Abs_Compare.osm

(10.28 KiB) Downloaded 109 times

[tor]

Thank you Disco_Steve!

Now I have replaced the difference equation in code to another form and the result is a perfectly stable filter in the low end.

If anyone want to have a look at it or maybe please with sugar optimize it in asm I would be very thankful.

See att's for the VST and the new code.

[tor]

I was a bit quick there.. seems like i have a small amount of fluctuation in the peak-filter's lowest frequencies. But this is so little I will accept it.

At least for now

[trogluddite]

Here's a leaner ASM version of the difference thingy...

DF2 ASM.osm

(945 Bytes) Downloaded 147 times

...see if it nulls when subtracted from your code just to be sure I haven't messed it up!

[tor]

Thank you very much troggie

[trogluddite]

No problem - reducing the code is the easy bit for me.
Getting the right equation and then finding the coefficients to make a filter, phew, then I am very lost

If you would like to learn a little ASM of your own, then this type of code which is just maths is the easiest to start with - you only need to know a handful of opcodes. With these smaller codes, it is simpler to write your own from scratch than to edit the SM generated code - that can get a bit confusing with all the unnecessary lines of code that it makes.