VST equalizer demo.

[aliasant]

Hi tor, sorry to leave you hanging like this - but I'm ill at the moment. I will try and write up a bit of a tutorial on this tomorrow.

-Andrew

Get well m8!

[Andrew J]

Thanks for the kind thoughts guys, I'm definitely on the improve!

tor, take a look at the updated wiki: How To Code An Oversampled Filter. I've also added an example schematic.

I hope I haven't made any dumb mistakes

[stw]

Thanks for the kind thoughts guys, I'm definitely on the improve!

tor, take a look at the updated wiki: How To Code An Oversampled Filter. I've also added an example schematic.

I hope I haven't made any dumb mistakes

Thanks for that very educating update and the well explaining .osms! Very much appreciated!

[aliasant]

Thanks for the kind thoughts guys, I'm definitely on the improve!

tor, take a look at the updated wiki: How To Code An Oversampled Filter. I've also added an example schematic.

I hope I haven't made any dumb mistakes

Thanks for that very educating update and the well explaining .osms! Very much appreciated!

120% Agree!!!!

Thank you Andrew. Finally some light in my mudded.......

[MegaHurtz]

Andrew please spare a minite of your time to explain to me how raising the hf content
over nquist again would not create aliasses.
Becouse there`s only two samples there, I beleive you are wrong.
Also I beleive nobody applying this in the right mind would ever use a digital filter.
Furthermore you cannot construct a Niq sinewave either. only a 0 1 0 -1.
wich will mess up your signal yet again.

[MegaHurtz]

Actually i made a case against it.. Here you can see filtering amounts to a flatter passband.
and less jitter. Albeit a nicely represented sinewave, the frequency by my da is lower (thus an alias).

os vs filth.JPG

[tor]

Good to hear you are getting better Andrew

And thanks for sharing your knowledge. I see the solution is much simpler than I anticipated, and it makes perfect sense when I see it. Learning new stuff every day is great

Again, THANK YOU VERY MUCH!

[MegaHurtz]

It does do a pretty nice job though. the jitter reduction to certain frequency`s is phenomenal.
must be the q im creating wich is a bit off. maybe we should collab on this to get a new design.
keep on doing the Great work!

[bootsy]

Great job Andrew (as always) and thanks for sharing!

[Dell]

It appears the problem with over-sampling is the misunderstanding of over-sampling.

Over sampling enables a given frequency range, less distortion, based on a wider bandwidth. The wider the bandwidth, the better the chance of achieving ultrasonic & infrasonic frequencies with the least amount of distortion.

If one is filtering frequencies severely @ 20,000 kHz and not the ending point of the frequency bandwidth, over-sampling will create a problem for the filtering occurs below the frequency’s bandwidth. The end result is a dip @ 20 kHz and rise roughly around 22 – 23 kHz.






In order to remedy such a peak, in the response, one would need to filter at the starting point of the frequency bandwidth.

The file used is pink noise sampled @ 96 kHz. This is equivalent to a frequency bandwidth of 48 kHz, which is common in the lowest grade of analogue hardware.

Cheers!

[MegaHurtz]

it`s very simple to mimic that passband response without oversampling.
And guess what it sounds quite well. So you could state gentle attenuation reduces aliasses along the line.
and does a proper yob at keeping away ringing, and converts any sine to some sine. becouse it
cant be amplitudes other than a sine becouse of the loudness.
The big question is what level to be per frequency?

EDIT to me atleast

[MegaHurtz]

Attenuation 6x no oversampling Vs Dry.
The PCM.

attband2.rar

The FFT.

attband.JPG

[MegaHurtz]

In order to remedy such a peak, in the response, one would need to filter at the starting point of the frequency bandwidth.

The file used is pink noise sampled @ 96 kHz. This is equivalent to a frequency bandwidth of 48 kHz, which is common in the lowest grade of analogue hardware.

Cheers!

aha, i see.. The trick would be getting it`s frequency at 44khz then.

[Dell]

Right!

The track is sampled/recorded pink noise @ 96 kHz. The playback is set to 44.1 kHz. This is why the frequency is chopped off beyond 22 kHz.

The rise beyond 20 kHz is based the low-pass filter’s starting point @ 20 kHz. Since the file is a higher sample rate, its frequency bandwidth extends beyond 20 kHz.

Having a low-pass filter offering a starting point of 20 kHz may be fine for a 44.1 kHz file. However, for a 96 kHz file, 20 kHz is not high enough to address frequencies beyond 20 kHz upon low-pass filtering.

Bear in mind if the file used were 44.1 kHz, there would be no rise in dB/SPL above 20 kHz for, 44.1 (22,050) is the frequency bandwidth limit.

I’ve over-sampled a few equalisers for my sound system however there are no low-pass filtering within the chain.

I will try to post some frequency graphs based of audio files based on various sample rates by the middle of the week.

Cheers!

[MegaHurtz]

Also you could yust use an interpolation on it. Maybe even keeping
the sampled audio at a 44k pitch, and upsample really nice.
But im really sold to the attenuation thing, got one that sounds really natural
but i think it could be a little better? Now it`s a liniar attenuation.