VSTFX: La Ride

[wlangfor@uoguelph.ca]

So, a Gain Rider is a useful thing that allows You to pick a threshold of volume.

The volume then tries to average between the min and max that You set. This is really handy for mastering an entire track and it is very oldschool in nature.

I really don't like brickwall limiters, this method is far less destructive.

LA Rider v073 public.fsm

Will update Presets and a few settings to make it not just an example but a finalized plug

(90.43 KiB) Downloaded 1264 times

[wlangfor@uoguelph.ca]

So, I like this prototype so far but I will be adding some settings. I feel that the riding could be made more finite if there were an intensity setting.

I could achieve this by making a second mono 2 float for the second iteration as an option. This would whittle down or bolster by percentile a bit more effectively in light of the first pass already have taken place.

I don't know the name of the setting, maybe coarse/precise or something like that.

[RJHollins]

a 'Strength' setting ?

[wlangfor@uoguelph.ca]

Well I guess it comes down to sampling, yes. Anyways, here's the new version:


Here's the changes I made to the look:

[wlangfor@uoguelph.ca]

Here's a selection of various fixes to improve the sound, look and the math is a bit more accurate.

Now all that's left is to make sure the presets are numbered correctly and I also wanted to add some gauge lines and an optional light up horizontal line average and peak detector, there will also be text for that average and peak. So, all in all this is a great mastering solution for people Who gave up on fancy products.

Once You figure where things should be, it's not rocket science and something like this sounds like hardware. In fact, I don't think I could tell the difference between this and hardware.

[wlangfor@uoguelph.ca]

OK, so If You'd like a preview, here's the next version which works fine - but it's so much better looking.

Now I guess i need to perfect the math, make a simple horizontal line VU meter that looks like analog/digital combo to go behind My gauges and then I'll make an average peak, average lesser dB as well as average high and low. These things are really important for professionals to have.

Here's the Images (click to see larger in new window):




And here's the schematic:


And now even humbly I can say that this thing is one of the best VST plugs ever made and it sounds so sweet. It'll be even better with ordered plugins and 50 presets too. I'll literally match like 50 of the greatest songs by michael jackson etc and the weekend so that people can reference with class.

BTW, I'll make sure to add all knobs I just made from scratch to the knobman gallery

EDIT: So I wanted to add, this is no mere final version or anything like that. I have actually been finding ways around some of the bugs found in FL Studio 12 without experiencing issues by a continually redrawing prim. To counter this I have set the Constant redraw rate to 5 ticks a second, Furthermore I've made it so that if a knob or slider etc is in action, then the global ticker turns off.

This is big news for producing plugins for FL Studio 12 and 20. Furthermore, especially in the case of visual parametric EQ's this will prevent some bugs.

Regards and I will post tomorrow with what i have come up with.
[new dl on last post]

[lalalandsynth]

You might want to change that hand pointer to a regular one . I always do that , preference I guess.

Quick look , it seems that the gain manipulation is done with greens and a 100 tick , there is a response time problem with that , and I think it would vary . Also , faster then realtime rendering would not sound the same and be random basically.

[wlangfor@uoguelph.ca]

You might want to change that hand pointer to a regular one . I always do that , preference I guess.

Quick look , it seems that the gain manipulation is done with greens and a 100 tick , there is a response time problem with that , and I think it would vary . Also , faster then realtime rendering would not sound the same and be random basically.

Not really, because there is a tick 100 and averages used, the intermittent variance just serves for the averaging to smooth the signal. A way of using Math to be a peak limiter in a minute sense.

And yes to the pointer, but to do that I would have to use Ruby. It's an aesthetic thing. But then again, maybe I could do the same as MvdLee did with the popup modal I use for credits and info... Hmm..

[trogluddite]

Quick look , it seems that the gain manipulation is done with greens and a 100 tick , there is a response time problem with that , and I think it would vary . Also , faster then realtime rendering would not sound the same and be random basically.

Not really, because there is a tick 100 and averages used, the intermittent variance just serves for the averaging to smooth the signal. A way of using Math to be a peak limiter in a minute sense.

Lalalandsynth is correct - the method is fundamentally flawed mathematically. Without including the contribution of the samples which you skip, an accurate reading of the audio signal's power isn't possible.

Essentially it's resampling the audio at a lower sample rate, so the Nyquist Rule applies - frequencies higher than half the sampling rate cannot be accurately represented (in this case: f > ~50Hz). Any frequencies higher than this will become aliasing noise, and with such a low sampling rate, much of that is going to be at such low frequencies that it will still affect the output averaging (which is effectively a low-pass filter.) During a non real-time render, the down-sampling ratio would effectively be higher, compounding the problem.

Another way to look at it is this. If we assume for a moment that the ticker really does run at exactly 100Hz, what would happen to a waveform also of 100Hz (or a harmonic thereof)? It would depend on the phase relationship between the two - we might capture a series of peaks, a series of zero crossings, or somewhere in between - completely at random depending when playback started (and ticker jitter). As whole series of samples would be biased by the phase relationship, post-measurement averaging won't compensate for this.

Such errors might average out for a sufficiently broadband signal, but (un)fortunately, music relies on discrete relationships between frequencies; so in practice, this is highly unlikely. Hence the averaging must always be done over every single audio sample BEFORE sampling the resulting average, whatever the output sample rate - averaging AFTER sampling simply won't give the same mathematical results.

This can be seen clearly by feeding an oscillator into the plugin and comparing input to output on a scope at different frequencies - you can clearly see the output amplitude wavering, even though the input has a fixed amplitude. It also won't help that you're modulating the output level with a green signal - every tick is effectively producing a discontinuity in the waveform, so you're adding noise, too.

And now even humbly I can say that this thing is one of the best VST plugs ever made

[NB: Personal opinion, not a moderation message.]
There is nothing "humble" about such a claim. This is a forum for the kind of people who will run your plugins through a scope and frequency analyser, stick every kind of signal they can imagine through them, and have a good old poke around inside. Hence, posting your plugins for objective criticism is a good route to learning our vocation better. But your constant hyperbolic marketing up-talk is becoming very tiresome and is not relevant here - please save it for your personal shop-front website.

[wlangfor@uoguelph.ca]

lol, stuffy trog.

But anyways, moving on; enevlopes do not run nearly as fast; People rely on them everyday, but alas they are limited by the same. And is it also in Your realization that regardless of some contrast that there will be an averaging occurring?

It's always dawned on Me that envelopes are slow, are worse than spectral limiters in their destructive appliance and are a blunt and weighty axe while a percentage however intermittent is a finely crafted chisel.

It's stuffy comments like that that people post less. Plainly, envelopes are destructive and You've been confused in My posts before about z-index of microphones and polar patterns etc. And then You get stuffy and sharp about terminology being a pedantrist. Which is all well and good. I mean, free speech.

But given the fact that ubiquitous envelopes which mind You have no respect for highs or lows and cause latency in any application how can one justify not trying to leave the "mud" behind. Take a step back, think about it and realize that the reason why this is one of the best plugs is that it is the result of teamwork.

Teamwork of Yours Mister Trog, Mr. Vicanek's MvdLee's, Mine, Infuzion's, Cytosonic's, Myco's, Spogg's, Nubeat7's Wassaka's and various others. It's the reason plainly why such a good plugin could be manufactured. So that's all I have to say to your banter.

It all slides off of Me. All lemons and lemons aren't so bad.

But here's a new version and images for You to use for free lol:







Download Schematic here:
[download newest version on last post]

Download Knobs Here:
https://board.leattol.com/viewtopic.php?f=2&t=2

You should really have faith in your hard work and what it can achieve . It's why I'm a big fan.