DSP Robotics Support

I don't know that plugin but I made a self-adjusting volume control to avoid clipping. Is that anywhere close?

Thk martin for reply

Martin I have only audio distorted as out of autovolume, anyway based on what I see your project is targeted to down the volume upon peak, very nice and good, and verynear of what I need.

Anyway what I trying to do is a more complex I think, I need automatic volume control for act as soft leveler.
Imagine I want do make compliant the audio to EBU R 128, can be any other normalization in this case I choose EBU R128, where max output program should be -23lufs with tolerance of +- 1LU

So the idea is create a limiter, can be adjusted to any STANDARD, like EBU R128, so I can adjust audio program in real time, with less possible noticiable audio artifacts.
The volume should go up and down all the time in a soft way for prevent at maximum audio artifacts, but need recover the volume if the audio material incoming is a lot dow under -23 and volume of audio material must go down when is over of target volume, in this case -23LUFS
Probably what I need is a very faster compressor/lmiter/acg lol?


BS1770 (rev.3), EBU R128, ATSC A/85 and many more are similar standard where the BS1770 (rev.3), EBU R128 are adopted now worldwide.
The audio industry already adopt lufs as metering and very soon LUFST will be what we talk, not any other meter.
I really suggest guys start to talk in LUFS as reference for all.
Broadcast industry already do, now music industry is following.
The new real 0db for audio will be -23lufs, already is for broadcast and music industry is moving faster in this new direction.

For audio industry take some time, is very complex jump form the current volumes to -23lufs but audio industy is already moving, really not have sense master clipping to 0db, and then your music, is LIMITED AND RECRUNCHED due limitation on TVS, Radio Stations, IOS devices, when audio check, Spotify, etc.

What I try to do is a audio NORMALIZER or can be called limiter too lol, on radio stations audio limiters do this in really very good way, ORBAN processors , some commercial software, etc can do this in real-time.
My problem is I think the math involved locate to me really out of this.

The RMS limiter from you martin do super good job, is very near to do what I want achieve, but I listen very noticiable audio artifacts, distortion, don't know if can be improved and for sure if can work with LUFS instead dbs.

Finally, again I repeat, I think will be a good practice start to talk in LUFS and LU
Very nice job form Youlean here:
viewtopic.php?f=3&t=3018
Act as compartator, just enter a stereo signal to both inputs and you have a real INTEGRATED LUFS meter super accurate.
Sorry this not is directly related to audio limiter, but very nice project related to LUFS

Mmmm I love to see your autovolume working with adjustable target in lufs instead to peak, and with some autorecover logic

http://www.hornetplugins.com/hornet-aut ... mage-48812
Wow, but apparently use compression, I think use compression is needed , a mix of gain reduction with compression and limiting is the way if want achieve nice resulst with lower audio artifacts events.

This is the plugon I was referring to earlier:
http://www.hornetplugins.com/plugins/hornet-vu-meter/

It does not touch the envelope ... only GAIN. We have great compressor/limiters for that job.

Also ... there is a very nice JS coded utility for REAPER that uses the EBU R128 [option] to balance gain staging.

I wanted to convert the JS code to a FS [VST] plugin ... but my skills are not ready

This is like martin tool.
But I cant make the martin tool work, only have distorted sound.
And will be great the target of gain can be adjusted, target choose, instead to fixed peak at 0db.

Here is a version with settable limit. When the input exceeds the limit, the volume control will fall back to a lower level. There may be a distortion for a fraction of a second. You would have to stay in stream domain (or maybe Ruby?) to avoid that.

OK ... now this is really cool !!!

still testing it out ...

Martin this is awesome!!!!
Thank you so much!!!
There is a big trouble, this should be a little more faster and have some mechanism of autorecover, something in constant way push for recover the volume, if audio program change to lower value under limit then the audio always remain at lower volume.

For sure for now I not listen any noticiable distortion, continue testing, awesome job!

The original idea of the device was to avoid clipping, so it will only reduce, not increase volume automatically. I thought about adding a Reset button, but that would be the same as turning up the Volume knob which is already there. Would it make sense to add an option for automatic gain build-up at some (adjustable) rate like 0.1 dB per second? It would go on until it hits the limit, gain is reduced, starts to creep up again, and so on? Is there a better way?

This type of utility can have many requirements I think.

Having a level 'knock down' is definitely one of them.

A constant monitoring/adjusting gain to maintain a 'specified' output would be another.

Should this develop further ... it would be nice to have some 'OPTION' button on the GUI to set how the plug is to work.

Just my idea/suggestion.