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R128Norm, A simple normalizer based on EBU R128
markanini
post Feb 14 2011, 03:26
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If you get more time to tweak this plugin I think you'll find this audio material useful. It's really a stress test for R128Norm wink.gif

http://www.villex.com/downloads/villex.mp3
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geoslake
post Apr 30 2011, 11:23
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Thanks a lot for this plugin.
However, could you add some tweakable settings (threshold etc), as it is, it's almost unusable for classical music (for which this kind of effect is almost necessary), as even the silent parts are maximized to 0db or so...

Thank you
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Nowings69
post May 21 2011, 21:04
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Since Replay Gain changed to R128,
R128Norm adds Noise first and irregularly.

Because I detour around it with Replay Gain's normalize(94db?)
If R128Norm will be one as Replay Gain's normalize,
please unify them definitely(e.g. Target MP3 alteration volume level : 89db to 94db)



This post has been edited by Nowings69: May 21 2011, 21:54
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archetwist
post Aug 5 2011, 20:04
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It's a nice plugin and very useful but I have to say I also experience the blasts in the beginning of some songs.
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SamDeRe81
post Nov 5 2011, 20:22
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I'm not sure if you're still updating this, but I'd prefer to use this over tagging because I regularly use MusicBrainz Picard to update my tags and I have CLEAR TAGS checked so it re-writes everything. I figure there might be a change and a tag gets removed entirely so if I don't do that the older one would stay present. I want it to mirror their database completely.

Thanks for the great work!

This post has been edited by SamDeRe81: Nov 5 2011, 20:22
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GeSomeone
post Dec 19 2011, 21:02
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Just tried version 1.11. It is really impressive. I think you nailed it.


--------------------
In theory, there is no difference between theory and practice.
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SamDeRe81
post Dec 20 2011, 01:48
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Thanks for continuing updates to this plugin
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geoslake
post Dec 20 2011, 09:35
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QUOTE (TuNk77 @ Jan 31 2011, 18:13) *
Thank you for making this DSP kode54, now I can finally get rid of that horrible Vlevel.


I wish I could but I cant do so, on classical music, r128norm is unusable sad.gif
When there's a super quiet section, it stays quiet, then suddenly volume rises a lot, its far from being as smooth as Vlevel (which probably only reacts quicker, i dunno).
I think we should have access to a few parameters at least..

Thanks anyway
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GeSomeone
post Dec 20 2011, 18:28
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QUOTE (geoslake @ Dec 20 2011, 10:35) *
I wish I could but I cant do so, on classical music, r128norm is unusable sad.gif
[..] its far from being as smooth as Vlevel

Yes, two different beasts. Vlevel is designed to be smooth and you can configure it to match your liking best.
This plugin however tries to implement the EBU R128 (broadcasting) norm in a usable way. That means the kind of compression commonly used on pop/rock/country etc. (FM) radio.
I think that is not usable on classical music and that Vlevel is the better choice there.
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geoslake
post Dec 21 2011, 10:58
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Maybe with an "advanced" mode or something, which would allow user to set their threshold, gain etc ?

This post has been edited by db1989: Dec 21 2011, 16:02
Reason for edit: removing unnecessary full quote of above post
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kode54
post Dec 21 2011, 22:54
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Yes, I did make a lot of constants into variables, because I plan to add a configuration dialog. The current variables are:

Startup latency. The minimum amount of input sample data which will be processed into the EBU R128 scanner before any volume level is calculated and eventually applied. Currently 10 seconds.

Processing window. The duration worth of gated loudness samples, taken 2.5 times per second by the library, which will be retained and processed every time a gain correction is made. Currently 20 seconds.

Minimum latency / look-ahead. The minimum amount of sample data which will remain buffered after the startup period occurs. Volume changes occur at least this much behind the data which has been fed through the scanner. Currently 4 seconds.

Volume change sensitivity. This is the rate at which volume changes occur between the current level and the new level. Currently 1dB per 50ms.

Reference loudness level. This is the volume level that the scanner tries to achieve. Currently -18 LU, which is equivalent to the ReplayGain scanner component.
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thuneau
post Jan 10 2012, 20:32
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This plugin is great. It's perfect for evening background music playback without using radio station style compression, which hurts the sound.
Would it be possible for you to compile it into a VST plugin? I would buy a VST provided it wasn't priced outrageously.
By the way, Waves just released a R128 compliant meter and they are asking a lot of money for it- $299.00. http://www.waves.com/content.aspx?id=11884
Your plug could display the measured values plus ride the gain in real time.

This post has been edited by db1989: Jan 10 2012, 22:38
Reason for edit: removing unrelated full quote of above post
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kode54
post Jan 10 2012, 23:46
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The thing is, I'm not sure if VST supports such latency as this component needs. It needs to absorb the first 10 seconds worth of sample data fed into it, and then it maintains a constant latency of 4 seconds. Not to mention that I've only ever worked on a VST (instrument) host, not the plug-in side of things, even if it shouldn't be that much different.
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thuneau
post Jan 11 2012, 08:19
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QUOTE (kode54 @ Jan 10 2012, 18:46) *
The thing is, I'm not sure if VST supports such latency as this component needs. It needs to absorb the first 10 seconds worth of sample data fed into it, and then it maintains a constant latency of 4 seconds. Not to mention that I've only ever worked on a VST (instrument) host, not the plug-in side of things, even if it shouldn't be that much different.


I'm not sure what the limit on pre-buffering audio in VST is. Considering that there are VST delay processors, which can be set up to output fully wet or 100% effect signal, I see no problem with having your plugin act like one. Yes, it would not be a real time effect, as it would push audio back by 10 seconds every time, but for processing complete stems inside a VST host it could be valuable.
Also, there is technology in the VST spec that allows for outside of host processing as used by Melodyne or similar pitch correcting processors. Basically complete audio track is temporarily exported to the VST, processed in whole and then inserted back where it belongs on a time line. Something to consider for you.

People compress complete mixes or submixes with various dynamics processors. Automatic gain riding like your processor does, could be used to very transparently compress audio to a different standard- just another cool tool for music creators.
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thuneau
post Jan 17 2012, 20:22
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QUOTE (kode54 @ Dec 21 2011, 17:54) *
Yes, I did make a lot of constants into variables, because I plan to add a configuration dialog. The current variables are:

Startup latency. The minimum amount of input sample data which will be processed into the EBU R128 scanner before any volume level is calculated and eventually applied. Currently 10 seconds.

Processing window. The duration worth of gated loudness samples, taken 2.5 times per second by the library, which will be retained and processed every time a gain correction is made. Currently 20 seconds.

Minimum latency / look-ahead. The minimum amount of sample data which will remain buffered after the startup period occurs. Volume changes occur at least this much behind the data which has been fed through the scanner. Currently 4 seconds.

Volume change sensitivity. This is the rate at which volume changes occur between the current level and the new level. Currently 1dB per 50ms.

Reference loudness level. This is the volume level that the scanner tries to achieve. Currently -18 LU, which is equivalent to the ReplayGain scanner component.


May I suggest you make maximum gain your processor applies also a variable? Sometimes long quiet parts of music get amplified too much. This happens mostly with acoustic, jazz and classical. If you could specify a max gain of let's say 12-18dB all that ambiance would not come up too drastically in those moments.
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Masza
post Aug 10 2012, 17:44
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I think this plugin works quite nice. It gives a certain FM radio 'feeling' by compressing and pumping the volume. Sometimes it even gives a nice punch to the certain parts of the songs by doing that.

A very nice plugin when you want to listen to for example your whole library, there sure is some volume differences between some songs - especially if you haven't done ReplayGain scan to your files. I haven't and I have lot of music.. I tried it.. but it would have taken quite a long time to complete it.

But, I have two wishes:
1) sometimes it pumps levels too much, for example in the end of the songs that have a long fade out or in songs that have long quiet parts
2) it would be also nice if the reference level could be changed. Foobar's own preamping isn't working either with this plugin.
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jackda
post Jan 28 2013, 16:24
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Hey guys,

how is this plugin different from replaygain?
what's the benefit of using it ?

thanks smile.gif
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kode54
post Jan 29 2013, 05:08
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It uses the same algorithm of volume measurement as the current ReplayGain scanner, only it uses a pre-fill scan and latency window to apply a short-term volume correction on the audio in real-time. It may have a similar volume measurement and scaling effect, but it is more of a normalizer, as it only scans small amounts at a time, and is prone to volume swings being scaled to more of a flat response, while normal ReplayGain and R128 scan the volume level of the entire track or album at once and only apply a single correction level on a track by track or album by album basis.
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Fandango
post Mar 10 2013, 21:13
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Hm, having read the whole thread now and only been listening to some tracks for a short while, I wonder what this plugin does when a replaygain scanned track is being played.

In case it's always active and ignores replaygain information, it should still have an effect if the song has sections with highly differing dynamics, right? If that is the case wouldn't an option be good to disable this DSP if replaygain information is present? Well, ideally I'd also love to see an option to only activate this plugin on internet streams.

This post has been edited by Fandango: Mar 10 2013, 21:16
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marc2003
post Mar 10 2013, 22:56
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http://www.hydrogenaudio.org/forums/index....=0&p=823073

post 15 onwards will be of interest.
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Fandango
post Mar 11 2013, 00:57
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Well, the question has been raised in #18, but it wasn't answered. I might look into this Dynamic DSP, but I don't think it's worth the hassle just for a few radio streams.

This post has been edited by Fandango: Mar 11 2013, 01:01
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Denma Panter
post Mar 11 2013, 09:05
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@kode54
Can you add like this?

1. If There is ReplayGain information, R128Norm pass through.

or

2. Add check box for Preamp of foobar2000 (eg. Use R128Norm for Without RG info)

I think the latter is more better
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kode54
post Mar 12 2013, 17:46
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QUOTE (Denma Panter @ Mar 11 2013, 01:05) *
@kode54
Can you add like this?

1. If There is ReplayGain information, R128Norm pass through.

The third party Dynamic DSP component already handles this perfectly, as outlined in this post. Just be sure to add this instance of the Dynamic DSP to the top of your playback chain.

QUOTE (Denma Panter @ Mar 11 2013, 01:05) *
2. Add check box for Preamp of foobar2000 (eg. Use R128Norm for Without RG info)

I think the latter is more better

Sorry, that's beyond my control.
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Denma Panter
post Mar 12 2013, 21:30
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I thank you very much.

1. RG info with Daymic DSP to R128Norm = x645.249

2. RG info with nothing = x673.728

3. RG info with only R128Norm = x359.160

This post has been edited by Denma Panter: Mar 12 2013, 21:42
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Aldem
post Apr 26 2013, 02:57
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Got a question about how to use this component.

Before you created this component, I've scanned my entire FLACs library and applied RG in track mode.

So now, in order to try to hear the difference between the old RG and this R128 one, in options -> playback, I shout disable RG (source mode and processing set to none) and only use your R128 component in the DSP manager right ?

Thank you

Aldem
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