Download plugin here (http://sjeng.org/ftp/fb2k/foo_dynamics.dll)
Does as it says.
A lot of people wanted this, as if music isn't compressed enough already. Shame on you.
PS. By nature, it also works as an automatic gain control and it knows about clipping. On some very heavily compressed music, it will actually end up lowering the overall volume. Draw your own conclusions.
Thx for making the plugin
Not that I've missed it, but now that I've got it, I can use it to hear how compression affects the music.
Have read about compression lately, and how most new cd's today are victims of this.
The only thing I missed immediately was a slider scale to remember settings.
It would be nice if you could add it, if it isn't too much trouble.
THANKS!
THANKS!
THANKS!
Could you please add a label telling the actual value of each setting?
Uploaded new version.
There's a problem with the interface & rounding, will try to fix.
Fixed, yet another new version uploaded.
excellent!
thank you, Garf.
this is a very interesting plugin. Thank you
Thank you very much !!
I really needed this plugin !
So - thx again !
A lot of people wanted this, as if music isn't compressed enough already. Shame on you.
Absolutely. My first reaction when I saw this thread was: "A compressor for FB2K ?? Oh, the horror !!"
But then upNorth mentioned:
Not that I've missed it, but now that I've got it, I can use it to hear how compression affects the music.
OK, this seems to be a valid reason to use it.
But, to all the people who asked for it: please tell me this is the ONLY thing where you will use the plugin for...
Why are some people so dead-set against compressors and the like? Why does it really matter if some people want to screw up the sound coming out of their speakers?
Actually, for some classical recordings, it has real use. Or for movie stuff or so.
Face it: almost any piece of music you're listening to is already compressed. If you'd listen to uncompressed music most of you would find it very unnatural-sounding, believe me. :-)
Different instruments produce different levels. To make stuff equally loud (so that the bassline won't eat the lead riffs and such, volume-wise), music must be compressed. On album-level, this is a part of what "mastering" is all about.
These days, compression is often already applied in the production stage (especially on electronic music). Live recordings are compressed a lot, too - de-esser and plosive removal are essential compressor tasks to make vocals recordable and reproducable at a normal level via a PA (have you ever tried recording your voice into the PC via a mic? you noticed how sharp S, SH and CH sounds, and how loud P, K, T are - this is controlled with a compressor).
Further compression is often applied on the radio, to make the signal that is broadcast as loud and "even" as possible, utilizing the maximum of the frequency band (ever noticed how different records may sound when you compare the track as you heard it on the radio vs. the track from the CD on your home stereo?).
Summary: a plugin that is compressing your music is a nice gimmick and really not needed - unless you have loads of your own garage punk rock band. :-)
Actually, for some classical recordings, it has real use. Or for movie stuff or so.
Well, I'm not going to get into whether or not it has valid uses. That's beyond my area of expertise. I just think it's odd that so many people feel they need to crusade and try to save everyone else's music.
But on a different note, the compressor sounds quite nice. It does its job well. (I don't normally run any DSPs except the attenuator and the advanced limiter, so it doesn't really matter to me, but it's nice nonetheless.)
Very useful to listen music at low volume, night for example without disturbing wife or neighbors and without losing delicates passages.
In addition it is possible to make a plugin to increase dinamique? it is much less simple to make but that would be very useful!
In addition it is possible to make a plugin to increase dinamique? it is much less simple to make but that would be very useful!
Yeah, considering terrible dynamic compression on most of the "modern" Cds, I think such a plug-in would be really great. However I suspect it should be more difficult to acheive.
Thanx Garf for your compressor. I will use it for sure on my next insomnia night, when I listen to music and try to relax a bit
Very useful to listen music at low volume, night for example without disturbing wife or neighbors and without losing delicates passages.
This is why I have headphones
In addition it is possible to make a plugin to increase dinamique? it is much less simple to make but that would be very useful!
Compression is a one way process, sorry. No way to recover the original stuff.
In addition it is possible to make a plugin to increase dinamique? it is much less simple to make but that would be very useful!
Compression is a one way process, sorry. No way to recover the original stuff.
I understand that "original" dynamic is lost after compression : the louder and the softer parts of the waveform are closer after compression, and it is not possible to dertermine what it was before, and so to go back.
However, don't you think it would it be possible to make an "expander", that would increase arbitrarily the dynamic range of the sound, making the slope that separates louder and softer noises steeper, rather than flatter like a compressor ?
EDIT : replaced "expender" by "expander"...
I can't see a way to do so. I'm not an expert by any means, but audio just doesn't lend itself to such operations. I suppose it would be possible to arbitrarily pick a point, move that point, and redistribute the waveform data accordingly, but it would almost certainly sound terrible.
Very useful to listen music at low volume, night for example without disturbing wife or neighbors and without losing delicates passages.
This is why I have headphones
But the night, during kamasutra's meeting
headphones aren't very useful
Thank you GARF!
Perhaps man means something like an expander?
Perhaps man means something like an expander?
Yes that's it, an expander, not expender as I said...
I edited my previous post to correct.
Anyway I understand this sould not be easy to do, and even to use, especially because the arbitary point should be set at different specific values for different music files...
But I think the objective is great (to increase the dynamic). I can't see how this could be done...
The first thing I found on Google about expanders: http://www.oneonta.edu/academics/music/xla...or_expander.htm (http://www.oneonta.edu/academics/music/xlab/studio/courses/192/Signal_Processing/compressor_expander.htm)
I am not sure I understand...are you referring to a widener. M Pesch's 1by1 player has a compressor and widener as its sound enhancer. there is a plugin for winamp and of course you could download 1by1 as well. Sorry if this seems off topic but I thought that this was somthing that related.
http://www.rz.uni-frankfurt.de/~pesch/ (http://www.rz.uni-frankfurt.de/~pesch/)
you can take a look
...at night for example without disturbing wife...
But the night, during kamasutra's meeting
headphones aren't very useful
[span style='font-size:21pt;line-height:100%']
BUSTED[/span]
Compression is a one way process, sorry. No way to recover the original stuff.
dbx?
<edit> not that dbx will be useful as an expander towards the goal desired here, but rather that it is an example of compression-expansion. And Dolby's similar methods for that matter.
Compression is a one way process, sorry. No way to recover the original stuff.
dbx?
What dbx? They have different maths than the rest of the world?
well, mind explaining yourself when Dolby and dbx both contradict what you're saying?
Please post the exact statements here, I'd love to see them explain how they work around the laws of information theory.
you're unfamiliar with their tape noise reduction systems? Both companies employ fixed-slope compression and expansion schemes to reduce tape hiss and increase dynamic range.
fixed-slope
When's the last time you bought a CD compressed with a standard fixed slope?
fixed-slope
When's the last time you bought a CD compressed with a standard fixed slope?
that's not the point, I don't think that what's being asked for here is something that will bring every single compressed CD back to the exact state it was in before it was compressed. Obviously such a thing would be impossible, because you have no way of knowing the exact way it was compressed. The point I'm trying to make is that it is possible to expand something, since the principle's been in use in the industry for years and years and years, when Garf seems to be pretty sure that it's impossible. Yes, it would be impossible to do what you're talking about, and Garf's right in stating that getting the original signal back when you don't know how it was compressed in the first place isn't possible. But it also sounds like he's saying it's impossible to expand anything at all, actually. When clearly it isn't impossible, or else dbx noise reduction would never have worked. And it did.
they are 2 various typologies:
1) dbx or dolby ( a, b, c, s ) are compressor-expander systems to reduce hiss on analog tape.
original signal is compressed when recording and expanded when play and restore original dynamic.
2) expander is a signal processor what increases dynamic when play.
obviously it can't restore original dynamic but it can improved some recordings.
not to use all the time.
for exemple:
expander (http://www.ntrack.com/comp/comp.shtm)
GARF , do you think add preset support, same for equaliser?
But it also sounds like he's saying it's impossible to expand anything at all, actually.
I _NEVER_ said that. I said it was impossible to get the original back. Note that even if you know the compressor, it will be impossible to get the exact original back, if there's a multiple-to-one mapping. And this happens in all compressors.
So you can do some expansion and get something reasonable, but never what was the original. Simple maths. You can't make anything out of nothing.
Expansion is a lot harder to make work well than a compressor, and I don't know of any ones that I consider good enough in general use to make it worth the hassle.
GARF , do you think add preset support, same for equaliser?
Do you think it would be usefull?
Yes, i think it would be usefull, same as preset support is usefull for equaliser.
But I think that what I think is less important than you think on this subject, because it is you who made work...
if you think that it is useful and not too complicated to make I would think that it is well.
if not, the wise would say: compressor without preset support is better that no compressor!
EDIT: please , do you want explained how parameters act on signal, to educate ignorant people, me the first.
Someone posted a link to an explanation about compressors before in this thread, explains it better than I could probably.
I'll consider presets.
I _NEVER_ said that. I said it was impossible to get the original back. Note that even if you know the compressor, it will be impossible to get the exact original back, if there's a multiple-to-one mapping. And this happens in all compressors.
So you can do some expansion and get something reasonable, but never what was the original. Simple maths. You can't make anything out of nothing.
Expansion is a lot harder to make work well than a compressor, and I don't know of any ones that I consider good enough in general use to make it worth the hassle.
OK, explain this then:
Compress
1 amount over 5 is -4 so output 1
2 amount over 5 is -3 so output 2
3 amount over 5 is -2 so output 3
4 amount over 5 is -1 so output 4
5 amount over 5 is 0 so output 5
6 amount over 5 is 1 so output 5 + 1/2(1) is 5.5
7 amount over 5 is 2 so output 5 + 1/2(2) is 6
8 amount over 5 is 3 so output 5 + 1/2(3) is 6.5
9 amount over 5 is 4 so output 5 + 1/2(4) is 7
10 amount over 5 is 5 so output 5 + 1/2(5) is 7.5
Expand
1 amount over 5 is -4 so output 1
2 amount over 5 is -3 so output 2
3 amount over 5 is -2 so output 3
4 amount over 5 is -1 so output 4
5 amount over 5 is 0 so output 5
5.5 amount over 5 is 0.5 so output 5 + 2(0.5) is 6
6 amount over 5 is 1 so output 5 + 2(1) is 7
6.5 amount over 5 is 1.5 so output 5 + 2(1.5) is 8
7 amount over 5 is 2 so output 5 + 2(2) is 9
7.5 amount over 5 is 2.5 so output 5 + 2(2.5) is 10
Looks like the original was recreated to me. Explain how it wasn't, please.
Looks like the original was recreated to me. Explain how it wasn't, please.
Most audio formats (including CDDA) don't store floats, they store integer values.
Most audio formats (including CDDA) don't store floats, they store integer values.
so? Got absolutely nothing to do with what I just said.
<edit> Happy?
512
1024
2048
4096-2048->2048(0.5)->3072
8192-2048->6144(0.5)->5120
16384-2048->14336(0.5)->9216
512
1024
2048
3072 1024(2)+2048=4096
5120 3072(2)+2048=8192
9216 7168(2)+2048=16384
4096 4097
ABX the rounding errors then.
I can't ABX MPC --insane either, that means I can get the original back via some procedure you will no doubt explain us?
you're using integer storage as an excuse to not admit you're wrong.
<edit> in other words, the only reason you say it's impossible to get the original back is because only integers are stored in the one format you were thinking of. Even though it would more than likely be impossible to tell any difference if you actually did try compressing and then expanding some 16bit audio. Yes, I understood from the beginning that if you were dealing with 16 bit integers you would not be able to get exactly the same data back because of the margin of error when dealing with odd value samples. But with the actual results, I sincerely doubt you'd hear anything different, and I'm sure you do to.
Yes, I assume the music is not stored in floats after compression. This is true for any practical format I know of. Lossy audio can be floats but suffers from the same problem that the mapping is not invertible. Moreover, compressors have variable gain to attain a constant average loudness. You cannot figure out the original loudness levels either even if you use floats.
No, it does not only have to do with odd/even problems, the sharper the compression, the more values are mapped to the same output value.
It gets even more fun when the compressor causes hard limiting, nothing will save you in that case.
now you're mixing issues again. I'm not talking about guessing the compression method of any random CD in order to obtain exactly what was there before the compression occured, as my example clearly was a given scenario. In one instance we're talking about a compressor that does a straight 2:1 compression after a given point, and then you start talking about variable gain compressors as if they were even related to the process I'm talking about. Fine. You have problems admitting when you're wrong. Citing extreme and/or unrelated scenarios to try to prove you're right. Yes, you're right in your scenarios, I could not guess a random compression method from a random CD. I'll drop it. Fact remains, if you couldn't get out what was put in, dbx's product never would have worked. And it did. You're arguing that an expander would be worthless because I can't guess those values or have enough accuracy to get the original back even if I could. Well, it wouldn't be worthless because I don't need guess how it was compressed or have enough accuracy to get the original back after guessing, because I only need to make it sound better than its current state.
<edit> much like clipping restoration isn't useless just because it has no way of knowing whether or not the signal during the clipped portion would have been a smooth curve matching the slopes of the waveform before and after the clipped portion. Yes, there could have been more than a simple curve in that portion of the signal, but that doesn't make the clipping restoration process worthless. No, it doesn't necessarily give you what was originally there before it was clipped off, but it does give you an unclipped signal.
I'm not talking about guessing the compression method of any random CD in order to obtain exactly what was there before the compression occured, as my example clearly was a given scenario.
True. In such a case, you can do a reasonable reconstruction that would probably be audibly the same, though not identical to the original. Is this case typical? Not at all.
You have problems admitting when you're wrong.
My original point still stands as strongly. You cannot invert a compressor for all practical purposes, you can't invert a one-way function, and you cannot invert something that maps multiple values into one.
You haven't come an inch closer to refuting this.
Fine. You have problems admitting when you're wrong.
Is this another way of saying you have run out of arguments?
Citing extreme and/or unrelated scenarios to try to prove you're right.
There is nothing extreme to the scenario I quoted, it's the typical one on most CD's. It's not unrelated either, since it's the one you actually have to deal with in practise.
I refuted your theorethical example, and you say my refutation doesn't matter in practise. Right. But your example wasn't practical, so that's not an argument.
Well, it wouldn't be worthless because I don't need guess how it was compressed or have enough accuracy to get the original back after guessing, because I only need to make it sound better than its current state.
This I wont argue, it might be possible to make something that reasonably guesses the used method and makes some additional dynamics out of it. Would it work well? So-so, depending on what the original is.
But you can't get it back.
<edit> much like clipping restoration isn't useless just because it has no way of knowing whether or not the signal during the clipped portion would have been a smooth curve matching the slopes of the waveform before and after the clipped portion. Yes, there could have been more than a simple curve in that portion of the signal, but that doesn't make the clipping restoration process worthless. No, it doesn't necessarily give you what was originally there before it was clipped off, but it does give you an unclipped signal.
Yes, you are apparrently agreeing exactly with what I already said:
...I _NEVER_ said that. I said it was impossible to get the original back...
...So you can do some expansion and get something reasonable, but never what was the original. ..
So, if you agree with me, why are you wasting my time?
My original point still stands as strongly. You cannot invert a compressor for all practical purposes, you can't invert a one-way function, and you cannot invert something that maps multiple values into one.
You haven't come an inch closer to refuting this.
Fine. You have problems admitting when you're wrong.
Is this another way of saying you have run out of arguments?
Fine, I'll continue if you want, and I'll ask again, will you explain how dbx didn't work then? If you can't explain how dbx's noise reduction technique did not work then there is no point in you saying one more word about it, because you are stating that it could not work, yet, it does.
quite clearly I don't agree with you. dbx does what you say is impossible. heh.
Fine, I'll continue if you want, and I'll ask again, will you explain how dbx didn't work then? If you can't explain how dbx's noise reduction technique did not work then there is no point in you saying one more word about it, because you are stating that it could not work, yet, it does.
By doing a reasonable, though not perfect, reconstruction.
They know the exact input compressor settings, they know the exact output compressor settings. That also helps (loads).
Dolby C/B also work. They aren't perfect. They won't allow you to restore a tape that was encoded with something else.
so now you're using what I already stated as your own argument? heh. And while you're quoting yourself I suggest you go re-read what you wrote and what you were replying to. The guy was asking if it were possible to increase dynamics. Your reply was that it was not, and only after it was pointed out to you that you were wrong did you change your tune.
so now you're using what I already stated as your own argument? heh. And while you're quoting yourself I suggest you go re-read what you wrote and what you were replying to. The guy was asking if it were possible to increase dynamics. Your reply was that it was not, and only after it was pointed out to you that you were wrong did you change your tune.
I wrote it wasn't possible to get the original back. You are right that I did not answer the original question, namely is it possible to make an expander (yes).
Since I already made this clarification a few pages up, I still fail to see why you started arguing against my point that it was impossible to get a perfect reconstruction, even going as far as giving flawed examples to try to illustrate you can invert a compressor perfectly.
basically because first you were stating something that was wrong. then you were stating something that relied on degree of accuracy to be wrong. and dbx doesn't give a reasonable reconstruction, for all intents and purposes it gives a perfect reconstruction, since it takes place in the analog world, where your accuracy argument is pretty much meaningless. "Perfectly" doesn't apply. You're talking about dealing with digital audio, which doesn't represent audio perfectly in the first place, what with only sampling the audio at certain distances apart in time. If you want to look at it from a integer math point of view, fine. But then it just comes down to how accurately you're doing your math. In which case I'm right in one instance, and you're right in another.
basically because first you were stating something that was wrong. then you were stating something that relied on degree of accuracy to be wrong. and dbx doesn't give a reasonable reconstruction, for all intents and purposes it gives a perfect reconstruction, since it takes place in the analog world, where your accuracy argument is pretty much meaningless. "Perfectly" doesn't apply. You're talking about dealing with digital audio, which doesn't represent audio perfectly in the first place, what with only sampling the audio at certain distances apart in time. If you want to look at it from a integer math point of view, fine. But then it just comes down to how accurately you're doing your math. In which case I'm right in one instance, and you're right in another.
Of course I'm dealing with it from an integer point of view, since I was asked to make a plugin for FB2K, which is generally used to play back music from CDs.
Whatever is possible in the analogue domain is of no relevance to that.
...and foobar contains 64bit audio representation because...? heh.
The internal respresentation does not matter. 16 bits of resolution stored in 64 bits still remains 16 bits of resolution.
*sigh* You're right. You've never been wrong in your life. Level of data accuracy/resolution doesn't affect computed results at all. You said earlier that it does. But now you say it doesn't. So I guess it doesn't. This is my last post in the thread regarding my discussion with you Garf, so go ahead and get the last word in.
STOP FLAME
yeah, a little childish I guess. Sorry Garf, actually I'm annoyed at someone in the house right now, heh.
hi garf
i am using your fb compressor since you posted it. i really like it the way it is. it does what i want and better than many expensive sw's i use, sure, it's my opinion.
well, i have some questions for you. so, please, i would be very happy if you answer me.
first, i am using following parameters: peak limit 95%, release 50 ms, FSR 1.3, SRC 2.0 and input gain 3 db.
second, my mp3's are from net radio, most of them are dance or trance; i use 'mp3gain' normalization to max non-clip. after all, i convert to wav, using fb too.
questions: what mean 'fast and slow compression ratio'? how can i compare with attack/release? when do these parameters start to work?
again, thank you very much for this great piece of well done work!
The slow compressor is the one that slowly responds to changes in the overal loudness. The fast compressor responds almost instantaneously to short peaks and bursts.
Not sure if that helps you much.
no problems.
i will analyze what results it generates.
anyway, thank you for the reply.
Garf,
the slow compressor obviously uses some kind of absolute threshold, only kicking in if the peak is above some -20dB for a while. So if it is fed music with some very silent and some not-quite-so-silent passages, the former remain unchanged, while the latter are amplified. In this case, the compressor effectively does the opposite of what it's supposed to do, increasing dynamic range rather than compressing it.
I tried using the compressor for classical music, but this effect can be quite annoying. Very silent passages are just as hardly audible as before (they are actually unchanged, I visually verified this with an audio editor), plus there are now sudden, unexpected jumps in volume if the music gets just a tiny little bit louder.
Is this intended behaviour?
I cant seem to get the Dynamic compressor to work how I want it to. I want to basically have everything bought up to a 0dB level peak.
What seems to happen is with some bass boose in the eq which is ahead of the dynamics compressor in the DSP manager, is that when the bass kicks in, the compressor will turn down to prevent clipping as it should, but it never brings the level up again.
This is really obvuious in most of daft punk - discovery where the bass comes and goes all thru the first track. As soon as I touch any slider in the dynamics compressor screen, the level comes up again.
At the moment I have the compressor configured with
peak limit 100%
Release time 10ms (cant go any lower then that)
Fast ratio 40:1
Slow compressor 1:1
input gain 12dB
There is also a little gain on the EQ ahead of it to bring the bass up.
I want to get it so I can hear the whole of the songs the whole of the time, without the loud parts being audable in the next cubicle over. At the moment as soon as it attenuates something, thats all she wrote till I nudge one of the sliders in the DSP manager.
Garf,
do you have time to explain how the Dynamics Compressor works please?
It may sound like a stupid question, but just playing around with it doesn't make the algorithm clear at all.
I'm using 0.8 special, and (for example) Peak limit does this
0%=silence
1%=~6dB down
anything higher = no apparent effect.
I was thinking that the dynamic compressor would be the ideal thing to put at the end of the chain (adjusted so it's actually ineffective most of the time) to deal with albums that have been clipped by ReplayGain or (more likely) ReplayGain plus a positive pre-amp setting.
I was expecting to be able to set Peak Limit to 100%, Release time to something sensible (near-infinite would be nice, but quite fast would be good for some users), and put the compressors to 1:1. However, Release Time seems to be stuck at infinite, whatever the setting.
Can you shed any light on this?
Cheers,
David.
To put it another way: are me and richms doing something wrong, or has a bug crept in?
Cheers,
David.
Garf,
the slow compressor obviously uses some kind of absolute threshold, only kicking in if the peak is above some -20dB for a while. So if it is fed music with some very silent and some not-quite-so-silent passages, the former remain unchanged, while the latter are amplified. In this case, the compressor effectively does the opposite of what it's supposed to do, increasing dynamic range rather than compressing it.
I tried using the compressor for classical music, but this effect can be quite annoying. Very silent passages are just as hardly audible as before (they are actually unchanged, I visually verified this with an audio editor), plus there are now sudden, unexpected jumps in volume if the music gets just a tiny little bit louder.
Is this intended behaviour?
The behaviour is intended, the goal is not to boost up background noise to extreme levels. Turn the input gain slider up, it should be enough to bring the quiet pieces into the amplification range (if not, I'll have to extend the range),
Garf,
I'm using 0.8 special, and (for example) Peak limit does this
0%=silence
1%=~6dB down
anything higher = no apparent effect.
Peak limit determines the maximum output level ever used. I cannot reproduce your problem - it works as expected here.
For what you're doing, the Advanced Limiter would be much better though.
I want to basically have everything bought up to a 0dB level peak.
Uh? You mean as a peak normalizer? You'd have to disable all compression then. Not sure if this would work well.
What seems to happen is with some bass boose in the eq which is ahead of the dynamics compressor in the DSP manager, is that when the bass kicks in, the compressor will turn down to prevent clipping as it should, but it never brings the level up again.
This is really obvuious in most of daft punk - discovery where the bass comes and goes all thru the first track. As soon as I touch any slider in the dynamics compressor screen, the level comes up again.
This is the AGC. If you touch a slider, everything resets, and the AGC has to readjust. It doesn't turn the volume down to prevent clipping - it can compress after all. But it does correct everything to the same approximate level. Don't forget that if you have Input gain to 12dB and you click something, the first few seconds will go 12dB over peak scale!
But if I understand you correctly it seems that it doesn't properly adjust up again if the loud section is followed by a more quiet section? I certainly haven't noticed such a thing...
The behaviour is intended, the goal is not to boost up background noise to extreme levels.
Thought so. Of course, this will only happen if the compression ratios are very high.
Turn the input gain slider up, it should be enough to bring the quiet pieces into the amplification range
I tried this approach, but a) this does what the threshold is designed to prevent: increase background noise in completely silent passages (I have to use more than 10dB on the CD that I have problems with, and even then the very beginning remains unchanged), and b) it seems to me that the louder parts now get compressed more than I like. Don't know if this makes sense, I don't really know anything about dynamics compression.
(if not, I'll have to extend the range)
Would it be possible to make it adjustable?
Garf,
I'm using 0.8 special, and (for example) Peak limit does this
0%=silence
1%=~6dB down
anything higher = no apparent effect.
Peak limit determines the maximum output level ever used.
Yes, I was hoping that was the case!
I cannot reproduce your problem - it works as expected here.
I'll try re-installing it.
For what you're doing, the Advanced Limiter would be much better though.
For
me, I'd probably avoid both. But I was trying to find a "nice" solution for people who complain that ReplayGain makes things too quiet. I assumed they'd put the pre amp around +8dB. The behaviour I wanted was to leave anything which peaked below 0dB FS alone, but to compress anything above 0dB FS
nicely. If you use the Advanced Limiter for this, you'll find that it make the audio unlistenable. It must be possible to get a better result, because it's quite easy to take a file with a ReplayGain of 0dB (i.e. a quiet track), and compress it so it sounds like a typical pop track with a ReplayGain of -8dB. That's effectively what someone who enables ReplayGain, and pushes the pre-amp to +8dB is wanting to do, but the advanced limiter just shreds the audio, and I can't make the DRC do the job either.
Like I said, I'll try re-installing it. It should be possible to make the dynamic compressor do this, shouldn't it? Or will other "unseen" parameters and processing prevent it from dealing usefully with stuff that is over 0dB FS on input?
Cheers,
David.
But if I understand you correctly it seems that it doesn't properly adjust up again if the loud section is followed by a more quiet section? I certainly haven't noticed such a thing...
I'm seeing that too.
richms - does the "peak" slider work for you? if not, we could be seeing the same set of problems.
Cheers,
David.
Hello,
Quick question: I would like to achieve an effect "kind of like" Audiostocker for Winamp does, that is, set a whole bunch of songs from widely different sources and volumes in random, for background music while in the office, and not have to worry about the volume knob in my speakers. In this situation, I couldn't care less about the relative volumes of songs in their albums.
Any advice? I would think this plugin is the right one to achieve that, am I right?
That sounds more like trackgain's job to me...
Any advice? I would think this plugin is the right one to achieve that, am I right?
To me it sounds like replaygain (trackgain) does what you want, but since I'm quite sure you know / have tried it already, I might have misunderstood what you want...
edit: oops ... (much) too slow
Yep, I thought so, but then I tried selecting all my playlist, right-clicking, Replaygain > Scan per-file track gain, and got a bunch of errors about not being able to update the info in the files... Any hints?
In any case, I would like to still use a dynamics compressor to avoid problems with songs that have big differences in volume (not hearing the quiet passages).
But of course, the thing about trackgain is going a bit OT for this thread...
Ok, I am stupid, I was completely sure I had already unset the read-only flag on the files on my hard drive, but it turns out I hadn't... Now the scan track-gain operation works fine... Not surprising...
Yep, I thought so, but then I tried selecting all my playlist, right-clicking, Replaygain > Scan per-file track gain, and got a bunch of errors about not being able to update the info in the files... Any hints?
If you have database enabled, RG info for files without tags that can store replaygain information (e.g. .wav files) will be stored in fb2k's database. The "error message" just tell you that the RG info isn't written to the file directly, but fb2k will still be able to use it.
edit: didn't see your last reply when writing this one...
Sorry about asking a stupid question but I want to listen to music in a noisy enviroment(for example doing the dishes while listening to music) and I want to hear detailes a little better, will the default setting do or could tweaking them be a good idea? Is the VLevel plugin better of worse than the Dynamic Compressor for the way I want to use it?
It's been a while!
I understand that:
The 'Fast Compression Ratio' has a fast attack.
The 'Slow Compression Ratio' kicks in after -24.3 dB and the attack speed is much slower.
What I still don't get:
The 'Fast Compression Ratio' will not work independantly of The 'Slow Compression Ratio'. How does this work?
And how does this dependancy apply to the 'Release Time' slider?
I guess for now I can ignore the 'Fast Compression Ratio' since it destorys transients but I'd still like to know how this plugin works.
I don't really understand the question, I fear. This plugin is really two compressors working alongside each other.
I use vlevel when compressing my music to vorbis to listen at work (I'm a projectionist, and the projectors make an surprising amount of amount of noise.) I also use it at home; It's nice 'cause you don't have to mess around with the volume nearly as much. It's really handy with some classical music that has what might be considered an "excessive" amount of dynamic range, with some parts whisper quiet, and other full-blast loud.
Sorry to revive such an old topic, but (I'm still using Foobar 0.8.3) I'm playin' around with the dynamics compressor and I'm seeing the same 'problem' as previous posters (oudalrich & 2Bdecided).
On a track which start with a low volume and the suddenly becomes much louder, I got the good behaviour: the first part volume is increased and when arriving to the louder part, the perceived volume is still the same.
On a track that starts with a louder part then a quieter one, the volume doesn't increase (or very few) when arriving to the quieter one.
I tried different settings, without success. Did I miss something or is it a limitation of this plugin?
I was resurrecting this old thread to see if I could find a basic explanation for each of the sliders in this DSP. Also, especially what settings have users found work best. I've seen a lot of posts talk around settings, without actually seeing any suggested settings posted.
I am trying to make my music "louder" in soft passages for use in the car. Sort of an FM radio effect. At home I always use all dynamic range available.