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Yeah... MP3Gain changes the actual volume in the file, so the CD/MP3 player doesn't have to "recognize" anything.

If you are not satisfied with MP3Gain, you might try adjusting the volumes "manually" by ear with MP3DirectCut.    That will take some time and you'll have to choose a standard reference track.  You might want to only  adjust the ones that need it after  applying MP3Gain.  And, a few songs will always be a little too-quiet unless you push the peaks into clipping.

Oh...  It's usually best to leave the default MP3Gain volume of 89dB.  If you increase the MP3 target volume and you don't allow clipping, you are not leaving MP3Gain enough "room to work" and fewer songs will be adjusted.    Even with the default, some  songs can't be adjusted-up without clipping (or can't be adjusted enough  to hit 89dB).
For example, with foobar2000, the option is hidden (apply gain to files) by default

... Shift, right-click, ReplayGain and then Apply. Then choose album or track mode. (Album, of course ;-) )
MP3 - General / Re: Lame 3.100 has been released
Last post by Jebus -
Sorry for the absence of compiles on Rarewares, but I have been away from home for the last 2 weeks and I don't return until the end of this week. I will post compiles as soon as I'm able. :)

Thanks John! Can you update the libmp3lame (x86 and x64) compiles too? I might be the only one using them, but I do use them!
Are you using the application named "MP3Gain", or you simply use a software that has replaygain support?

If the player does not explicitly support replaygain (as would be most CD car players), then you need the files themselves to be modified, and that's what the application MP3Gain does.
There are other tools that can also do this, but not by default. For example, with foobar2000, the option is hidden (apply gain to files) by default and only the replaygain option is visible.
3rd Party Plugins - (fb2k) / Re: R2YS a script for reddit
Last post by zeremy -
It it still working  with JScript Panel v.1.3.0
Can you provide an output from the console panel where it crashes?
I figured it out. Anyone looking to make any field capable of using multiple tags, this will work as long as in the foobar settings you enable that field to be able to have multiple settings:
Preferences > Advanced > Display > Properties dialog > Multivalue fields.

In case anyone else is wondering about this issue: I finally figured it out by looking at the formatting from the album list tab in the settings. Apparently the field needs to be %<artist>% instead of %artist% however it is not as simple as that in the media tree as it is in XML format so simply changing it like that wont' work.

If you open the XML file in a text editor
Instead of %<artist>%
you need to use

Hope this helps someone
3rd Party Plugins - (fb2k) / Re: R2YS a script for reddit
Last post by onirwai -
Script no longer works nowadays. I get this:

Reloading or fiddling with the config doesn't help.
MP3 - General / Re: Lame 3.100 has been released
Last post by xanadu6291 -
Remove the line containing 'lame_init_old' from the file 'include/libmp3lame.sym'

Now I can compile lame 3.100 successfully!!  I'm so happy!! :-)

Does the removed line is really needless one?
The major ways that 19kHz subcarrier leakage might hurt is if you send the signal to old analog circuitry.   Even then, -60dB is pretty low.  However, if you are seeing higher levels, then it could be a problem on electronics that has bad HF nonlinearities.   Old transistor designs (e.g. simple, three transistor gain circuits without appropriate management of HF range in the design -- and other such situations) can happily and every effectively produce audible intermodulation distortion (not necessarily as bad as having birdies, but simply an obvious edge to the sound.   Also, old tape gear didn't like strong levels of HF (again, -60dB is okay, -20dB might start causing problems.)
In the digital world, it is VERY easy to get rid of the 19kHz signal, and frankly, if it bothers you for some appropriate reason, then just nuke it with a nice filter.  If you use Sox (I live in mostly a command line world, since I am an EE/software person, not a music professional), then just add the option 'sinc -15k' or 'sinc -14.5k'.   These commands will do a sharp filter without adding terrible phase or peaking issues.  (Sox is free and even though it has a fairly ugly command line, it generally works fairly well.)   So, if you are using sox, then the following command might be useful:

sox infile.wav outfile.wav sinc -15k

I use sox on Windows and on Linux, and is a nice little swiss army knife that doesn't require firing up a big GUI to do a simple task.

John Dyson
MP3 - General / Re: Lame 3.100 has been released
Last post by robert -
Remove the line containing 'lame_init_old' from the file 'include/libmp3lame.sym'