I must admit I have no idea under which circumstances the Advanced > Standard Sort Patterns settings are used. That is a curious one.
To sort the playlist, it depends which library viewer you are using to send files to the playlist. Each one has its own sort string...
Windows Explorer - If you drag and drop files/folders from outside foobar, the Preferences > Shell Integration > 'Sort incoming files by' setting is used. So you would paste the string there.
Facets - Has its own sort string in Preferences > Media Library > Facets > 'Default sort order'. Paste the string there.
Album List - The current view pattern is the playlist sort order. So if using "by artist/album" view, you would edit the view in Preferences > Album List > unders Views. This one ideally needs to keep the specific tree branch format though, so just replace the one or two fields. For example, where the pattern has %album artist%, replace that with $swapprefix(%album artist%) - which also has the effect of changing the display of artists as well.
Other library viewers probably have their own sort settings somewhere, too.
To change the display (and sort, if clicked upon) of individual columns in Facets or the playlist, you would of course edit the column pattern. Once again, replacing %album artist% with $swapprefix(%album artist%) - etc. Facets preferences also has a check box to 'Ignore leading The when sorting' which might be preferred, as it displays the artist name intact while sorting alphabetically. The same can be achieved for any other prefix using a custom sort column in the same pane and dragging it out of view.
May I ask how can the peak of an album of FLAC tracks be higher than 0dBFS ?
It must calculate so called true peak. Intersample peaks can exceed digital fullscale by a few dB.[/list]
Each sample cannot exceed 0dBFS, but the samples may not correspond to the peaks of the reconstructed waveform . Hence the peaks can exceed 0dBFS, even with a lossless codec.
True peak should be the default in r128gain (when in R128 mode), although the album peak looks suspicious. Performance varies dramatically depending on whether true peak is being measured or not.
Last post by fbuser -
Although not impossible writefile is probably not suitable for your purpose, as you can only write one single value for each record to a different file with it.
But you can just do a "SELECT * FROM MediaLibrary" in the SQLite console, write click on the result and copy all. You will get the content of the result window in a CSV format (separator is a comma and field delimiter a single quote) copied to the clipboard.
You might need to increase the SQLite console row limit in "Preferences -> Advanced -> Tools ->SQLite viewer" to get all your data listed. In case you have a large library you also might consider to split the result sets to a reasonable size to not run into memory problems.
Tags and values for a FLAC file: The Great Gig in the Sky // The Dark Side of the Moon /// Pink Floyd
Tags
loudgain
r128gain
Remarks
Album replay gain peak
1.013261
1.000000
r128gain rounds the values to 1 decimal place (unless the value is below 1) and pads the value with 0 for the number of places required by the ReplayGain v2 specification.
loudgain tries to be as precise as it can be.
May I ask how can the peak of an album of FLAC tracks be higher than 0dBFS ? I would actually say that loudgain cannot be correct here. Of course, I cannot be sure if what is seen here is the incorrect usage of ffmpeg or something else.[/list]
Last post by magicgoose - https://pypi.org/project/r128gain/ there's an okay Python lib for that, should work on GNU/Linux just fine (but not only on it) my wife tested it a while ago, it seemed to be able to write RG tags in Opus files correctly (in a way that foobar2000 understands). -- oops, this was already posted, sorry so, yeah, I can confirm that r128gain is just fine
Last post by rohbear -
I have had the same thing happen to me. It's not consistent, though. I have had great responses (and quickly) to some questions, while others are viewed but not answered. Maybe it's all about how the question is worded?
Last post by darkflame23 -
Been thinking about this a lot again recently, would really appreciate native VST plugin support, any chance it's even on the cards?
We are currently developing a pair of over ear noise canceling headphones and we want to make sure that voice call quality and background noise canceling gets as big of a priority as audio performance. As that's where most headphones lack nowadays.
We are looking for a lab or some company that specializes on algorithms related to mics on headphones, dsps for mics etc. I was wondering if anyone has any idea of best guys in the industry for this. I know this sub is full of pros
