Skip to main content

Recent Posts

Opus / Re: Opus killer @128 kbps
Last post by halb27 -
Would you mind trying --bitrate 140?

(I have 2 tracks I care about which I can't really ABX at --bitrate 128 with 'reasonable effort', but I repeatedly got at 6/8. Too low a safety margin for my taste. Using --bitrate 140 I get real random ABX results.)

The wiki recommendation isn't wise to me either. We have seen in another thread that --bitrate 80 yields results which are fine usually. Even --bitrate 64 is very usable for a good quality. And if someone wants a setting which is fine also in rare situations -- bitrate 128 isn't a bad choice, but a somewhat higher bitrate maybe useful. What an exact bitrate to use depends on personal preference and personal judgement about those tracks requiring a higher bitrate (relevance and quality wise).
I'd prefer a recommendation for good quality music which reads like 64 ... 160 kbps.
Last post by Porcus -
(@eahm: If placebo is so amazing, why do we try to ruin it all the time? :-D )

Archiving in wav? That is also a nice way to give "bit rot" a free pass to your sound data without noticing it at first.
Flac uses checksums to ensure data is OK, and it has tagging and embedding functionality. It also mutes when a bad block is fed into the decoder.
I have had a USB drive with a defective USB controller on the motherboard.  It would lose the disk while it was writing, corrupting the files. That happened in a tag update on ... what file? I could check the FLAC files. And restore from backup. Of course, this happened to my "live working" set, as I was tagging, not to the backup set.

PS. FLAC is always lossless. Not to make things more complicated, but there does exist a way to losslessly archive into FLAC, that is lossywav as a preprocessor
A typo: there should be "lossy", not "losslessly". But that is, as you say, lossless-compression of a signal that first is processed into lossy.
And for the nitpickery: one cannot use FLAC to compress 32-bit floating-point PCM. There are such .WAV files around, some artists upload them. WavPack can contain 32-bit floating-point, FLAC cannot.

AAC - General / Re: Which encoder to use today?
Last post by Nichttaub -
I found that on my most demanding recordings I can barely ABX 128k AAC encodings from the CD original.  I have been using 256k VBR just so that I don't even worry about it, and have never been tempted to look back.
Trying to figure out how to have the foobar media server output files with multiple tags.

For example I just added my mp3 folder of "Frozen" the movie soundtrack to my library. All the files have multiple tags with multiple artists for example they all have "Disney" as the artist but then also they all have the individual singer as well.

When I look them up in foobar (view by artist), all is as expected and well. For example the song "Let It Go" comes up in two places:
Disney-->Frozen-->05. Let It Go
and also
Idina Menzel-->Frozen-->05. Let It go

However this does not happen when looking things up via the UPnP Media Server, it doesn't split them up properly and show them as separate artist entires. How can I change/modify or add to the media server code to get it to display correctly? The code I am using is listed below (only the artist section is listed):

<SubTree label="Artists">
<spec type="object.container.person.musicArtist">%artist%</spec>
<spec type="object.container.album.musicAlbum">%album%</spec>
Not sure I got it but I hadn't thought of this. So on ELPlaylist I stored all tracks by playcount, removed the ones with 0 playcount, and then sorted all tracks again by album. The result includes, for example, albums of 10 tracks, showing with only 3 tracks (obviously because I haven't heard the remaining 7 ones) I don't want these on the list because these are albums I haven't heard entirely. Is there any way to quickly weed these cases out or will I have to, for each one, "open containing folder" > see how many tracks there are > manually remove it from playlist if it has more tracks than I've listened to?
AAC - General / Re: Which encoder to use today?
Last post by AlexQc -
Thanks everyone for all you time. My choice is made and I did some test CDs to test it.

I will use iTunes AAC encoder via qaac (under EAC). This is the qaac command line I'm using:
--no-smart-padding -v%bitrate% -q2 %source% -o %dest%

The bitrate I will be using is 192. So to be short I will be using CVBR at 192 kbps.

Do you see anything obviously missing about the command line I use? I'm tagging my files manually with an external program.

PS: I will also keep a FLAC master for my desktop computer.
So I'm not sure where to report this issue but this is the only page I'm finding for the bs2b library.

When trying to build this using the mingw cross compiler on Ubuntu 16.04 for ffmpeg using the ffmpeg build helpers script, one gets an error:

/ffmpeg/sandbox/cross_compilers/mingw-w64-i686/i686-w64-mingw32/lib/libbs2b.a(bs2b.o):bs2b.c:(.text+0xd): undefined reference to `rpl_malloc'
collect2: error: ld returned 1 exit status
After googling the heck out of this, I found here ( that AC_FUNC_MALLOC is not a cross compiler friendly option. After removing AC_FUNC_MALLOC  from it started compiling okay.

I was wondering if the owner/author can make this correction to the file to make it cross compiler friendly
No, I don't want to modifiy the files.
Opus / Opus killer @128 kbps
Last post by Steve Forte Rio -
3 years ago one of the users from my forum published the ABX log (russian) which shows that he hear difference between lossless and opus -b 128. As I understand that was opus 1.1 (October 2014)

Now I tried that sample with libopus 1.2.1 (also -b 128) and here's the result:

Spoiler (click to show/hide)

The difference is almost at hearing threshold but is perceptible and sounds like some additional high-frequency noise (Opus sounds with more sharpness than original sample).

The sample: 02_the_riddler_hide_and_seek.flac
Encoded sample: 02_the_riddler_hide_and_seek.opus

As I read in Wiki, Opus recommended bitrate for music is 96—128 kbps, but why it's restricted there to maximum of 128? Maybe it should be raised to ~160 kbps?
mp3gain doesn't write replaygain tags as such.  It modifies the internal gain settings for the audio.  It also, or just, writes tags indicating what changes it made so that they can be undone at a later date.  The tags are not the same as the replaygain tags that have come to be a standard now, and are not likely to be recognised by any decoder you are using.  The (reversibly) modified audio will be played back at the adjusted gain by any decoder.  It is worth noting that the gains from mp3gain are not very accurate compared to modern replaygain tags.