I have tried to compile new lame 3.100 on macOS 10.13, but fails with the following portion:
libtool: link: gcc -dynamiclib -o .libs/libmp3lame.0.dylib .libs/VbrTag.o .libs/bitstream.o .libs/encoder.o .libs/fft.o .libs/gain_analysis.o .libs/id3tag.o .libs/lame.o .libs/newmdct.o .libs/presets.o .libs/psymodel.o .libs/quantize.o .libs/quantize_pvt.o .libs/reservoir.o .libs/set_get.o .libs/tables.o .libs/takehiro.o .libs/util.o .libs/vbrquantize.o .libs/version.o .libs/mpglib_interface.o -Wl,-force_load,../libmp3lame/vector/.libs/liblamevectorroutines.a -Wl,-force_load,../mpglib/.libs/libmpgdecoder.a -lm -install_name /usr/local/lib/libmp3lame.0.dylib -compatibility_version 1 -current_version 1.0 -Wl,-single_module -Wl,-exported_symbols_list,.libs/libmp3lame-symbols.expsym
Undefined symbols for architecture x86_64:
"_lame_init_old", referenced from:
-exported_symbol[s_list] command line option
ld: symbol(s) not found for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)
make: *** [libmp3lame.la] Error 1
make: *** [all-recursive] Error 1
make: *** [all-recursive] Error 1
make: *** [all] Error 2
lame 3.99.5 can compile fine. I don't know what is wrong. Please help me!!!
Your encoder settings should not matter, unless you check 'Convert lossless files' or 'Convert these lossy file types'.
Whats the console output when that happens?
As a workaround you can set some folder on your computer as the destination. Then, copy the contents of that folder to your android using windows explorer. Do playlist copy without issue when done this way?
Although you requested 128 kbps, the download segment is actually 175 kbps, presumably trying real hard to be transparent.
I'm trying to set the background of my ELPlaylist as an artist picture that I store only in the root folder of each artist, not under each album, so the path to it is usually (%path%)\..\artist.* . But sometimes I need to go a level higher, when an album I'm currently playing is divided into multiple CDs, where each of them have a separate folder. And sometimes it's in the same level as the song, if it does not belong to any album.
To set an artist picture, this code is being used by default:
I've been going through the list of functions but cannot find a single one that would work in my case. What I tried was to check for the file size- if it's greater than 0, or a certain small number, it picks the picture. Since there is probably no way to make a 4-way elseif-like statement, I thought of putting several of those checks under each other, like this (except I have no idea how to get the %filesize% from that selected artist file):
$set_ps_global(do.artist.pic,$ifgreater($directory_path(%path%)\..\artist.*,0,$directory_path(%path%)\..\artist.*))Any idea about a solution that might work here? And how to check for if a certain file exists?
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 in this respect 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. (OK, the wiki table is meant to provide a start-off, but I'm not sure whether everybody reads this. Even for a start-off the lowest recommended bitrate should be 80 kbps rather than 96 IMO).
(@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.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 preprocessorA 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.
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
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):
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?
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.