15 Apr
2023
15 Apr
'23
9:07 p.m.
Hello,
I have a server with limited storage that I want to run a private radio
station from, a randomized mix of my complete music collection.
Locally I have about 80G of music in all sorts of formats, codecs and
bitrates.
This is way too large for the server's storage, I can use half of that
at best.
Additionally I don't want the stream to have too much bandwidth so it
will work even over flaky (mobile) network connections.
My thought is to transcode all of it to the same reduced format, then
upload.
That way the music server could just push it out without transcoding
again (and I could still listen to separate tracks remotely).
The Big Question:
Which format should I choose?
I found these 2 articles that seem to have an answer:
https://wiki.hydrogenaud.io/index.php?title=Transcoding#Lossy-to-lossy_tran…
https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio
Combined, it sounds to me like I really should use either FDK AAC or
Opus* at less than 100kbps (I listen to 64k AAC music streams that are
OK imo).
What do you think?
Is this even the right approach to solve the problem?
TIA!
FWIW, here's a breakdown of my music's codecs/bitrates:
vorbis: 97 files (974M), average bitrate 332kbps (from 67 to 452)
wmav2: 10 files (31M), average bitrate 131kbps (from 129 to 133)
flac: 1216 files (45505M), average bitrate 1191kbps (from 330 to 5170)
opus: 173 files (1024M), average bitrate 129kbps (from 76 to 177)
mp3: 1975 files (32823M), average bitrate 197kbps (from 96 to 420)
aac: 308 files (1592M), average bitrate 152kbps (from 64 to 334)
alac: 1 files (621M), average bitrate 768kbps (from 768 to 768)
* personally, I always had the feeling that opus (used a lot by
youtube) isn't so good with noisy, grungy, fuzzy, guitarry music
15 Apr
15 Apr
9:18 p.m.
Don't bother with transcoding.
Just use the Logitech Media Server (open source, perl!) and it can
handle all of the above.
Players for just about any device you can name, and control apps for
any browser as well as most devices.
There's also Airsonic, which is web-based and quite nice, and also
offers no rationale for transcoding to disk.
On Sat, Apr 15, 2023 at 19:07, D.T. <ohnonot-github(a)posteo.de> wrote:
Hello,
I have a server with limited storage that I want to run a private
radio station from, a randomized mix of my complete music collection.
Locally I have about 80G of music in all sorts of formats, codecs and
bitrates.
This is way too large for the server's storage, I can use half of
that at best.
Additionally I don't want the stream to have too much bandwidth so it
will work even over flaky (mobile) network connections.
My thought is to transcode all of it to the same reduced format, then
upload.
That way the music server could just push it out without transcoding
again (and I could still listen to separate tracks remotely).
*The Big Question:*
Which format should I choose?
I found these 2 articles that seem to have an answer:
<https://wiki.hydrogenaud.io/index.php?title=Transcoding#Lossy-to-lossy_transcoding>
<https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio>
Combined, it sounds to me like I really should use either FDK AAC or
Opus* at less than 100kbps (I listen to 64k AAC music streams that
are OK imo).
What do you think?
Is this even the right approach to solve the problem?
TIA!
FWIW, here's a breakdown of my music's codecs/bitrates:
vorbis: 97 files (974M), average bitrate 332kbps (from 67 to 452)
wmav2: 10 files (31M), average bitrate 131kbps (from 129 to 133)
flac: 1216 files (45505M), average bitrate 1191kbps (from 330 to
5170)
opus: 173 files (1024M), average bitrate 129kbps (from 76 to 177)
mp3: 1975 files (32823M), average bitrate 197kbps (from 96 to 420)
aac: 308 files (1592M), average bitrate 152kbps (from 64 to 334)
alac: 1 files (621M), average bitrate 768kbps (from 768 to 768)
* personally, I always had the feeling that opus (used a lot by
youtube) isn't so good with noisy, grungy, fuzzy, guitarry music
9:42 p.m.
Forgot to mention, it's a virtual server.
I want to be able to stream on the move, also, not only in my local
network.
Server storage is not so cheap (per month).
A dedicated music server will not help with the storage problem, I
really need to transcode (or elsewise reduce) before I upload.
On Sat, 2023-04-15 at 13:18 -0600, Paul Davis wrote:
Don't bother with transcoding.
Just use the Logitech Media Server (open source, perl!) and it can
handle all of the above.
Players for just about any device you can name, and control apps for
any browser as well as most devices.
There's also Airsonic, which is web-based and quite nice, and also
offers no rationale for transcoding to disk.
On Sat, Apr 15, 2023 at 19:07, D.T. <ohnonot-github(a)posteo.de> wrote:
Hello,
I have a server with limited storage that I want to run a private
radio station from, a randomized mix of my complete music
collection.
Locally I have about 80G of music in all sorts of formats, codecs
and bitrates.
This is way too large for the server's storage, I can use half of
that at best.
Additionally I don't want the stream to have too much bandwidth so
it will work even over flaky (mobile) network connections.
My thought is to transcode all of it to the same reduced format,
then upload.
That way the music server could just push it out without
transcoding again (and I could still listen to separate tracks
remotely).
The Big Question:
Which format should I choose?
I found these 2 articles that seem to have an answer:
https://wiki.hydrogenaud.io/index.php?title=Transcoding#Lossy-to-lossy_tran…
https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio
Combined, it sounds to me like I really should use either FDK AAC
or Opus* at less than 100kbps (I listen to 64k AAC music streams
that are OK imo).
What do you think?
Is this even the right approach to solve the problem?
TIA!
FWIW, here's a breakdown of my music's codecs/bitrates:
vorbis: 97 files (974M), average bitrate 332kbps (from
67 to 452)
wmav2: 10 files (31M), average bitrate 131kbps
(from
129 to 133)
flac: 1216 files (45505M), average bitrate
1191kbps (from 330 to
5170)
opus: 173 files (1024M), average bitrate 129kbps
(from 76 to
177)
mp3: 1975 files (32823M), average bitrate 197kbps
(from 96 to
420)
aac: 308 files (1592M), average bitrate 152kbps
(from 64 to
334)
alac: 1 files (621M), average bitrate 768kbps
(from 768 to
768)
* personally, I always had the feeling that opus (used a lot by
youtube) isn't so good with noisy, grungy, fuzzy, guitarry music
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9:27 p.m.
On April 15, 2023 9:07:41 AM HST, "D.T." <ohnonot-github(a)posteo.de>
wrote:
Hello,
I have a server with limited storage that I want to run a private radio
station from, a randomized mix of my complete music collection.
Locally I have about 80G of music in all sorts of formats, codecs and
bitrates.
This is way too large for the server's storage, I can use half of that
at best.
Hmm, maybe increase the server's storage? Drives are pretty cheap. My server has 5
terabytes in it and another 7 TB attached via USB3.
For your collection, a 500GB SSD at $39 (Amazon) would be worth it to me.
Additionally I don't want the stream to have too
much bandwidth so it
will work even over flaky (mobile) network connections.
My thought is to transcode all of it to the same reduced format, then
upload.
That way the music server could just push it out without transcoding
again (and I could still listen to separate tracks remotely).
The Big Question:
Which format should I choose?
I found these 2 articles that seem to have an answer:
https://wiki.hydrogenaud.io/index.php?title=Transcoding#Lossy-to-lossy_tran…
https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio
Combined, it sounds to me like I really should use either FDK AAC or
Opus* at less than 100kbps (I listen to 64k AAC music streams that are
OK imo).
What do you think?
Is this even the right approach to solve the problem?
TIA!
FWIW, here's a breakdown of my music's codecs/bitrates:
vorbis: 97 files (974M), average bitrate 332kbps (from 67 to 452)
wmav2: 10 files (31M), average bitrate 131kbps (from 129 to 133)
flac: 1216 files (45505M), average bitrate 1191kbps (from 330 to 5170)
opus: 173 files (1024M), average bitrate 129kbps (from 76 to 177)
mp3: 1975 files (32823M), average bitrate 197kbps (from 96 to 420)
aac: 308 files (1592M), average bitrate 152kbps (from 64 to 334)
alac: 1 files (621M), average bitrate 768kbps (from 768 to 768)
* personally, I always had the feeling that opus (used a lot by
youtube) isn't so good with noisy, grungy, fuzzy, guitarry music
I use mostly Ogg at highest quality, with flac (lossless), mp3/mp4.
---
David W. Jones
gnome(a)hawaii.rr.com
exploring the landscape of god
http://dancingtreefrog.com
Sent from my Android device with F/LOSS K-9 Mail.
9:40 p.m.
Forgot to mention, it's a virtual server.
I want to be able to stream on the move, also, not only in my local
network.
Server storage is not so cheap (per month).
On Sat, 2023-04-15 at 09:27 -1000, David W. Jones wrote:
On April 15, 2023 9:07:41 AM HST, "D.T." <ohnonot-github(a)posteo.de>
wrote:
Hello,
I have a server with limited storage that I want to run a private
radio
station from, a randomized mix of my complete music collection.
Locally I have about 80G of music in all sorts of formats, codecs
and
bitrates.
This is way too large for the server's storage, I can use half of
that
at best.
Hmm, maybe increase the server's storage? Drives are pretty cheap. My
server has 5 terabytes in it and another 7 TB attached via USB3.
For your collection, a 500GB SSD at $39 (Amazon) would be worth it to
me.
Additionally I don't want the stream to have
too much bandwidth so
it
will work even over flaky (mobile) network connections.
My thought is to transcode all of it to the same reduced format,
then
upload.
That way the music server could just push it out without
transcoding
again (and I could still listen to separate tracks remotely).
The Big Question:
Which format should I choose?
I found these 2 articles that seem to have an answer:
https://wiki.hydrogenaud.io/index.php?title=Transcoding#Lossy-to-lossy_tran…
https://trac.ffmpeg.org/wiki/Encode/HighQualityAudio
Combined, it sounds to me like I really should use either FDK AAC
or
Opus* at less than 100kbps (I listen to 64k AAC music streams that
are
OK imo).
What do you think?
Is this even the right approach to solve the problem?
TIA!
FWIW, here's a breakdown of my music's codecs/bitrates:
vorbis: 97 files (974M), average bitrate 332kbps (from 67 to
452)
wmav2: 10 files (31M), average bitrate 131kbps (from 129
to 133)
flac: 1216 files (45505M), average bitrate 1191kbps (from 330
to 5170)
opus: 173 files (1024M), average bitrate 129kbps (from 76 to
177)
mp3: 1975 files (32823M), average bitrate 197kbps (from 96 to
420)
aac: 308 files (1592M), average bitrate 152kbps (from 64 to
334)
alac: 1 files (621M), average bitrate 768kbps (from 768
to 768)
* personally, I always had the feeling that opus (used a lot by
youtube) isn't so good with noisy, grungy, fuzzy, guitarry music
I use mostly Ogg at highest quality, with flac (lossless), mp3/mp4.
---
David W. Jones
gnome(a)hawaii.rr.com
exploring the landscape of god
http://dancingtreefrog.com
Sent from my Android device with F/LOSS K-9 Mail.
_______________________________________________
Linux-audio-user mailing list --
linux-audio-user(a)lists.linuxaudio.org
To unsubscribe send an email to
linux-audio-user-leave(a)lists.linuxaudio.org 
9:43 p.m.
Apr 15 2023, D.T. has written:
...
My thought is to transcode all of it to the same
reduced format, then
upload.
...
I seem to remember that there is an algorithm to support low quality
OGGs, which is quite good. I'm not sure if ffmpeg supports it, but I
believe it would. Perhaps you could use one of ffmpeg's audio
effects/filters to further compress/limit the audio to get the best
milage.
This is rather vague and I'm sorry about that, yet I thought some of
that might be helpful as a further step on your way to a good solution.
Best wishes,
Jeanette
--
* Website: http://juliencoder.de - for summer is a state of sound
* Youtube: https://www.youtube.com/channel/UCMS4rfGrTwz8W7jhC1Jnv7g
* Audiobombs: https://www.audiobombs.com/users/jeanette_c
* GitHub: https://github.com/jeanette-c
Here I go, on my own now <3
(Britney Spears)
16 Apr
16 Apr
12:58 a.m.
On Sat, 15 Apr 2023, D.T. wrote:
The Big Question:
Which format should I choose?
Opus is suposed to be very good at dealing with poor bandwidth or
dropouts. That said, even though it is one of the defaults(standard
formats) for webm, I think there are a lot of systems that don't have the
codec installed. So, Ok for local use where you control the os receiving
the stream, maybe not so good for wide distribution if that is what you
are doing. Given what you have said below though, maybe that is no longer
true.
* personally, I always had the feeling that opus (used
a lot by youtube)
isn't so good with noisy, grungy, fuzzy, guitarry music
Well, don't use something you don't like :)
I would almost suggest to try them all. I would think changing the codec
is just a change the parameter, restart kind of thing. It doesn't have to
be on the target machine for testing, just use which ever machine has the
files on (or at least a good chunk of them).
--
Len Ovens
www.ovenwerks.net
3:02 a.m.
On Sat, Apr 15, 2023 at 03:58:12PM -0700, Len Ovens wrote:
On Sat, 15 Apr 2023, D.T. wrote:
To see what a given input file sounds like after being transcoded to
each of several different output formats, at a couple of different
bitrates in your range of interest, just run a for loop in Bash, no?
Something like (untested):
for i in {m4a,ogg,opus}; do
for j in {64k,96k}; do
ffmpeg -i infile.flac -b:a "$j"
outfile_"$j"."$i"
done
done
Tweak this for more detail if desired.
Then listen to the results, and go with whichever combination of bitrate
& codec you find most satisfying?
Sam
* personally, I always had the feeling that opus
(used a lot by youtube)
isn't so good with noisy, grungy, fuzzy, guitarry music
Well, don't use something you don't like :)
I would almost suggest to try them all. I would think changing the
codec is just a change the parameter, restart kind of thing. It
doesn't have to be on the target machine for testing, just use which
ever machine has the files on (or at least a good chunk of them).
7:23 p.m.
Good advice, and I did just that.
First I created various 25s audio samples, then:
br=64000 # 64kbps
for track in *.*; do
# trac.ffmpeg.org/wiki/Encode/AAC#fdk_aac
ffmpeg -i "$track" -vn -ac 2 -c:a libfdk_aac -profile:a aac_he_v2 -b:a $br
-vbr 5 "${track}.HEv2_$br.aac"
ffmpeg -i "$track" -vn -ac 2 -c:a libopus -b:a $br -vbr on
"${track}.vbr$br.ogg"
done
The results are equal in size, but in quality Opus clearly wins in this
scenario.
Big Question No. 2:
Is it possible to stream in VBR?
On Sun, 2023-04-16 at 01:02 +0000, Sam Kuper wrote:
On Sat, Apr 15, 2023 at 03:58:12PM -0700, Len Ovens
wrote:
On Sat, 15 Apr 2023, D.T. wrote:
To see what a given input file sounds like after being transcoded to
each of several different output formats, at a couple of different
bitrates in your range of interest, just run a for loop in Bash, no?
Something like (untested):
for i in {m4a,ogg,opus}; do
for j in {64k,96k}; do
ffmpeg -i infile.flac -b:a "$j"
outfile_"$j"."$i"
done
done
Tweak this for more detail if desired.
Then listen to the results, and go with whichever combination of
bitrate
& codec you find most satisfying?
Sam
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Linux-audio-user mailing list --
linux-audio-user(a)lists.linuxaudio.org
To unsubscribe send an email to
linux-audio-user-leave(a)lists.linuxaudio.org * personally, I always had the feeling that opus
(used a lot by
youtube)
isn't so good with noisy, grungy, fuzzy, guitarry music
Well, don't use something you don't like :)
I would almost suggest to try them all. I would think changing the
codec is just a change the parameter, restart kind of thing. It
doesn't have to be on the target machine for testing, just use
which
ever machine has the files on (or at least a good chunk of them).
581
days inactive
582
days old
linux-audio-user@lists.linuxaudio.org
8 comments
6 participants
participants (6)
-
D.T. -
David W. Jones -
Jeanette C. -
Len Ovens -
Paul Davis -
Sam Kuper