Hello list!
Off topic, but maybe some of you guys got any advice?
I've mainly been using Thunderbird and Evolution as my mail readers,
because they have powerful search capabilities and good folder and
account management. But I find both seriously lacking in contact mangement.
Editing contacts require loads of clicks and mouse navigation to enter
info to those fields I use most often:
Name, email (might be multiple), phone (might be multiple), Street
Address, Organization / Company, Birthday, Notes.
I'd like a spreadsheet approach instead: a simple list view of the
preferred columns, right click to get a "Mail to" context and double
click in a field to edit. Or something similar. And keyboard navigation,
please.
I want sync across devices (that is Android phone, PC and occasional
webmail) but I don't want Google, so I'm using a CardDAV address book.
Thunderbird doesn't even let me use cardDAV as default address book but
saves all contacts to "Personal address book". Evolution opens contacts
in a tabbed window without CTRL-TAB navigation. Etc.
So. Does anyone here have pointers to mail readers with powerful search,
good folder and account management AND good contact management?
I'll happily accept personal mail to keep the OT clutter minimal.
Thankful for any input.
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
Gotcha.
The format isn't going to matter much, so just convert the FLAC to some
lossy codec and you're done. None of the lossy codecs are close to an
order of magnitude better than any of the others, especially if you are
space-limited.
On Sat, Apr 15, 2023 at 19:50, D.T. <danter(a)posteo.de> wrote:
> I really don't think you're getting the problem:
> I want to stream from a remote server that has limited storage.
> Whether the software can do transcoding on the fly is beside the
> point because as it currently is
> there isn't enough space on the server, therefore I must do
> /something/ before I upload.
>
> The ability to then listen to the stream is not under debate. That
> part is covered.
>
> To simplify, I'd really like an answer to the question how I can best
> reduce my music collection to a unified, much reduced format.
>
> On Sat, 2023-04-15 at 13:40 -0600, Paul Davis wrote:
>> both LMS and airsonic can do that
>>
>> On Sat, Apr 15, 2023 at 19:32, D.T. <danter(a)posteo.de> wrote:
>>> It's a virtual server.
>>> I want to stream while mobile also, not only at home.
>>>
>>>
>>> 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
>>>> _______________________________________________
>>>> Linux-audio-user mailing list --
>>>> linux-audio-user(a)lists.linuxaudio.org
>>>> <mailto:linux-audio-user@lists.linuxaudio.org>
>>>> To unsubscribe send an email to
>>>> linux-audio-user-leave(a)lists.linuxaudio.org
>>>> <mailto:linux-audio-user-leave@lists.linuxaudio.org>
>>>
>> _______________________________________________
>> Linux-audio-user mailing list --
>> linux-audio-user(a)lists.linuxaudio.org
>> <mailto:linux-audio-user@lists.linuxaudio.org>
>> To unsubscribe send an email to
>> linux-audio-user-leave(a)lists.linuxaudio.org
>> <mailto:linux-audio-user-leave@lists.linuxaudio.org>
>
both LMS and airsonic can do that
On Sat, Apr 15, 2023 at 19:32, D.T. <danter(a)posteo.de> wrote:
> It's a virtual server.
> I want to stream while mobile also, not only at home.
>
>
> 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
>> _______________________________________________
>> Linux-audio-user mailing list --
>> linux-audio-user(a)lists.linuxaudio.org
>> <mailto:linux-audio-user@lists.linuxaudio.org>
>> To unsubscribe send an email to
>> linux-audio-user-leave(a)lists.linuxaudio.org
>> <mailto:linux-audio-user-leave@lists.linuxaudio.org>
>
Hi folks,
I'm trying to set up a fairly minimal system as a dedicated Digital Organ.
I started with Ubuntu Mate 22.04, as that's what I normally use for my
personal systems.
During the install I requested a minimal system, no Office, etc. and
ended up with what looked OK.
However, When it came to setting up the audio side, I wanted to run
Jack(D2/DBUS) and use Qjackctl
for initial setting up. I'm hoping to dispense with Qjackctl once its
all working, though the patchbay
might keep it in. I want a system that comes up headless (normally) so
the organist just switches on,
waits for a couple of minutes for it to start, then starts playing.
I'm having great difficulty in getting Qjackctl and Jack to start
reliably, I've tried jackd2 and jackdbus
(from the standard Ubuntu repositories) and Qjackctl nearly always fails
with messages saying it can't
contact jack: 'Server communications error, plesae check the message
window for more info'.
The window then says 'Cannot read socket fd = 36 err = Success' which
seems contradictory!
If anyone can help I'd like opinions on whether I should be pushing for
jackd2 or jackdbus.
The idea is to have a startup script which starts jack, then qjackctl,
then the organ software (GrandOrgue).
So far it's a mess.
Someone must know what I need to do. I'm happy to collect any
information and report back if you can
tell me what is needed.
Many thanks in advance!
Bill
--
+----------------------------------------+
| Bill Purvis |
| email: bill(a)billp.org |
+----------------------------------------+
Hey hey,
I would like to present my song "Ends of the earth" in a shorter edit with a proper video and an announcement:
https://youtu.be/f-ERmWxPFi0
The song is now entered for the "International Low-vision Song Contest 2023" (ILSC 2023). It is an accepted entry for the German national semi-final. This contest is organised by several national organisations for the blind and visually impaired.
Why do I post it here? It is possibly the only entry produced with Linux open source software, including: Csound (which gets a special mention in the introduction), LinuxSampler, Yoshimi, Midish, Nama, several LADSPA plugins, as well as free and commercial sample libraries, field recordings and purely synthesized sounds. Including My Csound handpan instrument, several risers and effects also created in Csound.
The German semi-final will air on
April 14, at 20:00 CEST (18:00 UTC, 14:00 EDT, 12:00 MDT)
The YouTube live stream will be here:
https://youtu.be/CJ5kJ1PDS7g
The official website for the whole contest is here:
https://www.dbsv.org/ilsc.html
This year's ILSC sees a greatly extended list of countries across the world joining in. From what I have been able to gather (by looking at the German entries), a broad range of genres and musical background will be represented.
For the curious, the lyrics will follow below. I would like to invite you to listen and vote.
Best wishes and enjoy,
Jeanette
*** LYRICS ***
Sprite on the ice
In a mist laden dream.
Like rain from the skies,
You touch
And are gone with the stream.
Now, here...
I feel you close
Soft like a tear,
Sharp as a rose.
[Chorus]
You're so near but far away,
Out of reach, across the seas.
Seeking you I sail astray,
I'll strive for you without cease.
Frozen and seared
By the rays of your moon.
My heart burns to dust
If I can't find you soon.
Here, now...
I make a vow.
I'll cross the earth,
Volcanoes and all... To you.
[Chorus]
You're so near but far away,
Out of reach, across the seas.
Seeking you I sail astray,
I'll strive for you without cease.
*** END OF LYRICS ***
--
* 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
I love the things you say
And I love the love your touch conveys <3
(Britney Spears)
Can anyone recommend a mid range one *known* to work with Linux, including
Jack audio?
--
Will J Godfrey {apparently now an 'elderly'}
https://willgodfrey.bandcamp.com/http://yoshimi.github.io
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
Hi all,
Shortly after 0.6.6 this release corrects some issues found in the
meantime mainly thanks to lv2lint and the old jack_interposer.so in the
last 0.6.6 version. It also introduces a cmake build environment with
the hope of making packaging easier.
Enjoy
Frank
qmidiarp-0.6.7 (2023-04-11)
New Features
o Package: a cmake build environment is now available with the goal
of less pain with libtool.
Fixed Bugs:
o fixed two small non-realtime function calls that had sneaked in last
year.
o fixed several lv2lint FAILS: lv2:resize extension was removed,
lv2ui:qt5ui is no longer exposed, globally visible library symbols
were hidden. lv2lint still reports being unable to instantiate the
uis, but all tested hosts were able to do it.
Download
https://sourceforge.net/projects/qmidiarp/files/qmidiarp/0.6.7/qmidiarp-0.6…
Github
https://github.com/emuse/qmidiarp
Website
https://qmidiarp.sourceforge.net/
Hi all,
A highly overdue release of various improvements and bugfixes cumulated
over the last five (!) years is finally out.
The main changes are in the Global store that got some handy attributes
added to the storage locations, i.e. you can now set a number of
repetitions counter. You can also click on a spot within the location
buttons to interrupt the automatic progression across locations
temporarily. This may improve its usability a bit.
Regarding LV2, there is now a X11 UI descriptor provided (thanks to Rui
rncbc for a model how to do this in synthv1), but note that the qmidiarp
plugins *UIs* are still not self-contained and link to the Qt5 libraries
that were used for building, as long as nobody (including me) feels
ready to rewrite a library-independent LV2UI for them. They were checked
working nicely with qtractor, ardour 6, reaper, carla, jalv and bespoke
synth locally, but be warned.
The plugin dsps are, however, independent of Qt.
Thanks go to Matthew McGuire for adding a phase setting to the
calculated LFO waveforms and to all of you reporting bugs and suggesting
features.
qmidiarp-0.6.6 (2023-04-08)
New Features
o LFO: New control for phase of calculated waveforms (Matthew McGuire)
o Global Store: Each storage location now has a "number of repetitions"
property accessible through the context menu.
o Global Store: Each storage location now has a "Go here and stay"
storage sub-button to force remaining at that location.
o ALSA MIDI: Sending MIDI Clock to a specified ALSA port is now
available as a new preferences option
Improvements
o LFO and SEQ: More size and resolution values including odd values
Although there is compatibility check in place for saved sessions,
this may lead to wrong values for saved LV2 states if someone
uses this at all.
Long sequences only make sense for lower resolution. Drawing would
not be possible with high resolutions and lengths.
o SEQ: Improved display of loop marker
o SEQ: Increased lower octave transpose range
o Global Store: Preferences option to choose whether to store mute states
o LV2: a X11 plugin UI interface is now available
Fixed Bugs:
o Regression: Arp strayed in notes with zero velocity
o LV2: restored compatibility with suil > 0.10.2
o Crash when session managers tried to save/restore empty session
o Regression with application of compact widget style
o Regression when reading files with time module index -1
o Output port count could not be changed on commandline
Changes:
o Jack-Session support has been removed due to "official deprecation"
Download
https://sourceforge.net/projects/qmidiarp/files/qmidiarp/0.6.6/qmidiarp-0.6…
Github
https://github.com/emuse/qmidiarp
Website
https://qmidiarp.sourceforge.net/
Hi all,
not posting to any of LA* mailing lists often in the last years, but today I thought
I should make an exception to point out that it's now exactly 20 years ago that the
first Linux Audio Conference or "LAC" for short (then still called "Linux Audio
Developer's Meeting") took place at the ZKM in Karlsruhe, Germany.
https://lists.linuxaudio.org/hyperkitty/list/linux-audio-dev@lists.linuxaud…
Wow, 20 years come and gone so quickly! It's great to see a couple of "early
adopters" are still around today here, and many new names have entered the scene
in the meantime and left/leave a lasting footprint in it.
As a quick "trip down memory lane", here's a short list of things that happened back then:
- Takashi Iwai dove into the innards of an ALSA driver
- Fernando Lopez-Lezcano introduced the PlanetCCRMA distribution, based on Fedora
- Steve Harris explained the concept of a Bode frequency shifter as a LADSPA plugin
- Paul Davis held the keynote, AND spoke about JACK, AND shared his experience of writing
a DAW that is at the forefront of free, open-source and cross-platform DAWs today.
- Dave Philipps looked at the historic development of Linux audio, from OSS to ALSA and beyond
- Jörn Nettingsmeier managed to get the audio part of the presentations both recorded AND live-streamed
at a time when "streaming" was still a term unknown to most of us.
- The term "Linux Sound Night" already existed, but was..perhaps not as musical yet as you might have expected :-).
- And, we also learned that schedules are not easy to keep - I believe after the first day we
ran some >2 hours late, and people begin to STARVE.
- Posing in front of the ZKM "Kubus": https://linuxaudio.de/LAC2003_Posse.jpg
For some more memories, see also http://lac.linuxaudio.org/2003/zkm/
Of course, on the downside it has to be noted that Corona has impacted us too -
after an almost perfect track record of conferences or mini-conferences every year,
2021 and 2022 didn't see any event happening. There is a certain risk now that it
won't recover, but that's in our hands to change (sidenote: I have talked to my
contacts at ZKM here in Karlsruhe in January, but they are unable to finance and/or
organize an event of this size at the moment - certainly somewhat influenced by
current economics, but I am hearing ZKM has actually reduced their headcount quite
a bit, and after the recent passing of its artistic-scientific chairman, Peter
Weibel, it will have to be seen how his successor, Alistair Hudson, will steer
it into the future).
Whatever LAC's future will be, free and open-source audio software is certainly
flourishing, and will continue to do so. It just would be soo nice to enjoy the
results together with real, tangible people :-\.
If anyone sees an option for hosting a future LAC, lac-team(a)lists.linuxaudio.org
is willing to listen to proposals you might have (or just discuss it right here).
Well, the heck - there is no steering committee or anything in place, so I guess
the first one brave enough to say "I think we can do it" will get the job :-).
I am forever grateful to all the folks who have enriched and continue to enrich
our open-source audio life by writing or presenting software, creating
documentation and tutorials, hosting (AND attending) conferences like the LAC,
and whatever else helps to keep the penguin dancing!
Greetings,
Frank