Having sorted out my previous problem with Jack, I've got round to
trying to sort out the multi-channel outputs
via a MiniDSP UDAC-8. This is an 8-channel USB output device. I plugged
it in and it shows up with 'aplay -':
---------------------------------
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: ALC887-VD Analog [ALC887-VD Analog]
 Subdevices: 1/1
 Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 1: ALC887-VD Digital [ALC887-VD Digital]
 Subdevices: 1/1
 Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]
 Subdevices: 1/1
 Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1]
 Subdevices: 1/1
 Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2]
 Subdevices: 1/1
 Subdevice #0: subdevice #0
card 1: UDAC8 [U-DAC8], device 0: USB Audio [USB Audio]
 Subdevices: 1/1
 Subdevice #0: subdevice #0
------------------------------
Going into QjackCtl and changing the output device to hw:UDAC8 and it
seems to work - jack restarts and the Graph
window now shows system with 8 playback channels. I then tried using
jack-play to play a short test .wav file
which works fine on the ALC877 but fails with the UDAC-8:
------------------------------------------
Cannot read socket fd = 8 err = Success
Cannot open jack-play-3712 client
CheckRes error
JackSocketClientChannel read fail
JackShmReadWritePtr1::~JackShmReadWritePtr1 - Init not done for -1,
skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1,
skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1,
skipping unlock
jack_client_open() failed: jack-play-3712
-------------------------------------------
I then tried using just alsa
aplay -D plughw:UDAC8
which works, while
aplay -D hw:UDAC8
fails with:
------------------------------
Playing WAVE 'test1.wav' : Signed 16 bit Little Endian, Rate 48000 Hz,
Stereo
aplay: set_params:1349: Channels count non available
------------------------------
qjackctl only allows devices with hw:* and not plughw:* so that could be
something, though I don't
understand the internals of jack and alsa.
Anyone able to set me right and get this to work will win my undying
gratitude!
Bill
--
+----------------------------------------+
| Bill Purvis |
| email: bill(a)billp.org |
+----------------------------------------+
> Hello all,
>
> Since four days now, the melody I try to reproduce below is
> haunting my brain. Don't know how i picked it up.
>
> It's a popular classical piece which I have heard many times,
> from the romantic period and as I remember it, for string
> orchestra.
>
> But I can't find out what it is. Tried some of the 'find a
> melody' sites, no succes.
>
> 6/8 (or maybe 3/4)
> - = continuation of previous note by 1/8
>
> | A-- G#-- | A---- B | C-- B-- | Bb----- |
> | Bb-- A-- | G----- | F-- G Bb A | Bb----- |
>
> Can anyone identify this ??
>
Doesn't ring a bell here, figuratively speaking. But maybe you can
find it via http://www.themefinder.org/ ?
I couldn't find it with themefinder though, but I'm not sure if I used
it correctly.
Thank you both for your suggestions. Having experimented further, I
find that, as I expected, the earpieces are completely inaudible when
they are not in my ear (they are just the little ear-bud type of
thing).
The default connections shown bu aJackCtl are:
System(capture) to qTractor and PulseAudio JACK Source
qTractor and PulseAudio JACK Sink to System(playback)
This seems about right, to me, I experimented with removing the
connections to/from PulseAudio, but this doesn't make any difference to
the problem I have been getting. I also experimented with muting
various things in Pavu, without any success.
So either I am doing something very basic completely wrongly (quite
likely!), or perhaps there is some sort of crossover going on in my
sound card between input and output.
I did attempt to repeat my experimental recording on my desktop
computer, but couldn't figure out how to get the input from my webcam's
microphone into qTractor. When I start Jack, the System(capture) node
doesn't appear in the connections graph at all.
David
Several years ago I managed to produce some multi-track midi files
using qTractor.
Now I want to do something fairly simple using audio tracks. Â Just to
get myself back into using qTractor after a very long break, and to
find out how to do what I want, I decided to try something very simple
to start with. Â I want to record a guide track using my voice, and then
record 3 other voice tracks (separately), listening to the guide track
through an earpiece. Â Then I shall delete the guide track and
experiment with altering the pan settings on two of the other 3 tracks
to see if I can get a good stereo effect. Â I'm using an HP laptop and
its internal microphone, with an earpiece plugged into the computer's
headphone socket; no sound comes from the computer's speakers, of
course.
Recording the guide track is no problem. Â To record the second track, I
set it up as an unmonitored audio track and set the guide track to
solo. Â I then record the new track whilst listening to the guide track
through the earpiece.
But after recording the second track, even if I delete the guide track,
I can still faintly hear what was on the guide track - in other words,
the new track has recorded the contents of the guide track at the same
time as the new input from the microphone. Â I have tried this several
times, and I can't figure out what I'm doing wrong - probably because
I'm a musician and not a sound engineer.
If anyone has the patience to try to steer me though this, I shall be
extremely grateful.
David
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.Â
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 |
+----------------------------------------+
