I have several soundcards for USB. One of them doesn't initialize on
boot or reboot. I have always to unplug and replug the cable USB
manually that Fedora25 recognizes and initializes the soundcard in USB
and PulseAudio. This is annoying.
What does the same as manually unplug and replug an external USB
device(soundcard)? How to do this? Is there a command for
reinitializing USB devices?
Is this done by triggering UDEV?
Rgds AW
Very pleased to see these up so quickly. It's always difficult to decide which
presentations to see on the day.
Were the workshops recorded as well?
Thanks to Laurent and the team. The whole event went with impressive smoothness.
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
While looking over old archives it suddenly occurred to me that it's exactly 10
years since my first submission to the KVR cafe contest. Not wanting to pass up
on that, I quickly grabbed the old files (yes, I do keep stuff that long) and
added a repeat with a new arrangement. Even now it's very short - under 2:30 in
total.
So here it is. The original made with Zyn, and the latest with Yoshi.
Hope you like it.
https://soundcloud.com/soft-sounds/soft-cheese-aniversary
--
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.
d.adler(a)posteo.de:
>
> On 21.05.2017 14:14, Kjetil Matheussen wrote:
> ...
> >
> > By using jack_capture instead of ecasound, this should be enough:
> >
> > jack_capture --jack-transport
> > jack-smf-player -n foo.mid &
> > sleep 5
> > echo locate 0 | jack_transport
> > echo play | jack_transport
>
> I also had jack_capture in mind first, it always stopped though whenever
> a sleep command finished that had been issued after it. Thus, being
> lazy, I used something else instead of investigating this.
> It turns out that jack_capture's --no-stdin option does the magic.
>
>
That's interesting. Maybe it has something to do with the piping.
Another alternative would be to open
jack_capture in a new terminal. Then you also
avoid cluttering the terminal with text output
from both jack_capture and jack-smf-player:
xterm -e jack_capture --jack-transport &
jack-smf-player -n foo.mid &
sleep 5
echo locate 0 | jack_transport
echo play | jack_transport
Hello audio users,
this is a "I have a .mid and I want an .ogg", Aeolus edition.
Longer version: I use Aeolus organ emulator with my midi keyboard and
it works like a charm.
Today I wanted to grab a midi file and make Aeolus play it, recording
the result in an ogg. I managed to do it but boy it was cumbersome;
here is what I did:
1. aeolus -A
2. aconnet 14 128 # midi through to aeolus
3. fire up alsamixer and switch CAPTURE from Front Mic to Stereo Mix
4. arecord foobar.wav
5. aplaymidi -p 14 somemidi.mid
Lots of plumbing to do! I am particularly not happy with step 4 (alsamixer)
because I have to do it manually and I cannot just put it in a bash
file.
So I ask to you: "is there a simpler way"? Thanks in advance
-Francesco
The subject is a reference to Paul Davies talk at LAC2017
(https://youtu.be/tZbMfyBGPKo?list=PLpEMKw8jHgdRA3ydjZvt73q4Nc4W0jvuz&t=11804)
I happened to come across a user interface for a DAW recently in a
Swedish magazine for home studio (online). The program is called Amped
Studio and is a cloud-based version of an DAW. Seems to be a sensible
user interface, maybe a bit simple for an DAW. But it could be how a
"The Future" might look like.
https://www.ampedstudio.com/
launch the app.
https://app.ampedstudio.com/
best,
--
/bengan
I realized that my RME 9652 has 2 channels of output via spdif, which can
be configured to go out on the ADAT1 optical interface! This seems like a
much better solution than using 2 sound cards, as long as I only need a
couple more channels of output. But I'm having trouble understanding how
to make this work. I ran hdspconf and checked the box for "SPDIF out" on
ADAT1. But which channels do I use, like in Jack? I see 26 system outputs
in Jack Connections. Two of those must be the spdif outs; is it 25 and
26? Do I disconnect my optical cable from the RME 9652 adat1 port, the one
that currently goes to my adat d/a box, connect it to the adat2 port
instead, and then connect the RME 9652 adat1 port to my spdif d/a device?
Sorry for being so clueless...
Note also that I did find a way, more than one way I think, to use 2
separate sound cards. I can configure Jack to use a "dummy" sound card,
and In Ecasound can write directly to the different hardware devices.
Seems to work fine, though awfully non-standard. Or I can use an alsa
application that makes the 2 sound cards look like one alsa device. But,
until/unless I need more channels, I would much prefer to use the spdif on
the rme 9652.
Thanks very much for helping,
John
On Thu, May 18, 2017 at 3:48 PM, Jörn Nettingsmeier <
nettings(a)stackingdwarves.net> wrote:
> On 05/17/2017 08:06 PM, john gibby wrote:
>
>> Hi,
>> Just a little question - I have an RME9652 connected to a Tango24 to give
>> me 8 d/a outputs via ADAT; but I really need 9 outputs. I only really need
>> ONE input, for testing/crossover design. The 9 outputs I need are: front
>> L/R low/mid/high, center, L/R surround.
>>
>
> Hmm. You mean you want active three-way on left and right, but run the
> center fullrange? Nothing against passive designs, but I don't think it's
> wise to have a vastly different design for the center (since the rest of
> your mail sounds like you're after high quality and detailed control). So
> maybe upgrade to 16 ADAT outs anyway, to ease the pain of missing just one
> extra out? ;)
>
> I understand you are not going to connect the speakers via ADAT (and if
> you are, you might want to reconsider...), but there's gonna be a DA box in
> between, right?
>
> You could abuse the analog outs, or use the SPDIF out pair for the
> surrounds... if you need sample accuracy in the lab, it might be wise to
> measure as the output latency between ADAT and SPDIF/Analog might differ,
> but in practical use it will be negligible.
>
> So I'm wondering the nicest way to do get my 9 ADAT outputs. Maybe I
>> should just look for a good deal on another Tango24... or, get a Behringer
>> 8000 d/a for less money, and use it for the L/R surround channels.
>>
>
> If you go the Behringer way, get the new one (8200 I believe), they fixed
> the power supply, so the thing is less likely to die on you in a bad
> moment...
>
> Both those options give me 8 more inputs that I don't really
>
>> need right now... Is there something reasonable I might buy (and sell my
>> Tango24), that has ADAT, with from 10 to 16 outputs and 8 or less inputs?
>>
>
> With these simple options like the ADA 8200, you really can't save much by
> leaving out the inputs. If you want to go up a notch, there's the DirectOut
> Andiamo 2 DA (no inputs), but I guess that thing is around 2k euros, and it
> only takes MADI input. And then there's the Ferrofish boxes, quite
> full-featured with ADAT and MADI I/O in addition to all the analog lines.
>
>
> --
> Jörn Nettingsmeier
> De Rijpgracht 8, 1055VR Amsterdam, Nederland
> Tel. +49 177 7937487
>
> Meister für Veranstaltungstechnik (Bühne/Studio), Tonmeister VDT
> http://stackingdwarves.net
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user(a)lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
I am interested in AVB (now called TSN) since some time ago.
I have various items/devices for AVB:
- Netgear-Switch GS724Tv4 with licence for AVB
- Intel I210
- Motu 1248 AVB
- DSP4you AVB-DGK http://www.dsp4you.com/products/avb-oem-series/avb-dgk
- MOTU LP32 running in mode AVB-expansion http://motu.com/products/avb/lp32
Motu 1248, Motu LP32, both DSP4you AVB dongles are on the switch for
AVB DSP4you. This makes a small running network for AVB.
Motu 1248 and LP32 are only connected by Ethernet-AVB.
https://www.minidsp.com/media/kunena/attachments/14880/DSC_0218.JPG
I am looking now for a pair of good studio monitors running as listeners on AVB
Meyer Sound USA has AVB-capable speakers.
https://meyersound.com/product/cal/https://meyersound.com/wp-content/uploads/2017/01/cal_side_full.jpg
Does someone know real good high-end studio monitors for AVB?
I also want to try to hook an apple device with Thunderbolt 2 (no,
Apple hasn't implemented vers. 3 yet in most of their HW yet) for
AVB. Apple can do AVB. This is the sole reson for me for Apple.
Best regards
Andreas WERNER
>
> > Not exactly simpler, though scriptable. For sure there's many ways to do
> > this, here's one example that uses Jack for MIDI and audio.
> > The only manual step involved is to select stops in aeolus, though that
> > could be automated by selecting a preset via MIDI (haven't tried that
> > part).
> >
> > # start aeolus without the -A option (defaults to Jack)
> > aeolus &
> > # start a jack-MIDI aware smf file player like jpmidi or jack-smf-player
> > from jack-smf-utils
> > jack-smf-player -n foo.mid &
> > # give a little time for ports to show up
> > sleep 5
> > # using ecasound as a jack-aware command-line recorder
> > ecasound -i jack -o foo.ogg &
> > sleep 1
> > # ==== make connections in jack
> > # == midi
> > jack_connect 'jack-smf-player:midi_out' 'aeolus:Midi/in'
> > # == audio
> > jack_connect 'aeolus:out.L' 'ecasound:in_1'
> > jack_connect 'aeolus:out.R' 'ecasound:in_2'
> > # not required but you might want to listen
> > jack_connect 'aeolus:out.L' 'system:playback_1'
> > jack_connect 'aeolus:out.R' 'system:playback_2'
> > sleep 1
> > # make sure to locate transport TL to zero
> > echo locate 0 | jack_transport
> > # start jack_transport
> > echo play | jack_transport
> > # ..recording..
>
>
By using jack_capture instead of ecasound, this should be enough:
jack_capture --jack-transport
jack-smf-player -n foo.mid &
sleep 5
echo locate 0 | jack_transport
echo play | jack_transport
Hi,
Just a little question - I have an RME9652 connected to a Tango24 to give
me 8 d/a outputs via ADAT; but I really need 9 outputs. I only really need
ONE input, for testing/crossover design. The 9 outputs I need are: front
L/R low/mid/high, center, L/R surround.
So I'm wondering the nicest way to do get my 9 ADAT outputs. Maybe I
should just look for a good deal on another Tango24... or, get a Behringer
8000 d/a for less money, and use it for the L/R surround channels. Both
those options give me 8 more inputs that I don't really need right now...
Is there something reasonable I might buy (and sell my Tango24), that has
ADAT, with from 10 to 16 outputs and 8 or less inputs?
Thanks!
John
