https://www.linuxquestions.org/questions/linux-newbie-8/multi-channel-autop…
> Hello, I have a Raspberry Pi 3B+ with a Terratec Aureon 7.1
> usb Sound Card. I'm looking for a player/sequencer which
> starts at startup and automatically starts playing a 6-channel
> audio track. Can anybody help me with this?
>
> I've tried VLC and set everything on Dolby surround but it
> didn't work.
Just wondering. The post has lots of views but no answers.
--
David W. Jones
gnome(a)hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com
Sent from my Android device with F/LOSS K-9 Mail.
This should have been 1.5.11 but after carefully making sure I had the code and
all the associated bits fully up-to-date I forgot to add in the new user
guide - DOH
Anyway...
1.5.11 Waxwing
Main feature additions/improvements:
The Mixer panel now presents parts with separate left and right VU meters.
There is an overall mono/stereo button in the main window.
If any controls in an effect are different from the listed preset, the colour
of the preset background is changed to a strong blue.
When an AddSynth voice or modulator is using the oscillator from a lower
numbered one, on entering the waveform editor there is a warning in red at the
top of the window.
In the microtonal section the 'A' note frequency range has been fixed as 329Hz
to 660Hz.
All controls that operate only when others are in particular states are shown
as inactive when not available.
There is a small information window that opens instantly, and only remains
visible until the main window is displayed.
On a first time start there is a window overlaying the centre of the main one,
prompting you to check settings via the Yoshimi drop-down menu.
More details in /doc/Yoshimi_1.5.11_features.txt
Yoshimi source code is available from either:
https://sourceforge.net/projects/yoshimi
Or:
https://github.com/Yoshimi/yoshimi
Full build instructions are in 'INSTALL'.
Our list archive is at:
https://www.freelists.org/archive/yoshimi
To post, email to:
yoshimi(a)freelists.org
--
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.
Hi everyone,
LAC is just a few days ahead now! If you can't make it to the conference,
we'll be video streaming paper presentations. Instructions on how this will
work can be found on the conference website:
https://lac.linuxaudio.org/2019/#streaming
Feel free to ask if you have any questions!
Cheers,
The LAC-19 Team
Hi all,
I'm working on an open source (well public domain) computer music system for python called Pippi. It's a library for non-realtime composition: python scores. It's in the 4th beta of the 2.0 -- the 1.0 series was all python2 and based around bytestrings, while the 2.0 is python3 and hopefully a bit more python-ish making use of numpy-backed classes that do some operator overloading and provide a set of affordances for transforming buffers, processing or synthesizing sounds.
I'm also working on a live / "just in"-real-time (meaning asynchronous processing mixed in realtime) system called astrid meant for performance / live-coding / development etc. This is very much alpha in its current form, also a rewrite from previous built-in features from the 1.0 series...
Anyway! Here's a snippet of two instruments running in astrid:
https://www.dropbox.com/s/5fbglgx7ts1656l/asmallthing.mp3?dl=0
One is just a simple granular smoosh of a recording of a particularly harmonic water pump. The other is a 2d pulsar OSC using a stack of wavetables derived from a convolution of some vocal and guitar recordings my friend made, occasionally also processed with waveset-based procedures (ripped from Trevor Wishart's Audible Design book) and some other time domain processes.
This is pippi: https://github.com/luvsound/pippi
I've been working on this for a long time but documentation sucks... I'm more than happy to field any questions anyone who is kind enough to try to dive in might have, and really do plan to work on the docs soon!
Here's a start: http://htmlpreview.github.io/?https://github.com/luvsound/pippi/blob/master…
Erik
Hi all.
As usual the second Tuesday of the month is upon us. I'll be in the
mainhall from 20:00.
See you there.
Cheers
/Daniel
PS Yes, I did copy-paste from the previous mail... DS
Hi,
i recently re-installed Arch Linux on my laptop after a disk swap and suddenly JACK2 (non-dbus) doesn't run with my Behringer UMC1820
anymore. I kept the old system up-to-date and it was working perfectly well. Also, I don't remember making any special adjustments to
get it up and running.
My other cheapo Behringer (UCA202) still works fine, so i assume it's not a general USB audio problem ?
The device is still shown in alsamixer, levels can be set.
A friend tried it on her Mac and it worked there, so I assume the interface itself is still running.
uname -a says:
Linux atarashii 5.0.13-arch1-1-ARCH #1 SMP PREEMPT Sun May 5 18:05:41 UTC 2019 x86_64 GNU/Linux
I usually start jack2 like this (works with the other device):
jackd -P98 -dalsa -dhw:2 -p512 -n3 -o2 -r44100
The error message is:
ALSA: use 3 periods for capture
ALSA: final selected sample format for playback: 24bit little-endian in 3bytes format
ALSA: cannot set channel count to 2 for playback
ALSA: cannot configure playback channel
Released audio card Audio2
audio_reservation_finish
Cannot initialize driver
JackServer::Open failed with -1
Failed to open server
Has anybody experiences something similar ? I don't have any idea right now ...
Best,
Niklas
I sent a long email about testing Ubuntu Studios Controls.
Since it got dumped because it was too long thanks to inserting screen
captures, I then wrote a fixed version and the subject changed creating
broken thread. I'll paraphrase (and I hope this reply is going to the
original thread. Sorry.)
Short version:
- Ubuntu Studio Controls doesn't know about ffado.
- I attempted to force things from the command line, but was unsuccessful.
In fact, I'm not sure I can even get Ubuntu Studo Controls to do nothing.
- Ubuntu Studio Controls also has a problem with retaining the selections
for bridging, etc. If you check them off, then logout/login or reboot they
are back to default (checked). The last time I logged in and started Carla,
there were 12 in/out's for the AF12 and 2 captures for the on board sound
card...
At first during my messing around I could only get the long form of the
port names to show up (i.e. 00148605c4409ac_Unknown_out,
00148605c4409ac_Unknown0_out, etc.) and not human readable names like
capture_1, etc. But, something I did is now causing the human readable
version to appear...don't what caused this and not sure if it will be there
when I boot tomorrow...
I've been doing some experiments with the system I have.
The system: the mobo has onboard 8 channels (front panel L/R, rear panel
L/R, rear panel line out L/R, rear panel sub L/R) and a Echo AF12 on
firewire.
At this point UBS Controls can select between the on board and the AF12 (I
assume when the AF12 is selected and 12 channels appear in Carla or
Patchage this is via ALSA)
Observations:
- alsamixer shows both onboard and AF12
- when AF12 is selected in alsamixer, alsamixer shows nothing
- when on board is selected in alsamixer, alsamixer shows the 8 outputs
- when selected in UBS Controls the associated ports are shown in
Carla/Patchage
- when on board is selected, 8 ports are shown, but only the front panel
L/R produce any signal when connected in Carla/Patchage
(all 8 are unmuted and at 0db in alsamixer)
anybody know how to get the other 6 outputs working?
- when AF12 is selected 12 channels are shown in Carla/Patchage
With on board selected and players/plugins started and connected between
player and front panel L/R all appears good and sounds ok.
With AF12 selected and same player/plugin combo is made sounds good, except
occasionally REALLY nasty high volume (as in speaker/ear damaging loud)
digital burst happens.
In order to test with just ffado I attempted to adapt a script I use on
another pc to force only jack. The adapted part shown below:
#!/bin/bash
set -x
killall -9 jackd jackdbus
#jack_control ds alsa dps capture none dps playback none
jack_control ds firewire dps capture none dps playback none
#jack_control dps device hw:PCH
jack_control dps rate 48000
jack_control dps nperiods 2
jack_control dps period 1024
jack_control start
pactl unload-module module-udev-detect
pactl unload-module module-alsa-card
pactl unload-module module-jackdbus-detect
pactl load-module module-jack-sink client_name=PulseOut channels=2
connect=no
pactl load-module module-jack-source client_name=PulseIn channels=2
connect=no
set +x
exit
This results in the channels of the AF12 being listed like this:
firewire_pcm
00148605c4409ac_Unknown_out
00148605c4409ac_Unknown0_out
00148605c4409ac_Unknown1_out
00148605c4409ac_Unknown2_out
00148605c4409ac_Unknown3_out
00148605c4409ac_Unknown4_out
00148605c4409ac_Unknown5_out
00148605c4409ac_Unknown6_out
00148605c4409ac_Unknown7_out
00148605c4409ac_Unknown8_out
00148605c4409ac_Unknown9_out
00148605c4409ac_Unknown10_out
00148605c4409ac_Unknown11_out
Where Unknown_out is the 1st channel of the AF12, Unknown0_out is the 2nd
channel, etc. (Unknown11_out is the midi port)
I do not know how to get human readable names as can be done in Qjack...
Anybody know how to do this from the command line? Or set the name of the
device to something other than firewire_pcm?
So far I have NOT been able to duplicate the crazy noise spikes I was
getting with alsa running the AF12.
Oh, and while alsamixer will allow the selection of the AF12, alsamixer
then says the AF12 has no controls.
So, to control the gains and such one has to figure out how to stop the
alsa control of the AF12 and then start ffado to make any changes?
Len said:
"It does not use ffado at all but rather the built in ALSA firewire module
stack. This means, do not try to use ffado-mixer, bad things will happen.
Use alsamixer or the qasmixer (which is GUI based) instead.
So far, anyone I have talked to that has had problems with fw devices in
the past few years, since fw was added to ALSA, has had those problems
because they tried to use ffado bits without first removing the alsa bits.
So I am wondering how many people have firewire devices that have found
the alsa drivers for firewire, when used correctly, don't work for them."
So, I admit to some confusion...
Until yesterday (when I added the back port) after boot the sound
system would show a 12 channel device.
At that point I would start Qjack. It's configuration is set to start
ffado with 12 channels.
I then hit start in Qjack and then 12 in/out ports would show in Qjack
> Connect.
If I understand your comment, this would be running the alsa stack AND
the ffado stack against the
firewire device? (In this case an AF12.)
So, now if I start JACK from the UBS Controls, it is using only the ALSA stack?
And the ports that show up in, say Patchage, are ALSA ports?
So, when I'm doing "pro" audio with UBS I don't need ffado now?
And, if so, how does one, for instance, send firmware updates to a fw device?
(not that I expect to do that with an AF12...they stopped making it. :( )
Mac