Good evening!
I went to make some sounds today and discovered that somehow - in my
migration to a new laptop five months ago - that I lost a bunch of
banks/instruments. So I have two questions.
Where to get more Yoshimi/Zyn instruments?
I also copied a bunch of Zyn "banks" into Yoshimi's banks folder, but
they're all subfolders and Yoshimi doesn't seem to find them. How to fix
this?
Along the way, I tried Zyn with its shiny zyn-fusion interface. Makes no
sense to me, although the ability to enlarge the window and make it
visible/usable on my 4K display is nice. Any ideas on how to convince X
to provide real high-rez fonts, or use scalable fonts?
Thanks for any advice.
--
David W. Jones
gnome(a)hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com
"My password is the last 8 digits of π."
Hi,
Another Asterisk PBX related question. I have the Asterisk service
periodically sending System commands on the host machine when particular
phone extensions are called. The Asterisk service is run by the user
asterisk. I made asterisk a member of the 'audio' group so that it can
use 'aplay' with the System command. eg.
System(aplay -c 1 -D pulse /home/iain/sounds/line1.wav)
This plays OK but plays on the laptop's internal hardware rather via
PulseAudio. Jack is running and is configured to use an external USB
sound card and the PulseAudio Jack Sink.
If I run the following as user iain, it plays correctly through the USB:
aplay -c 1 -D pulse /home/iain/sounds/line1.wav
However via the Aterisk service, or by running the command in a console
as asterisk, it plays through the internal speakers. eg.
sudo -H -u asterisk bash -c 'aplay -c 1 -D pulse
/home/iain/sounds/line1.wav'
I don't know if the "-D pulse" option is necessary, I'm trying lots of
things...
As well as the audio group, I've tried adding asterisk to the pulse and
pulse-access groups, but it doesn't help (and yes, I logged out and in
again).
If i run the last code given above with aplay in verbose mode (-vvv), it
throws the error:
xcb_connection_has_error() returned true
But this Is not error because adding "unset DISPLAY ; " to the start of
the command removes the error but it continues to play through the
external speakers. Here is the print for the command:
iain@samsung:~$ sudo -H -u asterisk bash -c 'unset DISPLAY ; aplay -vvvv
-c 1 -D pulse /home/iain/sounds/line1.wav'
Tocando WAVE '/home/iain/sounds/line1.wav' : Signed 16 bit Little
Endian, Taxa 48000 Hz, Mono
ALSA <-> PulseAudio PCM I/O Plugin
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat : STD
channels : 1
rate : 48000
exact rate : 48000 (48000/1)
msbits : 16
buffer_size : 24000
period_size : 6000
period_time : 125000
tstamp_mode : NONE
tstamp_type : GETTIMEOFDAY
period_step : 1
avail_min : 6000
period_event : 0
start_threshold : 24000
stop_threshold : 24000
silence_threshold: 0
silence_size : 0
boundary : 6755399441055744000
iain@samsung:~$
Any suggestions please?
Thanks,
Iain
dear list members,
just a question: is there anybody else on the list using soundcraft
notepad mixers, especially notepad-5 and/or notepad-8fx, and would
provide some basic info in order to help Jim Ramsay, the author of
soundcraft-utils (https://github.com/lack/soundcraft-utils) to develop
the package?
I’m in touch with him and testing and giving some feedback for the
notepad-12fx. but since both of us have this device he can not go on to
adapt his code for the other two devices.
is there anybody owning a notepad 5 or 8fx?
have a good day!
christoph
Hello,
for a new ultra-portable laptop I am looking for a USB sound device to
do linux low latency audio.
The most important factor is a small size, then audio performance, then
price.
This is not a full production system but rather a "travel sketchbook",
but still, that needs low latency and DSP load. What I'm saying is:
microhpone A->D is not *that* important here, if at all necessary.
I am aware that the Behringer UCA 222 or 202 exists, which has
dimensions 18,3 x 14 x 3,6 cm according to Amazon.
Is smaller possible?
Yours,
hgn
Hello,
I have several soft-phone (VoIP) daemons (linphone-daemon) running at
the same time on the same machine and connecting to the same audio card
via PulseAudio Jack sink. They each connect to an Asterisk server on
different ports and do their SIP registrations with Asterisk under
different user names. While the daemons start together with no problems
and with their corresponing SIP phones registered properly, when I start
sending call messages to them, the results can be unpredictable.
Sometimes I can hear concurrent calls together, other times one call
will block the other, and other times all calls will be blocked.
If I have 2 calls running on 2 daemons with one call blocking the other
and I terminate the call that was unblocked and playing through the
speakers, the blocked calls resumes playing but at the point in time
where it should have been playing if it were not blocked. This makes me
think that there might be a problem with PulseAudio Jack sink, but I
don't know.
All concurrent daemons on a call show up in pavucontrol named
"linphone-daemon" and connected to Jack sink in the pull down menu. If
i change the device in pavucontrol of a blocked call to the internal
sound card (from the sink, which is connected to a USB card via Jack),
it does not stop the blockage.
In one attempt to fix, I tried giving each process a different name
with, for example:
bash -c "exec -a linphone-daemon1 linphone-daemon --config
/home/iain/.config/linphone/linphonerc1 --pipe linphone-daemon1.soc" )&
but this didn't change the way each process was named in pavucontrol,
with each process still named linphone-daemon rather than
linphone-daemon1, linphone-daemon2 and linphone-daemon3. The blocks
continue as well.
Might there be some port issue with the sink?
It could of course be a problem relating to my Asterisk configuration,
but I thought i'd ask this question here about the sink in case there
are issues running multiple processes of the same type. I've posted a
related question to a linphone list as well.
Hope someone can help. Thanks, all the best,
Iain
PS. Here's the message I posted to the linphone-users list. It gives
more details. I said in this message that i didn't think the problem is
related to the Jack sink, but i don't know:
Hello,
I need to run several linphone-daemons at once on the one Linux machine
with an Asterisk server. I have set this up so that each daemon loads a
separate config file with a unique UDP/TCP port and SIP username and
corresponding entries in Asterisk's sip.conf. All the daemons use the
PulseAudio Jack sink to enable audio to be played concurrently with that
of other processes.
While each daemon works successfully on its own, the daemons only
sometimes work properly when run together. For example if daemon 1 is on
a call and I send a call message through to daemon 2 to make another
call, the outcome is unpredictable, sometimes both can be heard, other
times the sound of one is blocked and yet other times both audio streams
get jittery.
There are no problems with CPU or memory. Also, i don't believe it is a
problem with the Jack sink as i can run multiple other audio programs
concurrently through the sink successfully, including ones in addition
to the linphone-daemon itself. Also, if i call in from several different
Android phones with SIP software (again Linphone) to extensions in
Asterisk (but not the server-based phones), the sound streams with no
problems.
One difference I see is that the SIP phones all have different IPs,
whereas all the server-based phones are on the same IP. Might this be it?
Can anyone please offer some suggestions?
I'll paste below the script that I use to launch the daemons, a script
to send call messages to the servers and also the entries in Asterisk's
sip.conf.
I've also uploaded a log file of one of the daemons here which includes
log info of an event where the audio stopped (towards the end):
https://ufile.io/088vjf45
Relevant looking messages include:
mediastreamer-message-MSAudio_stream_iterate[0x55d89bd96ff0], local
statistics available:
Local current jitter buffer size: 0.0ms
ortp-warning-ortp_loss_rate_estimator_process 0x55d89bd5c1d0: Suspected
discontinuity in sequence numbering from 503 to 250
ortp-warning-Receiving packet with unknown payload type 0
Hope someone can help,
Iain
#!/bin/bash
# startlinphones.sh
# Start 3 linphone-daemons
process=$(pgrep linphone-daemon)
set -- $process
process=$1
if [ -z $process ]; # to check if at least one daemon is already running
then
echo "Starting linphone-daemon 1. Please wait a few seconds."
$(/home/iain/linphone-desktop/build/OUTPUT/bin/linphone-daemon \
--config /home/iain/.config/linphone/linphonerc1 \
--disable-stats-events \
--log /home/iain/tmp/daemon1.log \
--pipe linphone-daemon1.soc)&
echo "Starting linphone-daemon 1. Please wait a few seconds."
$(/home/iain/linphone-desktop/build/OUTPUT/bin/linphone-daemon \
--config /home/iain/.config/linphone/linphonerc2 \
--disable-stats-events \
--log /home/iain/tmp/daemon2.log \
--pipe linphone-daemon2.soc)&
echo "Starting linphone-daemon 3. Please wait a few seconds."
$(/home/iain/linphone-desktop/build/OUTPUT/bin/linphone-daemon \
--config /home/iain/.config/linphone/linphonerc3 \
--disable-stats-events \
--log /home/iain/tmp/daemon3.log \
--pipe linphone-daemon3.soc)&
else
echo "linphone-daemons already running"
fi
exit 0
--------------------------------------------
#!/bin/bash
# callnumber.sh
# call number on a linphone-daemon
daemonnum=$1
phonenum=$2
if ( [ -z $phonenum ] || [ -z $daemonnum ] );
then
echo "Requires both daemon number and phone number as arguments"
else
process=$(pgrep linphone-daemon)
set -- $process
process=$1 # the 1st number in list
if [ -z $process ];
then
echo "Starting linphone-daemons"
startlinphones.sh
sleep 5
fi
echo "Calling"
$(echo "call sip:$phonenum@127.0.0.1" | \
socat STDIN UNIX-CONNECT:/tmp/linphone-daemon$daemonnum.soc)
fi
exit 0
--------------------------------------------------
excerpt from sip.conf
[daemon1]
type=friend
context=outgoing
host=dynamic
username=daemon1
secret=mypassword
disallow=all
;allow=opus
allow=gsm
allow=ulaw
;direct_media=true
direct_media=false
language=pt_BR
[daemon2]
type=friend
context=outgoing
host=dynamic
username=daemon2
secret=mypassword
disallow=all
;allow=opus
allow=ulaw
;direct_media=true
language=pt_BR
[daemon3]
type=friend
context=outgoing
host=dynamic
username=daemon3
secret=mypassword
disallow=all
;allow=opus
allow=gsm
allow=ulaw
;direct_media=true
direct_media=false
language=pt_BR
----------------------------------
The conf files for each daemon specify unique port numbers
Hi.
Maybe not directly related to Linux Audio, but:
Does the MOTU AVB OSC or HTTP API provide some sort of level metering?
I see it mentioned as a capability, but I see no keys related to
metering info. Does anyone know if this is a thing, and if it can be
done programatically?
If no, I guess I need to use JACK. Being a pure console user, I
wonder if there is something which I could use right away?
Writing a simple (braille display friendly) metering tool shouldn't be
too hard I guess. But if I don't have to...
I'd like to have:
* A configurable update rate/window size so that it doesn't scroll too fast.
* dB as number and perhaps an ASCII bar.
* Clipping indicator
* Configurable mono/stereo port mapping so that each logical
channel does only occupy one line on the screen.
--
CYa,
⡍⠁⠗⠊⠕
Hi all!
I'm running reaper 6.05 on a raspberry pi 4 running raspbian/debian buster. I can't get rid of the last 5ms of latency between the midi note send and the trigger of my midi-to-cv interface (controlling my modular).
I played around with buffer settings in jack and the delay gets smaller as I decrease the buffer. However, no matter how low I set the buffer size, the "offset output to this device" in the preferences will not make the delay less than 5ms.
My setup is like this: reaper -> midi mate ii (usb midi interface) -> midi splitter -> cv.ocd (midi to cv interface) gate out -> soundcraft mtk 22 (mixer with usb audio interface). I also tested with the midi splitter out of the picture, same results.
I realize there will always be some delay, what surprises me, it that I can't adjust the delay so lo that the midi note and the recorded trigger align. With offset at 0 I get 15 ms delay with "reasonable audio buffer", adjusting the offset even to -100ms will still give me the trigger 5ms after the mid note. I'd expect -100ms to over compensate to the point of the trigger being recorded 85ms early...
Any input would be highly appreciated!
https://www.dropbox.com/s/hlvmls7c850i2ia/midi_delay_reaper.png?dl=0
--
Atte
http://latestyoutube.a773.dk | http://a773.dk
Hi list,
I guess it's a known issue that zita-a2j will eat all memory if the
device it was bridging was disconnected. Is there a way to make it
handle disconnects more gracefully without taking down the system after
some time?
m.
