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.
Hello,
using zita-njbridge between two x86_64 systems works great. However,
when I try to connect an arm-based system (Banana Pi M1) with a 64 bit
system things get less stable (dropouts even with large buffers, error
messages, segmentation fault on the 64bit side). I tried the Ubuntu
18.04 provided version (0.1.1) as well as a freshly compiled version
(0.4.4). Did anyone successfully try zita-njbridge on arm-based systems?
Giso
(when the segfault happens it is always in netrx.cc, line 264 in the
function write_zeros)
Hi list,
I am running a telephony daemon on a laptop that gives no audio
interface configuration options and I wish to use it with Jack. I'm
confused as to whether or not it is using ALSA. If Jack is already
running (with pulseaudio-module-jack uninstalled) and configured to use
an external USB sound card, starting the daemon results in its audio i/o
being connected with the laptop's internal mic and speaker. If Jack is
not running, the daemon connects with card 1 listed by "aplay -l", which
happens to be the USB card. However, if Jack is running with
pulseaudio-module-jack installed and all other non-jack audio apps are
routed via the sink to the USB card, the telephony daemon again connects
and runs correctly with laptop's internal hardware.
I don't think ALSA-Jack loopback scripts will work as I think
pulseaudio-module-jack does this job now. Is there a solution with
kernel modules? Can someone please put me in the right direction?
Thanks!
Iain
Thanks to everyone for the tips and suggestions. I thought I'd write a
quick update to say it's all sorted now. tl;dr: Ryzen 3 with Vega works
just fine on 5.4.x series kernels.
I started shopping around for the cards Paul mentioned but couldn't find
versions with that chipset but without fans for the Radeon and the nVidia
route also turned up blanks.
Luckily, one of my machines is a work from home machine and I managed to
bag an nVidia Quadro K1200 from the stores at work. Four DP outputs and
it seems much more performant. That's the one with more monitors, but not
the low latency kernel.
The studio machine has a 4K display and couldn't use the nVidia drivers
because of the RT kernel patches. Turns out though that the solution was
staring me in the face: this is a Ryzen 3 motherboard and apparently has
built in graphics. It didn't work when I first built the machine but
upgrading to the latest kernel (5.4.x) all now seems good and the
performance is the best I've had from ardour, and that's running full
screen at 4K. I can scroll!
Anyway, seems that the Vega 8 (Ryzen built-in) graphics option is more
than adequate so if anyone reads this thread looking for answers, that's
another option.
bjb
On Sun, Mar 29, 2020 at 10:43:44AM +0100, Ben Bell wrote:
> Apologies for this being only tangentially on-topic, but has anyone got a
> recommendation for a decent video card for driving a 4K monitor under
> Linux?
>
> The audio connection here is that I'm looking for something that is passively
> cooled (or at very quiet if not), performs well, and is low-latency friendly
> (so probably not proprietary drivers)? With those requirements I don't
> really fancy my chances in the "google, buy and hope" approach.
>
> I've been using cheap nvidias with nouveau for a while but performance of
> certain operations (scrolling, dragging/resizing windows) is painfully slow,
> particularly since I made the jump to a 4K monitor.
>
> Has anyone got some success stories?
>
> bjb
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user(a)lists.linuxaudio.org
> https://lists.linuxaudio.org/listinfo/linux-audio-user
On 3/28/20 11:43 PM, Ben Bell wrote:
> Apologies for this being only tangentially on-topic, but has anyone got a
> recommendation for a decent video card for driving a 4K monitor under
> Linux?
>
> The audio connection here is that I'm looking for something that is passively
> cooled (or at very quiet if not), performs well, and is low-latency friendly
> (so probably not proprietary drivers)? With those requirements I don't
> really fancy my chances in the "google, buy and hope" approach.
>
> I've been using cheap nvidias with nouveau for a while but performance of
> certain operations (scrolling, dragging/resizing windows) is painfully slow,
> particularly since I made the jump to a 4K monitor.
>
> Has anyone got some success stories?
>
> bjb
I don't know how helpful this will be to you, but I have a Dell laptop
(XPS 15 7590) with 4K HDR OLED display. For video hardware, it includes
both Intel HD Graphics 630 hardware and an NVidia GTX 1650 with 4GB of
GDDR5.
I've never used the NVidia side at all, just the Intel HD that's
supported out of the box with Debian. I'm also running an RT kernel and
audio setup.
I've never experienced the performance issues you mention above.
Those issue do sound familiar, though. My wife had a netbook with 2GB of
RAM and some onboard Intel hardware with shared system memory for
handling the display. It handled the built-in 1280x800 display with no
problems. But when connected to an external 1920x1080 monitor, those
kinds of issues would happen. I think it was simply because it forced
the video hardware to use more shared memory and that affected the rest
of the system.
My laptop's Intel hardware has driven 4K HDR videos full screen in the
background while I'm going other things with it. So you might see if
getting something like that would be sufficient.
That might mean a motherboard replacement. I don't know if the Intel HD
630 is available as a standalone video card.
As far as cooling noises go... I've never heard a fan sound from my
laptop. It seems to have some kind of passive cooling system going. But
Hawaii is a warmer place than most of the United States. Right now the
CPU's at 129F, running at 900MHz. When the CPU really cranks up - all 8
cores/16 threads running all out - it hits over 200F at 4.2GHz. I've
never seen it hit its nominal 5GHz. I have it sitting on a laptop cooler
that seems to keep it under control.
So, anyway, I think an Intel HD 630 would do what you want without any
problems.
Hope that helps!
--
David W. Jones
gnome(a)hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com
"My password is the last 8 digits of π."
