<freemangordon>
Wizzup: I am out of time, however: telepathy-farstream package has example application (in examples directory) that exhibits the same behaviour but is easier to play with
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<Wizzup>
freemangordon: good to know alsasrc works :)
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<Wizzup>
freemangordon: what is this test program, is it packaged?
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<Wizzup>
oh, just examples/call-handler.c I guess
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<freemangordon>
yes
<freemangordon>
just build the package
<Wizzup>
ok
<Wizzup>
I'll also rebuild vcm and try alsasink/alsasrc
<Wizzup>
because I want to see it work :)
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<arno11>
Wizzup: you also should definitely have a look @backend switching
<arno11>
it is really buggy and we never know which backend is in use
<freemangordon>
Wizzup: only alsasrc
<freemangordon>
there is no issue with pulsesink
<Wizzup>
a
<Wizzup>
ah
<Wizzup>
arno11: in sphone?
<arno11>
yep
<Wizzup>
I just implemented the sphone backend api, I don't know what else it does
<Wizzup>
but we might just have to make a new gtk/rtcom based ui for it
<Wizzup>
then I could look at fixing that
<arno11>
ok
<arno11>
btw with twinkle and no proxy, able to make great calls between ippi/leste and linphone/android through wifi or hsdpa (whatever)
<arno11>
and using tls :D
<Wizzup>
with alsasrc I also have working sip calls with sphone :)
<arno11>
oh great !
<freemangordon>
:)
<arno11>
really great news
<freemangordon>
but, we shall check why pulsesrc hangs the piupeline
<freemangordon>
though, I guess I am not the best to debug PA
<Wizzup>
I will try in a bit today
<Wizzup>
btw, telepathy-nonsense (qxmpp based) seems much better already currently than gabble
<Wizzup>
at least for message management and such
<Wizzup>
I'll get it packaged proper later, it's better for messaging
<Wizzup>
of course for xmpp calls we'll need gabble for now I guess
<freemangordon>
what about contacts?
<Wizzup>
no contacts, that needs to be fixed
<Wizzup>
same for telepathy-tank
<Wizzup>
I don't know yet what the missing piece is
<Wizzup>
must be in connection manager
<freemangordon>
mhm
<Wizzup>
it does seem to send contact avatars over dbus
<Wizzup>
and I saw abook crash once
<freemangordon>
does it?
<Wizzup>
yeah, but they are binary I think
<Wizzup>
or at least some encoding issue
<freemangordon>
does not matter
<Wizzup>
dbus-monitor hangs on it :)
<freemangordon>
heh
<freemangordon>
you may want to check evolution-data-server-addressbook-backend-telepathy logs
<freemangordon>
like, starting eds from cmd line with G_MESSAGES_DEBUG=all (and maybe some other env vars, can't remember) should be enough
<freemangordon>
/usr/libexec/evolution-addressbook-factory should be the binary
<freemangordon>
IIRC
<Wizzup>
ok, but maybe not today
<freemangordon>
I am just providing info, as I am mia till Sunday
<Wizzup>
understood
<Wizzup>
I will be going to the US for 2 weeks on Sunday so I will also be a bit less available
<Wizzup>
but available enough to see this all through
<Wizzup>
freemangordon: thanks for the help, it's really great to have this working
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<Wizzup>
freemangordon: hm, the example app doesn't compile for me
<Wizzup>
/usr/include/gstreamer-1.0/gst/gstcaps.h: In function ‘gst_clear_caps’:
<Wizzup>
/usr/include/gstreamer-1.0/gst/gstcaps.h:208:13: error: Not available before [-Werror]
<Wizzup>
I removed werror for now
<freemangordon>
it compiles fine here in the VM
<Wizzup>
do probably do not have werror
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<uvos__>
arno11: well could you describe what exactly is buggy with backend swiching?
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<arno11>
uvos__: if no sip account online, everything is fine. once a sip account is connected, sofiasip backend appears in caller ui and i'm able to select it (but just one time). then it seems you can't select it anymore, it always shows ring as backend
<arno11>
so that's a bit confusing
<arno11>
(sofiasip is still available in the backend menu but seems not possible to really select it)
<Wizzup>
uvos__: any luck with the xt1602?
<uvos__>
arno11: ok so i gues the difference here is that the sip backend is registerd after sphone has finished startig up
<uvos__>
this is not well tested
<arno11>
ah ok
<uvos__>
the ui probubly gets out of sync
<uvos__>
Wizzup: no it got stolen
<uvos__>
i had leste booting on it, but the display would not come back after it was turned off
<uvos__>
other than that most everything worked
<uvos__>
modem not tested
<Wizzup>
do you have the image somewhere?
<uvos__>
Wizzup: maybe have to look
<uvos__>
arno11: ill try to repoduce
<Wizzup>
ok, please let me know if you can find it
<arno11>
uvos__: ok thx
<Wizzup>
uvos__: also, do you by chance remember what we had to do for atrix2 dts?
<uvos__>
Wizzup: do for what?
<Wizzup>
I think we used another mapphone dts on the atrix
<uvos__>
we looked at the stock dts and signal map
<Wizzup>
iirc we made a list of things that were different, so I'd like to make an atrix2 dts
<Wizzup>
yeah, I'll boot it up and see what dts I am using now
<uvos__>
bionic
<uvos__>
its the same as bionic
<Wizzup>
ty
<uvos__>
minus a sensor or something
<uvos__>
yeah it lacks a gyro i think
<uvos__>
(like d4 but unlike bionic
<uvos__>
)
<Wizzup>
ok
<uvos__>
arno11: you can maybe play around with sphone -c modprobe commtest
<uvos__>
*arno11: you can maybe play around with sphone -c insmod commtest
<uvos__>
and
<uvos__>
arno11: and sphone -c rmmod commtest
<uvos__>
to see if you can repo without sip
<uvos__>
since i dont have a sip acc
<arno11>
Wizzup: btw could you share alsasrc stuff ?
<arno11>
uvos__: ok i'll try, ty
<uvos__>
as you might expect insmod and rmmod remove and add sphone modules at runtime
<arno11>
yep
<Wizzup>
arno11: I was thinking of building voicecall for -devel with alsasrc and making an issue for pulsesrc
<Wizzup>
then you can apt update and we have an issue for having to debug this
<arno11>
uvos__: rmmod commtest removed sofiasip but can't use insmod commtest:
<arno11>
Module rtconf-libprofile has the same provides as
<arno11>
module rtconf-libprofile, and will not be loaded.
<arno11>
bbl
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<uvos__>
huh
<uvos__>
thats wierd
<uvos__>
why would it try to insert rtconf-libprofile
<uvos__>
seams really confused
<uvos__>
maybe there is memory corruption somewhere
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<Wizzup>
from a quick test a sip call on the d4 doesn't work yet, will have to check if it is really the alsasink now
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<Wizzup>
maybe I lack some plugins
<Wizzup>
hm nope
<Wizzup>
ok yeah forgot alsasink I think :)
<Wizzup>
(gstreamer1.0-alsa)
<arno11>
Wizzup: it works on n900 both ways but buggy
<arno11>
sound is distorded in one way and too slow on the other
<Wizzup>
that could be for various reasons
<Wizzup>
it works ok now with my d4 and fremantle
<arno11>
yes indeed
<arno11>
ok
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<arno11>
bugs could be codecs, resampling, tsched
<arno11>
oh in fact it works fine
<arno11>
just buggy at the beginning of a call
<arno11>
buffer issue i suppose
<Wizzup>
it might take a bit of time for it to settle
<arno11>
yep, otherwise it seems fine
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<arno11>
uhm only for receiving a call
<Wizzup>
yes outgoing doesn't work well, it is stuck on alerting
<Wizzup>
I never bothered to fix that yes because nothing was working at the time anyway
<Wizzup>
that's what you mean, right?
<arno11>
yep same issue for me
<Wizzup>
this will take some debugging, it could be in many things since we have an abstraction between tp code and tp code
<Wizzup>
so it is possible some things are lost or need to be defined differently
<arno11>
ok
<arno11>
Wizzup: btw i made a PR for n900 mic stuff: /leste/config/pull/51
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<arno11>
(if you have time to have a look)
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<Wizzup>
a bit later today
<Wizzup>
ty
<arno11>
sure, no rush
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<Wizzup>
arno11: btw, in the future, please make pull requests against master branch
<Wizzup>
is a volume of 118 good?
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<Wizzup>
uvos__: so yeah brightness control also didn't work on atrix2
<Wizzup>
and rotation of the device neither
<arno11>
Wizzup: ok @master
<arno11>
yes 118 is ok
<arno11>
only 71% iirc
<arno11>
maybe less
<arno11>
nope it is 98% for device mic (which is fine) and 46% for headset mic (very sensitive)
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<arno11>
so 118 and 84 are ok
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<arno11>
uvos__: backend switching works fine, the problem seems only with the gui, showing always ring even if sofiasip is selected. and it happens only if you switch back to ring and then to sofiasip again.
<arno11>
*it shows always ring on the main window, but things are ok in the submenu.
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<Wizzup>
btw guys, qxmpp implements omemo2, and dino and conversations don't support it
<Wizzup>
only kaidan does omemo2, and it's not compatible with dino/conversations either (since they lag behind)
<Wizzup>
so we will implement omemo2 first, and then see if we can have qxmpp be compatible with both, or at least implement omemo1 for it or something...
<Wizzup>
it's kinda silly
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<Wizzup>
uvos this is one the one that doens't work I think
<Wizzup>
[ 6.552520] kxcjk1013: probe of 3-000f failed with error -121
<sicelo>
looks like something that could be (easily?) fixable
<sicelo>
what's the chip?
<Wizzup>
yeah should be, I have the device here but didn't (re)look at the android signal maps
<Wizzup>
(I have to fly on sunday and the end of the month is the nlnet deadline so I am looking at what tobring)
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<sicelo>
the matrix plugin for conversations - does it support session verification? it's one hell of a thing to do on many clients, and if one does use E2E, the verification is quite important
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