libera/#maemo-leste/ Friday, 2024-08-16

Wizzupfreemangordon: if you have any screenshots of the fb plugin in action (rtcom accounts ui plugin / conversations / contacts) which is suited for the news post, can you share it?01:26
freemangordonWizzup: I can make some07:11
freemangordonWizzup: I dumped the pipeline and it looks absolutely ok07:24
freemangordonumm... besides there is no output pipeline07:29
freemangordonok, with audiotestsrc as source I can hear test sound in fremantle07:45
freemangordonand with this as source element, there is also audio output07:47
freemangordonlike, FarstreamChannel::onSrcPadAddedContent gets called07:48
freemangordonWizzup: so, whatever the issue is, it is related to pulsesrc as input source07:53
freemangordonWizzup: looks like some gst threading issue09:09
freemangordonlike, for some reason elements are not properly initialized in vcm, because of the way it connects to FS signals09:09
freemangordonok, defiinitely an issue with pulsesrc09:23
freemangordonusing alsasrc audio works both directions09:24
freemangordonhttps://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/46609:25
freemangordonWizzup: 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 with09:38
Wizzupfreemangordon: good to know alsasrc works :)10:49
Wizzupfreemangordon: what is this test program, is it packaged?11:38
Wizzupoh, just examples/call-handler.c I guess12:09
freemangordon yes13:10
freemangordonjust build the package13:10
Wizzupok13:17
WizzupI'll also rebuild vcm and try alsasink/alsasrc13:17
Wizzupbecause I want to see it work :)13:17
arno11Wizzup: you also should definitely have a look @backend switching13:31
arno11it is really buggy and we never know which backend is in use13:31
freemangordonWizzup: only alsasrc13:31
freemangordonthere is no issue with pulsesink13:32
Wizzupa13:34
Wizzupah13:34
Wizzuparno11: in sphone?13:35
arno11yep13:35
WizzupI just implemented the sphone backend api, I don't know what else it does13:35
Wizzupbut we might just have to make a new gtk/rtcom based ui for it13:35
Wizzupthen I could look at fixing that13:35
arno11ok13:37
arno11btw with twinkle and no proxy, able to make great calls between ippi/leste and linphone/android through wifi or hsdpa (whatever)13:44
arno11and using tls :D13:46
Wizzupwith alsasrc I also have working sip calls with sphone :)13:47
arno11oh great !13:48
freemangordon:)13:48
arno11really great news13:48
freemangordonbut, we shall check why pulsesrc hangs the piupeline13:48
freemangordonthough, I guess I am not the best to debug PA13:48
WizzupI will try in a bit today13:49
Wizzupbtw, telepathy-nonsense (qxmpp based) seems much better already currently than gabble13:50
Wizzupat least for message management and such13:50
WizzupI'll get it packaged proper later, it's better for messaging13:50
Wizzupof course for xmpp calls we'll need gabble for now I guess13:50
freemangordonwhat about contacts?13:50
Wizzupno contacts, that needs to be fixed13:50
Wizzupsame for telepathy-tank13:50
WizzupI don't know yet what the missing piece is13:50
Wizzupmust be in connection manager13:50
freemangordonmhm13:51
Wizzupit does seem to send contact avatars over dbus13:51
Wizzupand I saw abook crash once13:51
freemangordondoes it?13:51
Wizzupyeah, but they are binary I think13:51
Wizzupor at least some encoding issue13:51
freemangordondoes not matter13:51
Wizzupdbus-monitor hangs on it :)13:51
freemangordonheh13:52
freemangordonyou may want to check evolution-data-server-addressbook-backend-telepathy logs13:52
freemangordonlike, starting eds from cmd line with G_MESSAGES_DEBUG=all (and maybe some other env vars, can't remember) should be enough13:53
freemangordonyeah, should be enigh https://github.com/maemo-leste/eds-backend-telepathy/blob/master/src/e-book-backend-tp-log.h#L3013:54
freemangordon*enough13:54
freemangordon/usr/libexec/evolution-addressbook-factory should be the binary13:57
freemangordonIIRC13:57
Wizzupok, but maybe not today14:01
freemangordonI am just providing info, as I am mia till Sunday14:02
Wizzupunderstood14:03
WizzupI will be going to the US for 2 weeks on Sunday so I will also be a bit less available14:03
Wizzupbut available enough to see this all through14:04
Wizzupfreemangordon: thanks for the help, it's really great to have this working14:04
Wizzupfreemangordon: hm, the example app doesn't compile for me14:26
Wizzup/usr/include/gstreamer-1.0/gst/gstcaps.h: In function ‘gst_clear_caps’:14:26
Wizzup/usr/include/gstreamer-1.0/gst/gstcaps.h:208:13: error: Not available before  [-Werror]14:26
WizzupI removed werror for now14:27
freemangordonit compiles fine here in the VM14:34
Wizzupdo probably do not have werror14:34
uvos__arno11: well could you describe what exactly is buggy with backend swiching?15:13
arno11uvos__: 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 backend15:43
arno11so that's a bit confusing15:44
arno11(sofiasip is still available in the backend menu but seems not possible to really select it)15:45
Wizzupuvos__: any luck with the xt1602?15:46
uvos__arno11: ok so i gues the difference here is that the sip backend is registerd after sphone has finished startig up15:52
uvos__this is not well tested15:52
arno11ah ok15:52
uvos__the ui probubly gets out of sync15:52
uvos__Wizzup: no it got stolen15:52
uvos__i had leste booting on it, but the display would not come back after it was turned off15:53
uvos__other than that most everything worked15:53
uvos__modem not tested15:53
Wizzupdo you have the image somewhere?15:53
uvos__Wizzup: maybe have to look15:53
uvos__arno11: ill try to repoduce15:53
Wizzupok, please let me know if you can find it15:54
arno11uvos__: ok thx15:54
Wizzupuvos__: also, do you by chance remember what we had to do for atrix2 dts?15:54
uvos__Wizzup: do for what?15:54
WizzupI think we used another mapphone dts on the atrix15:55
uvos__we looked at the stock dts and signal map15:55
Wizzupiirc we made a list of things that were different, so I'd like to make an atrix2 dts15:55
Wizzupyeah, I'll boot it up and see what dts I am using now15:55
uvos__bionic15:55
uvos__its the same as bionic15:55
Wizzupty15:55
uvos__minus a sensor or something15:55
uvos__yeah it lacks a gyro i think15:55
uvos__(like d4 but unlike bionic15:55
uvos__)15:55
Wizzupok15:56
uvos__arno11: you can maybe play around with sphone -c modprobe commtest15:56
uvos__*arno11: you can maybe play around with sphone -c insmod commtest15:56
uvos__and15:56
uvos__arno11: and sphone -c rmmod commtest15:57
uvos__to see if you can repo without sip15:57
uvos__since i dont have a sip acc15:57
arno11Wizzup: btw could you share alsasrc stuff ?15:57
arno11uvos__: ok i'll try, ty15:57
uvos__as you might expect insmod and rmmod remove and add sphone modules at runtime15:58
arno11yep15:58
Wizzuparno11: I was thinking of building voicecall for -devel with alsasrc and making an issue for pulsesrc15:58
Wizzupthen you can apt update and we have an issue for having to debug this15:58
arno11ah ok makes sense, cool15:58
Wizzupbuilding now16:09
Wizzupbrb16:09
Wizzuphttps://github.com/maemo-leste/bugtracker/issues/74616:09
arno11cool16:10
arno11uvos__: rmmod commtest removed sofiasip but can't use insmod commtest:16:17
arno11 Module rtconf-libprofile has the same provides as16:17
arno11module rtconf-libprofile, and will not be loaded.16:17
arno11bbl16:29
uvos__huh16:37
uvos__thats wierd16:38
uvos__why would it try to insert rtconf-libprofile16:38
uvos__seams really confused16:38
uvos__maybe there is memory corruption somewhere16:38
Wizzupfrom a quick test a sip call on the d4 doesn't work yet, will have to check if it is really the alsasink now16:45
Wizzupmaybe I lack some plugins16:50
Wizzuphm nope16:50
Wizzupok yeah forgot alsasink I think :)16:53
Wizzup(gstreamer1.0-alsa)16:53
arno11Wizzup: it works on n900 both ways but buggy16:55
arno11sound is distorded in one way and too slow on the other16:55
Wizzupthat could be for various reasons16:56
Wizzupit works ok now with my d4 and fremantle16:56
arno11yes indeed16:56
arno11ok16:56
arno11bugs could be codecs, resampling, tsched16:57
arno11oh in fact it works fine16:58
arno11just buggy at the beginning of a call16:59
arno11buffer issue i suppose16:59
Wizzupit might take a bit of time for it to settle16:59
arno11yep, otherwise it seems fine16:59
arno11uhm only for receiving a call17:00
Wizzupyes outgoing doesn't work well, it is stuck on alerting17:00
WizzupI never bothered to fix that yes because nothing was working at the time anyway17:01
Wizzupthat's what you mean, right?17:01
arno11yep same issue for me17:01
Wizzupthis will take some debugging, it could be in many things since we have an abstraction between tp code and tp code17:01
Wizzupso it is possible some things are lost or need to be defined differently17:01
arno11ok17:02
arno11Wizzup: btw i made a PR for n900 mic stuff: /leste/config/pull/5117:10
arno11(if you have time to have a look)17:11
Wizzupa bit later today17:12
Wizzupty17:12
arno11sure, no rush17:14
Wizzuparno11: btw, in the future, please make pull requests against master branch17:46
Wizzupis a volume of 118 good?17:46
Wizzupuvos__: so yeah brightness control also didn't work on atrix218:19
Wizzupand rotation of the device neither18:20
arno11Wizzup: ok @master18:39
arno11yes 118 is ok18:39
arno11only 71% iirc18:39
arno11maybe less18:39
arno11nope it is 98% for device mic (which is fine) and 46% for headset mic (very sensitive)18:51
arno11so 118 and 84 are ok18:53
arno11uvos__: 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.19:10
arno11*it shows always ring on the main window, but things are ok in the submenu.19:12
Wizzupbtw guys, qxmpp implements omemo2, and dino and conversations don't support it21:21
Wizzuponly kaidan does omemo2, and it's not compatible with dino/conversations either (since they lag behind)21:21
Wizzupso 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...21:21
Wizzupit's kinda silly21:22
Wizzupuvos this is one the one that doens't work I think21:47
Wizzup[    6.551879] kxcjk1013 3-000f: Error reading who_am_i21:48
Wizzup[    6.552520] kxcjk1013: probe of 3-000f failed with error -12121:48
sicelolooks like something that could be (easily?) fixable21:54
sicelowhat's the chip?21:56
Wizzupyeah should be, I have the device here but didn't (re)look at the android signal maps21:57
Wizzup(I have to fly on sunday and the end of the month is the nlnet deadline so I am looking at what tobring)21:58
sicelothe 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 important23:37

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