libera/#maemo-leste/ Thursday, 2024-08-15

antranigvGitHub is down?01:04
antranigvit's back01:40
antranigvinstalling devel01:48
antranigvwell, technically upgrading to it01:48
antranigvany chance to install regular window manager? like Window Maker?01:53
antranigvI'll just read some docs01:53
gnarfaceantranigv: my guess is it would work the same way as devuan04:14
gnarface(which, afaik is more or less the same way as debian; use any package manager to install it then use update-alternatives to switch to it if your login manager doesn't give you a menu)04:15
siceloyou probably are better off just bootstraping debian/devuan directly08:54
antranigvnice!09:29
antranigvokay, devel is installed, updated, and now rebooting09:35
antranigvtoo bad the power button doesn't work09:35
antranigvand many of the old maemo apps are not available as well, I guess I should package them09:35
sicelopower button doesn't work because?09:36
antranigv🤷‍♂️09:40
antranigvI press it, it doesn't do anything09:40
antranigvI think I bricked the device. oops. I'm getting "Nokia RX-51" :D09:40
antranigvok just boot stuck, I rebooted. huh! that was worrying for a second!09:42
antranigvto be clear, I mean when I'm in leste, that's when the power button doesn't respond09:44
antranigvwow it's much faster!09:46
antranigvdoes anyone know why I can't press 'enter' in profanity? did I forget something?11:18
sicelotry Ctrl+M11:21
freemangordonantranigv: power button should respond11:25
antranigvsicelo thanks <311:29
antranigvtotally forgot about keymaps11:29
inkyantranigv: i think i change something in xsession in /etc/X11 to switch to wmaker.14:28
inkycomment a line, add wmaker line. don't remember.14:28
Wizzupfreemangordon: so we keep 6.6 in devel or shall we push it to stable?17:33
freemangordonнот суре, гижен тхе поверофф/ребоот иссуе17:44
freemangordonoops17:44
freemangordonnot sure, given the poweroff/reboot issue17:44
Wizzupok17:45
Wizzupmany call things depend on it17:45
Wizzuplike, we can't even use tp-ring for calls without it17:45
freemangordonif we can live with that for stable, then yes17:45
freemangordonok, go for stable then17:45
freemangordonbut someone shall look at that issue17:46
Wizzupso far it's been a heisenbug, every time we look we can't reproduce17:46
Wizzupmaybe after the end of the month17:46
freemangordonI may look at it earlier is I find some spare time17:46
freemangordon*if I17:47
Wizzupgiven the nlnet deadline is end of month, maybe we can try to do as muhc there as possible :D17:47
freemangordonright17:48
freemangordonany SIP client for android to recommend?18:17
freemangordonWizzup: ^^^?18:20
Wizzupfreemangordon: don't know sorry18:42
Wizzupwe use twinkle with success on leste18:42
Wizzupfreemangordon: also fremantle has working sip18:42
freemangordonoh18:42
freemangordonok18:42
WizzupI can give you my fremantle setting in a bit18:43
Wizzup(if you need them)18:49
freemangordonsure, I have some experience with sip, but not on fremantle18:57
Wizzupwill send over dm19:16
freemangordonWizzup: why do you think tp-rakia uses gst?19:47
Wizzupfor farstream20:05
Wizzupdoes it not?20:06
Wizzupmy gst comments might have been about gabble20:06
freemangordonfarstream is skype, no?20:06
Wizzupno20:06
freemangordonanyway, I think tp managers does not take care about media20:06
Wizzupit looks like we don't have a fork for telepathy-rakia yeah heh20:06
Wizzupfarstream is part of telepathy FWIW20:07
freemangordonit is the client that shall take care20:07
Wizzupfreemangordon: what do you mean, does not take care about media20:07
freemangordoncodecs are not in TP20:07
Wizzupclearly arno already hears one side now with tp-rakia, so I would be surprised20:07
freemangordonhow's that?20:07
Wizzupwell, he makes a call with sphone with sip and gets one way audio20:07
Wizzuplet me clone tp rakia src, I don't think we have it20:08
freemangordonwho cares about org.freedesktop.Telepathy.Call1.Content.MediaDescription.Codecs20:08
Wizzupvoicecall-manager presumably20:08
freemangordonI cloned it, there is nothing about any audio processing there as far as I can find20:08
freemangordonaaah20:08
freemangordonright20:08
Wizzupplugins/providers/telepathy/src/farstreamchannel.cpp20:08
freemangordonthat's the 'client' I am talking about20:08
Wizzupon voicecall repo20:09
Wizzupyeah, ok :)20:09
freemangordonok, lemme see what happens there20:09
Wizzupty :)20:09
freemangordonwas there any special trick to enable vcm debug logs?20:09
freemangordonok "Got audio sink, initializing audio input"20:11
kivastarting calendar from calendar-widget works on pinephone now, thanks20:16
Wizzupfreemangordon: let me check20:16
Wizzupfreemangordon: I used this20:16
WizzupQT_FORCE_STDERR_LOGGING=1 QT_LOGGING_RULES='org.nemomobile.voicecall.debug=true' G_MESSAGES_DEBUG=all GST_DEBUG=3 /usr/bin/voicecall-manager20:16
kivabut calendar week numbers does not align well to lines in PP 1440x720 screen.20:17
kivathere is 6 week numbers and only 5 week lines.20:19
freemangordonWizzup: seems we are missing codecs20:20
freemangordonlemme check where does it look for them20:20
freemangordonWizzup: any clue where shall I look for fsrtpconference_disco20:21
freemangordonhmm, farsight20:21
kivaactually there is still calendar widget bug (starting calendar pushing widget)..it only worked two times..strange.20:22
freemangordonyes, known issue20:23
Wizzupfreemangordon: yes, you might need to install some gst codecs20:23
WizzupI don't know exactly which of the top of my head, but I believe I installed a bunch20:23
Wizzupthe codecs should all be gst plugins20:24
Wizzupfor example gstreamer1.0-nice for ICE20:24
freemangordonhttps://pastebin.com/PnpSF5Dg20:24
freemangordonok20:24
Wizzuplet me see20:25
Wizzupfreemangordon: ok, let's see if we can get those all installed20:25
freemangordonyes, trying to20:26
Wizzuphard to find..20:29
freemangordonseems most are in gst-libav20:29
freemangordonbut lemme check how farstream tries to init them20:29
Wizzupok, I do have that one20:29
Wizzupmaybe some are also libgstrtp.so ?20:30
Wizzupwork mtg, back in a bit20:31
freemangordonmaybe20:31
freemangordonno, it is some other issue20:38
antranigvguuuuuys20:43
antranigvI broke something20:43
antranigv:D20:43
antranigvit can't mount / properly, I guess? but I'm probably in initramfs? I assume20:43
antranigvokay so the root filesystem is in RO mode20:53
freemangordonWizzup: structure gststructure.c:2841:gst_structure_get_valist: Expected field 'channel-mask' in structure: audio/x-raw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ];20:54
freemangordonGST_PADS gstpad.c:2527:gst_pad_link_full: link between audioconvert98:src and rtpg729pay1:sink failed: no common format20:54
freemangordonhmm, I wonder if PA allows 8000 Hz21:10
arno11ah, so it tries to use g729, right ?21:13
freemangordonit tries to use 6 codecs21:13
freemangordonall fail21:13
arno11ok21:14
arno11even PCMA ?21:14
freemangordonsee https://pastebin.com/PnpSF5Dg21:14
freemangordonit does not try PCMA21:14
freemangordonthis is caling maemo fremantle->maemo leste21:15
freemangordonso I guess those are fremantle supported codecs21:15
freemangordonbut codecs should not be an issue21:15
freemangordonI see we have codecs21:15
freemangordonbut for some reason gst negotiation fail;s21:15
arno11ok btw 8000Hz should work21:16
freemangordonon d4?21:16
arno11ah, on d4 i don't know21:17
arno11on n900 it works because we have a custom daemon.conf21:17
freemangordonwell, it goes through audioresample21:17
freemangordonso it should be ok21:17
freemangordonbut I wonder if it is :)21:17
freemangordonlemme try something21:17
antranigvokay eMMC boots fine, but the look of it21:17
antranigvbut when I boot from SD card I can't mount /21:18
arno11antranigv: you mean you want to mount emmc from leste, right ?21:19
antranigvarno11 no no no21:19
antranigvwhen I choose "boot from SD card" in uboot, I get the console and then *everything* fails21:20
antranigvand finally I get "login"21:20
arno11weird21:20
antranigvbad SD card? not sure, I can login fine and see everything21:21
antranigvI have a video, should I share21:22
antranigv?21:22
Wizzupfreemangordon: ok, back, how can I help?21:27
Wizzupfreemangordon: to me 'no common format' means there is no codec that can be agreed upon with the remote party21:27
WizzupI think you can record any freq you want from pa, it will just resample21:28
sicelodo we have g.729 in gst+devuan-chimaery? it was proprietary before, and only freed in last couple of years21:30
freemangordongst-launch-1.0 audiotestsrc  ! capsfilter caps=audio/x-raw,rate=8000,channels=1 ! audioconvert ! rtpg729pay ! fakesink21:30
freemangordonthis fails21:30
arno11i tried different codecs and atm only PCMA, PCMU work between leste and android. g729 doesn't work when requested by android21:31
Wizzupwith this?21:31
WizzupWARNING: erroneous pipeline: could not link audioconvert0 to rtpg729pay021:31
Wizzuparno11: right but perhaps all pipelines fail for the same reason, so if we can debug one...21:31
arno11ok indeed21:31
siceloantranigv: you can share ...21:31
freemangordonWizzup: yes, that error21:32
freemangordonthe same error sip call fails with21:32
* Wizzup checks gst-inspect-1.0 audioconvert21:33
freemangordonhmm, I see avdec_g72921:33
freemangordonbut no encoder21:33
Wizzupwhat is the audioconvert supposed to do21:35
Wizzupdoes it figure out how to convert to rtpg729pay by querying it?21:36
freemangordonnot sure21:36
freemangordonlemme check21:36
freemangordonno21:36
Wizzup  SINK template: 'sink'21:36
Wizzup    Availability: Always21:36
Wizzup    Capabilities:21:36
Wizzup      audio/G72921:36
Wizzup               channels: 121:36
freemangordonhttps://gstreamer.freedesktop.org/documentation/audioconvert/index.html?gi-language=c21:36
Wizzup                   rate: 800021:36
Wizzupthis is the sink template for rtpg729pay21:36
freemangordonyes, I see21:36
freemangordonand I wonder how is that supposed to work21:37
freemangordonbug in farstream?21:37
Wizzupmaybe21:38
freemangordonaudioconvert cannot convert to sink requirements21:38
Wizzuplet's see if we can make it work standalone21:38
freemangordonmhm21:38
freemangordonencodebin?21:39
Wizzupmaybe my sink desc is wrong21:39
Wizzupbrb.21:39
freemangordonno, it is the same here21:40
Wizzupyes but it is src21:41
Wizzupnot sink21:41
Wizzupsee21:42
WizzupPad Templates:21:42
Wizzup  SRC template: 'src'21:42
Wizzup    Availability: Always21:42
Wizzup...21:42
freemangordonST_DEBUG=4 gst-launch-1.0 audiotestsrc  ! capsfilter caps=audio/x-raw,rate=8000,channels=1 ! audioconvert ! alawenc ! rtppcmapay ! fakesink21:42
freemangordonthis works21:42
Wizzupone sec21:43
Wizzupb21:50
Wizzupok, great21:50
freemangordonnot really, as encodebin should do it for us21:50
Wizzupwhat is alawenc?21:51
Wizzupshould do what for us?21:51
freemangordon"Convert 16bit PCM to 8bit A law"21:51
freemangordonor PCMA21:51
freemangordonhmm, lemme see if fremantle offers PCMA21:52
Wizzupok so one change was to make the right caps, yeah?21:53
Wizzupor rather, why did you swap out rtpg729pay ?21:53
Wizzupand does encodebin construct this?21:53
freemangordonI did swap rtpg729pay because I did not find the codec for G72921:55
Wizzupok21:55
freemangordonbut I found the codec for rtppcmapay (PCMA)21:56
freemangordonand no, encoderbin does not put alawenc for us21:56
freemangordonlemme see how exactly farstream decides which bins to create21:56
freemangordonoh21:58
freemangordon"GST_PADS gstpad.c:2585:gst_pad_link_full: linked audioconvert78:src and alawenc3:sink, successful"21:58
freemangordonso, seems arno11 is right and PCMA works22:00
Wizzupand this already works in the vm before the research we did, or?22:00
freemangordonI test on d422:00
freemangordonno idea22:00
Wizzupok, d4 then22:00
Wizzupfrom your paste it looks like we could not find any codec22:00
freemangordonlemme try again22:01
Wizzupmaybe the farstreamchannel.cpp isn't updated for more recent gst22:02
freemangordonno, this happens il libfarstream22:03
freemangordon*in22:03
Wizzupok22:03
freemangordonand actually it seems it negotiates PCMA/PCMU/speex and one more22:03
freemangordonno, lemme see if gst pa plugin works22:03
freemangordonit does22:04
Wizzupstandalone you mean?22:06
Wizzupwhere is libfarstream source22:06
freemangordonapt-get source22:07
freemangordonapt-get source libfarstream-0.2-522:07
freemangordonbut anyway, it seems some PA interaction issue22:08
freemangordonas there are no pa sink/src22:08
Wizzuphm..22:09
freemangordonhmm...22:11
freemangordonthere is "playback manager" stream22:11
Wizzupwhere do you see it and what is it a part of22:12
Wizzupbut 'pulsesrc' works for you and grabs the mic yeah?22:12
freemangordonpavucontrol22:13
freemangordonyes, it works22:13
freemangordonbut lemme check the flags of it22:13
freemangordonmaybe it does not allow non-native freqs22:13
Wizzupso this playback manager is farstream?22:13
WizzupI'll just let you do this, let me know how I can help22:14
freemangordonok22:14
freemangordonyes playback-manager is farstream22:15
freemangordonoh, sorry22:15
freemangordon"voice manager"22:15
arno11Wizzup: btw, shall i do a PR for mic stuff in hifi profile, or ?22:17
arno11(it makes twinkle sip calls working ootb)22:19
freemangordonhmm, pulse stuff is added by vcm it seems22:22
Wizzupfreemangordon: the pulsesrc and pulsesink?22:27
freemangordonyes22:27
Wizzupwell if you get the voice manager I think that is a step in the right direction, I don't think I ever got this22:28
Wizzupbut I am not sure what the current error is that you see22:28
freemangordonit happens only fisrt time after vcm is started22:28
freemangordonthere is no error22:28
freemangordonI wonder what are the volume22:28
freemangordon*volumes22:28
freemangordonlemme play a bit with alsa22:28
WizzupI think it adds a volume element at least22:28
Wizzupin FarstreamChannel::initAudioInput22:29
freemangordonmhm22:31
freemangordonand also   AUDIO_SINK_ELEMENT/AUDIO_SOURCE_ELEMENT22:31
Wizzupyeah22:32
freemangordonI wonder how vcm connects its pipeline with farstream pipeline22:35
freemangordonFarstreamChannel::addAndLink: binobj= audio-output-bin  src= (NULL)  dst= queue122:37
freemangordonso, how that gets connected to farstream pipeline?!?22:37
antranigvah, stupid video file22:38
antranigvok uploading22:38
freemangordonWizzup: I guess in onFsConferenceAdded(), but I don't see that called22:40
Wizzup    if (!gst_bin_add(binobj, ret)) {22:41
Wizzup        setError(QLatin1String("Could not add to bin "));22:41
Wizzup        gst_object_unref(ret);22:41
Wizzup        return 0;22:41
Wizzupdoesn't this do it?22:41
Wizzupand then the linking22:41
Wizzupin FarstreamChannel::addAndLink22:41
freemangordonno, no, see what onFsConferenceAdded does22:41
Wizzuparno11: when you said tls doesn't work, did you write sips as url scheme22:42
Wizzuparno11: nevermind for now I think22:42
Wizzupfreemangordon: I don't know what this does, I assumed itwas for calls were there are more than two people22:43
antranigvdone! https://antranigv.am/misc/n900_maemo-leste_mount-issue.mp422:43
Wizzupit does the same gst bin add22:43
arno11Wizzup: yes, as url scheme22:44
freemangordonWizzup: ok, it is called22:44
freemangordononFsConferenceAdded that is22:44
Wizzupok22:44
freemangordonthis is where it (presumably)links src/sink with fs pipeline(s)22:45
WizzupI don't see it actually22:46
Wizzupthe linking hppens in addAndLink as far as I can see22:47
Wizzupbut it sets it to a playing state there I guess22:47
freemangordonsee   gboolean res = gst_bin_add(GST_BIN(self->mGstPipeline), GST_ELEMENT(conf));22:48
freemangordonconf is FS pipeline22:48
freemangordonIIUC22:48
Wizzupbut isn't there the same in addAndLink22:49
Wizzup    if (!gst_bin_add(binobj, ret)) {22:49
freemangordonit is, but addAndLink is used for elements we create22:52
freemangordonlike pulsesink/pulsesrc22:53
Wizzupright22:53
freemangordonso, whatever happens:22:55
freemangordonGST_STATES gstelement.c:2769:gst_element_continue_state:<send_tee_1> completed state change to PLAYING22:55
freemangordonwhat about media role?22:55
freemangordonsee   setPhoneMediaRole()22:56
antranigvany thoughts?22:59
Wizzupantranigv: we're kind of in the middle of making sip calls work, will check in a bit23:01
Wizzupfreemangordon: I saw the media role thing, I don't think it should matter23:01
antranigvWizzup nice! can I join? :D23:01
Wizzupwhat do you mean, join?23:02
freemangordonWizzup: also:23:02
freemangordonI: [pulseaudio] sink.c: Cannot update sample spec, monitor source is RUNNING23:02
freemangordonmodule-stream-restore.id = "sink-input-by-media-role:phone"23:02
freemangordonI am almost sure volume for that role is 023:03
Wizzupantranigv: the video you posted, did you change /etc/fstab?23:03
antranigvWizzup I did, I think I disabled swapon, or enabled it, or something, not sure23:03
Wizzupfreemangordon: ok, that is a problem23:03
Wizzupantranigv: this is probably your problem23:03
antranigvwhat should it be like?23:03
Wizzupwhat it was before... :)23:03
Wizzupnot sure what you changed23:04
antranigvI'll try now!23:04
freemangordonWizzup: it would not have been like that, if it was not removed from volume-control appet :p23:05
freemangordon*applet23:05
Wizzupmhm23:05
WizzupI didn't think the media role assigned a volume23:06
WizzupI thought it was just used to cork streams23:06
freemangordonsee https://github.com/maemo-leste/maemo-statusmenu-volume/blob/d6c3ffa8de080e82fe66881163594da2ca73ce19/src/item.c#L64823:07
Wizzupok23:08
Wizzupif this is the problem I'll be sad23:08
freemangordonmaybe just comment that code and test23:08
freemangordonin setPhoneMediaRole() that is23:09
freemangordonWizzup: how would not media role assign a volume? like, playing music and making phone call must have different volumes23:10
freemangordonswitched automatically as soon as you answer the phone23:10
freemangordonor hangup23:10
freemangordonbut anyway, I am not sure that's the proble23:10
freemangordonm23:10
antranigvokay, but / is mounted as ro, I am trying to remount as rw, but it's not working. pretty sure `mount -o rw /` is all I need to do23:13
antranigvwait, do I need `-u` to "update"?23:14
antranigvnah, that's a Solaris thing, not a GNU thing, I think23:14
Wizzupantranigv: no, your fstab is probably wrong23:15
Wizzuptry to mount your sd card on your computer23:15
antranigvWizzup I see what you mean. I'll fix via the computer.23:15
antranigvnow I just need something that can mount ext4 :D23:15
antranigvah right! maybe I can try mounting it from regular maemo!23:17
freemangordonWizzup: volume: mono: 65536 / 100% / 0.00 dB23:18
freemangordonlooks kile volume is ok23:18
freemangordon*like23:18
Wizzupantranigv: you won't be able to from regular maemo I think23:19
Wizzupfreemangordon: ok, so where are you at now23:19
Wizzupdid you make any changes, do you hear something?23:19
freemangordonno changes23:19
freemangordonstill nothing23:20
freemangordonand, there is no PA source, only sink23:20
freemangordonalso, I run out of time23:20
freemangordon:)23:20
freemangordonhowever, there is no obvious error23:20
freemangordonmaybe I shall try in VM, to ignore d4 audio issues23:21
Wizzupno pa source would be a big problem23:22
Wizzupfreemangordon: so you also didn't install extra gst plugins?23:24
freemangordonyes23:24
Wizzupyou can try vm but make sure it's not some double/trip NAT thing23:24
freemangordonbtw, how to disable sphone rotation?23:24
Wizzupsee leste-config-n90023:24
Wizzupleste-config-n900/usr/share/sphone/sphone.ini.d/landscape-call.ini.leste23:25
Wizzup[Gui]23:25
WizzupLandscapeCall=123:25
freemangordonthanks23:25
freemangordonwell, does not work23:28
freemangordonstill rotates23:28
siceloLandscapeCalls  ...23:31
freemangordonheh23:32
Wizzupoh sorry23:33
WizzupI just pasted from the commit23:33
freemangordonyeah, that works23:33
freemangordonanyway, enough for today23:33
freemangordonwill just try to disable media role in vm23:34
Wizzupok23:34
freemangordonhard to say, streams are not created23:40
freemangordonthis needs further debugging23:40
freemangordonbut not now, perhaps on Sunday/Monday23:40
Wizzupok :)23:42
WizzupI wonder if it is related to stun or something, it doesn't work23:43
Wizzupthen there is no point to set up audio23:43
Wizzupif it doesn't work23:43
sicelostun shouldn't be always needed. it's mainly required for NAT issues23:45
Wizzupright23:45
Wizzupbut most networks are nat23:46
siceloyeah, anyway easy to see if problems are stun related - if the SDP that shows up on the wire contains private IP, then yes you need stun23:48
arno11btw i've been able to make a sip call with no stun, same issue (no sound)23:48
Wizzupsdp?23:48
freemangordonI have no stun configured in fremantle23:49
freemangordonand was able to make call to android linphone23:49
Wizzupok23:50
Wizzupthen it's probably not stun23:50
freemangordonI'll put some more traces in vcm23:51

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