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<gnarface> antranigv: my guess is it would work the same way as devuan
<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)
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<sicelo> you probably are better off just bootstraping debian/devuan directly
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<antranigv> nice!
<antranigv> okay, devel is installed, updated, and now rebooting
<antranigv> too bad the power button doesn't work
<antranigv> and many of the old maemo apps are not available as well, I guess I should package them
<sicelo> power button doesn't work because?
<antranigv> 🤷‍♂️
<antranigv> I press it, it doesn't do anything
<antranigv> I think I bricked the device. oops. I'm getting "Nokia RX-51" :D
<antranigv> ok just boot stuck, I rebooted. huh! that was worrying for a second!
<antranigv> to be clear, I mean when I'm in leste, that's when the power button doesn't respond
<antranigv> wow it's much faster!
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<antranigv> does anyone know why I can't press 'enter' in profanity? did I forget something?
<sicelo> try Ctrl+M
<freemangordon> antranigv: power button should respond
<antranigv> sicelo thanks <3
<antranigv> totally forgot about keymaps
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<inky> antranigv: i think i change something in xsession in /etc/X11 to switch to wmaker.
<inky> comment a line, add wmaker line. don't remember.
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<Wizzup> freemangordon: so we keep 6.6 in devel or shall we push it to stable?
<freemangordon> нот суре, гижен тхе поверофф/ребоот иссуе
<freemangordon> oops
<freemangordon> not sure, given the poweroff/reboot issue
<Wizzup> ok
<Wizzup> many call things depend on it
<Wizzup> like, we can't even use tp-ring for calls without it
<freemangordon> if we can live with that for stable, then yes
<freemangordon> ok, go for stable then
<freemangordon> but someone shall look at that issue
<Wizzup> so far it's been a heisenbug, every time we look we can't reproduce
<Wizzup> maybe after the end of the month
<freemangordon> I may look at it earlier is I find some spare time
<freemangordon> *if I
<Wizzup> given the nlnet deadline is end of month, maybe we can try to do as muhc there as possible :D
<freemangordon> right
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<freemangordon> any SIP client for android to recommend?
<freemangordon> Wizzup: ^^^?
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<Wizzup> freemangordon: don't know sorry
<Wizzup> we use twinkle with success on leste
<Wizzup> freemangordon: also fremantle has working sip
<freemangordon> oh
<freemangordon> ok
<Wizzup> I can give you my fremantle setting in a bit
<Wizzup> (if you need them)
<freemangordon> sure, I have some experience with sip, but not on fremantle
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<Wizzup> will send over dm
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<freemangordon> Wizzup: why do you think tp-rakia uses gst?
<Wizzup> for farstream
<Wizzup> does it not?
<Wizzup> my gst comments might have been about gabble
<freemangordon> farstream is skype, no?
<Wizzup> no
<freemangordon> anyway, I think tp managers does not take care about media
<Wizzup> it looks like we don't have a fork for telepathy-rakia yeah heh
<Wizzup> farstream is part of telepathy FWIW
<freemangordon> it is the client that shall take care
<Wizzup> freemangordon: what do you mean, does not take care about media
<freemangordon> codecs are not in TP
<Wizzup> clearly arno already hears one side now with tp-rakia, so I would be surprised
<freemangordon> how's that?
<Wizzup> well, he makes a call with sphone with sip and gets one way audio
<Wizzup> let me clone tp rakia src, I don't think we have it
<freemangordon> who cares about org.freedesktop.Telepathy.Call1.Content.MediaDescription.Codecs
<Wizzup> voicecall-manager presumably
<freemangordon> I cloned it, there is nothing about any audio processing there as far as I can find
<freemangordon> aaah
<freemangordon> right
<Wizzup> plugins/providers/telepathy/src/farstreamchannel.cpp
<freemangordon> that's the 'client' I am talking about
<Wizzup> on voicecall repo
<Wizzup> yeah, ok :)
<freemangordon> ok, lemme see what happens there
<Wizzup> ty :)
<freemangordon> was there any special trick to enable vcm debug logs?
<freemangordon> ok "Got audio sink, initializing audio input"
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<kiva> starting calendar from calendar-widget works on pinephone now, thanks
<Wizzup> freemangordon: let me check
<Wizzup> freemangordon: I used this
<Wizzup> QT_FORCE_STDERR_LOGGING=1 QT_LOGGING_RULES='org.nemomobile.voicecall.debug=true' G_MESSAGES_DEBUG=all GST_DEBUG=3 /usr/bin/voicecall-manager
<kiva> but calendar week numbers does not align well to lines in PP 1440x720 screen.
<kiva> there is 6 week numbers and only 5 week lines.
<freemangordon> Wizzup: seems we are missing codecs
<freemangordon> lemme check where does it look for them
<freemangordon> Wizzup: any clue where shall I look for fsrtpconference_disco
<freemangordon> hmm, farsight
<kiva> actually there is still calendar widget bug (starting calendar pushing widget)..it only worked two times..strange.
<freemangordon> yes, known issue
<Wizzup> freemangordon: yes, you might need to install some gst codecs
<Wizzup> I don't know exactly which of the top of my head, but I believe I installed a bunch
<Wizzup> the codecs should all be gst plugins
<Wizzup> for example gstreamer1.0-nice for ICE
<freemangordon> ok
<Wizzup> let me see
<Wizzup> freemangordon: ok, let's see if we can get those all installed
<freemangordon> yes, trying to
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<Wizzup> hard to find..
<freemangordon> seems most are in gst-libav
<freemangordon> but lemme check how farstream tries to init them
<Wizzup> ok, I do have that one
<Wizzup> maybe some are also libgstrtp.so ?
<Wizzup> work mtg, back in a bit
<freemangordon> maybe
<freemangordon> no, it is some other issue
<antranigv> guuuuuys
<antranigv> I broke something
<antranigv> :D
<antranigv> it can't mount / properly, I guess? but I'm probably in initramfs? I assume
<antranigv> okay so the root filesystem is in RO mode
<freemangordon> Wizzup: 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 ];
<freemangordon> GST_PADS gstpad.c:2527:gst_pad_link_full: link between audioconvert98:src and rtpg729pay1:sink failed: no common format
<freemangordon> hmm, I wonder if PA allows 8000 Hz
<arno11> ah, so it tries to use g729, right ?
<freemangordon> it tries to use 6 codecs
<freemangordon> all fail
<arno11> ok
<arno11> even PCMA ?
<freemangordon> see https://pastebin.com/PnpSF5Dg
<freemangordon> it does not try PCMA
<freemangordon> this is caling maemo fremantle->maemo leste
<freemangordon> so I guess those are fremantle supported codecs
<freemangordon> but codecs should not be an issue
<freemangordon> I see we have codecs
<freemangordon> but for some reason gst negotiation fail;s
<arno11> ok btw 8000Hz should work
<freemangordon> on d4?
<arno11> ah, on d4 i don't know
<arno11> on n900 it works because we have a custom daemon.conf
<freemangordon> well, it goes through audioresample
<freemangordon> so it should be ok
<freemangordon> but I wonder if it is :)
<freemangordon> lemme try something
<antranigv> okay eMMC boots fine, but the look of it
<antranigv> but when I boot from SD card I can't mount /
<arno11> antranigv: you mean you want to mount emmc from leste, right ?
<antranigv> arno11 no no no
<antranigv> when I choose "boot from SD card" in uboot, I get the console and then *everything* fails
<antranigv> and finally I get "login"
<arno11> weird
<antranigv> bad SD card? not sure, I can login fine and see everything
<antranigv> I have a video, should I share
<antranigv> ?
<Wizzup> freemangordon: ok, back, how can I help?
<Wizzup> freemangordon: to me 'no common format' means there is no codec that can be agreed upon with the remote party
<Wizzup> I think you can record any freq you want from pa, it will just resample
<sicelo> do we have g.729 in gst+devuan-chimaery? it was proprietary before, and only freed in last couple of years
<freemangordon> gst-launch-1.0 audiotestsrc ! capsfilter caps=audio/x-raw,rate=8000,channels=1 ! audioconvert ! rtpg729pay ! fakesink
<freemangordon> this fails
<arno11> i tried different codecs and atm only PCMA, PCMU work between leste and android. g729 doesn't work when requested by android
<Wizzup> with this?
<Wizzup> WARNING: erroneous pipeline: could not link audioconvert0 to rtpg729pay0
<Wizzup> arno11: right but perhaps all pipelines fail for the same reason, so if we can debug one...
<arno11> ok indeed
<sicelo> antranigv: you can share ...
<freemangordon> Wizzup: yes, that error
<freemangordon> the same error sip call fails with
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* Wizzup checks gst-inspect-1.0 audioconvert
<freemangordon> hmm, I see avdec_g729
<freemangordon> but no encoder
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<Wizzup> what is the audioconvert supposed to do
<Wizzup> does it figure out how to convert to rtpg729pay by querying it?
<freemangordon> not sure
<freemangordon> lemme check
<freemangordon> no
<Wizzup> SINK template: 'sink'
<Wizzup> Capabilities:
<Wizzup> Availability: Always
<Wizzup> audio/G729
<Wizzup> channels: 1
<Wizzup> rate: 8000
<Wizzup> this is the sink template for rtpg729pay
<freemangordon> yes, I see
<freemangordon> and I wonder how is that supposed to work
<freemangordon> bug in farstream?
<Wizzup> maybe
<freemangordon> audioconvert cannot convert to sink requirements
<Wizzup> let's see if we can make it work standalone
<freemangordon> mhm
<freemangordon> encodebin?
<Wizzup> maybe my sink desc is wrong
<Wizzup> brb.
<freemangordon> no, it is the same here
<Wizzup> yes but it is src
<Wizzup> not sink
<Wizzup> see
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<Wizzup> Pad Templates:
<Wizzup> SRC template: 'src'
<Wizzup> Availability: Always
<Wizzup> ...
<freemangordon> ST_DEBUG=4 gst-launch-1.0 audiotestsrc ! capsfilter caps=audio/x-raw,rate=8000,channels=1 ! audioconvert ! alawenc ! rtppcmapay ! fakesink
<freemangordon> this works
<Wizzup> one sec
<Wizzup> b
<Wizzup> ok, great
<freemangordon> not really, as encodebin should do it for us
<Wizzup> what is alawenc?
<Wizzup> should do what for us?
<freemangordon> "Convert 16bit PCM to 8bit A law"
<freemangordon> or PCMA
<freemangordon> hmm, lemme see if fremantle offers PCMA
<Wizzup> ok so one change was to make the right caps, yeah?
<Wizzup> or rather, why did you swap out rtpg729pay ?
<Wizzup> and does encodebin construct this?
<freemangordon> I did swap rtpg729pay because I did not find the codec for G729
<Wizzup> ok
<freemangordon> but I found the codec for rtppcmapay (PCMA)
<freemangordon> and no, encoderbin does not put alawenc for us
<freemangordon> lemme see how exactly farstream decides which bins to create
<freemangordon> oh
<freemangordon> "GST_PADS gstpad.c:2585:gst_pad_link_full: linked audioconvert78:src and alawenc3:sink, successful"
<freemangordon> so, seems arno11 is right and PCMA works
<Wizzup> and this already works in the vm before the research we did, or?
<freemangordon> I test on d4
<freemangordon> no idea
<Wizzup> ok, d4 then
<Wizzup> from your paste it looks like we could not find any codec
<freemangordon> lemme try again
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<Wizzup> maybe the farstreamchannel.cpp isn't updated for more recent gst
<freemangordon> no, this happens il libfarstream
<freemangordon> *in
<Wizzup> ok
<freemangordon> and actually it seems it negotiates PCMA/PCMU/speex and one more
<freemangordon> no, lemme see if gst pa plugin works
<freemangordon> it does
<Wizzup> standalone you mean?
<Wizzup> where is libfarstream source
<freemangordon> apt-get source
<freemangordon> apt-get source libfarstream-0.2-5
<freemangordon> but anyway, it seems some PA interaction issue
<freemangordon> as there are no pa sink/src
<Wizzup> hm..
<freemangordon> hmm...
<freemangordon> there is "playback manager" stream
<Wizzup> where do you see it and what is it a part of
<Wizzup> but 'pulsesrc' works for you and grabs the mic yeah?
<freemangordon> pavucontrol
<freemangordon> yes, it works
<freemangordon> but lemme check the flags of it
<freemangordon> maybe it does not allow non-native freqs
<Wizzup> so this playback manager is farstream?
<Wizzup> I'll just let you do this, let me know how I can help
<freemangordon> ok
<freemangordon> yes playback-manager is farstream
<freemangordon> oh, sorry
<freemangordon> "voice manager"
<arno11> Wizzup: btw, shall i do a PR for mic stuff in hifi profile, or ?
<arno11> (it makes twinkle sip calls working ootb)
<freemangordon> hmm, pulse stuff is added by vcm it seems
<Wizzup> freemangordon: the pulsesrc and pulsesink?
<freemangordon> yes
<Wizzup> well if you get the voice manager I think that is a step in the right direction, I don't think I ever got this
<Wizzup> but I am not sure what the current error is that you see
<freemangordon> it happens only fisrt time after vcm is started
<freemangordon> there is no error
<freemangordon> I wonder what are the volume
<freemangordon> *volumes
<freemangordon> lemme play a bit with alsa
<Wizzup> I think it adds a volume element at least
<Wizzup> in FarstreamChannel::initAudioInput
<freemangordon> mhm
<freemangordon> and also AUDIO_SINK_ELEMENT/AUDIO_SOURCE_ELEMENT
<Wizzup> yeah
<freemangordon> I wonder how vcm connects its pipeline with farstream pipeline
<freemangordon> FarstreamChannel::addAndLink: binobj= audio-output-bin src= (NULL) dst= queue1
<freemangordon> so, how that gets connected to farstream pipeline?!?
<antranigv> ah, stupid video file
<antranigv> ok uploading
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<freemangordon> Wizzup: I guess in onFsConferenceAdded(), but I don't see that called
<Wizzup> if (!gst_bin_add(binobj, ret)) {
<Wizzup> setError(QLatin1String("Could not add to bin "));
<Wizzup> return 0;
<Wizzup> gst_object_unref(ret);
<Wizzup> doesn't this do it?
<Wizzup> and then the linking
<Wizzup> in FarstreamChannel::addAndLink
<freemangordon> no, no, see what onFsConferenceAdded does
<Wizzup> arno11: when you said tls doesn't work, did you write sips as url scheme
<Wizzup> arno11: nevermind for now I think
<Wizzup> freemangordon: I don't know what this does, I assumed itwas for calls were there are more than two people
<Wizzup> it does the same gst bin add
<arno11> Wizzup: yes, as url scheme
<freemangordon> Wizzup: ok, it is called
<freemangordon> onFsConferenceAdded that is
<Wizzup> ok
<freemangordon> this is where it (presumably)links src/sink with fs pipeline(s)
<Wizzup> I don't see it actually
<Wizzup> the linking hppens in addAndLink as far as I can see
<Wizzup> but it sets it to a playing state there I guess
<freemangordon> see gboolean res = gst_bin_add(GST_BIN(self->mGstPipeline), GST_ELEMENT(conf));
<freemangordon> conf is FS pipeline
<freemangordon> IIUC
<Wizzup> but isn't there the same in addAndLink
<Wizzup> if (!gst_bin_add(binobj, ret)) {
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<freemangordon> it is, but addAndLink is used for elements we create
<freemangordon> like pulsesink/pulsesrc
<Wizzup> right
<freemangordon> so, whatever happens:
<freemangordon> GST_STATES gstelement.c:2769:gst_element_continue_state:<send_tee_1> completed state change to PLAYING
<freemangordon> what about media role?
<freemangordon> see setPhoneMediaRole()
<antranigv> any thoughts?
<Wizzup> antranigv: we're kind of in the middle of making sip calls work, will check in a bit
<Wizzup> freemangordon: I saw the media role thing, I don't think it should matter
<antranigv> Wizzup nice! can I join? :D
<Wizzup> what do you mean, join?
<freemangordon> Wizzup: also:
<freemangordon> I: [pulseaudio] sink.c: Cannot update sample spec, monitor source is RUNNING
<freemangordon> module-stream-restore.id = "sink-input-by-media-role:phone"
<freemangordon> I am almost sure volume for that role is 0
<Wizzup> antranigv: the video you posted, did you change /etc/fstab?
<antranigv> Wizzup I did, I think I disabled swapon, or enabled it, or something, not sure
<Wizzup> freemangordon: ok, that is a problem
<Wizzup> antranigv: this is probably your problem
<antranigv> what should it be like?
<Wizzup> what it was before... :)
<Wizzup> not sure what you changed
<antranigv> I'll try now!
<freemangordon> Wizzup: it would not have been like that, if it was not removed from volume-control appet :p
<freemangordon> *applet
<Wizzup> mhm
<Wizzup> I didn't think the media role assigned a volume
<Wizzup> I thought it was just used to cork streams
<Wizzup> ok
<Wizzup> if this is the problem I'll be sad
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<freemangordon> maybe just comment that code and test
<freemangordon> in setPhoneMediaRole() that is
<freemangordon> Wizzup: how would not media role assign a volume? like, playing music and making phone call must have different volumes
<freemangordon> switched automatically as soon as you answer the phone
<freemangordon> or hangup
<freemangordon> but anyway, I am not sure that's the proble
<freemangordon> m
<antranigv> okay, 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 do
<antranigv> wait, do I need `-u` to "update"?
<antranigv> nah, that's a Solaris thing, not a GNU thing, I think
<Wizzup> antranigv: no, your fstab is probably wrong
<Wizzup> try to mount your sd card on your computer
<antranigv> Wizzup I see what you mean. I'll fix via the computer.
<antranigv> now I just need something that can mount ext4 :D
<antranigv> ah right! maybe I can try mounting it from regular maemo!
<freemangordon> Wizzup: volume: mono: 65536 / 100% / 0.00 dB
<freemangordon> looks kile volume is ok
<freemangordon> *like
<Wizzup> antranigv: you won't be able to from regular maemo I think
<Wizzup> freemangordon: ok, so where are you at now
<Wizzup> did you make any changes, do you hear something?
<freemangordon> no changes
<freemangordon> still nothing
<freemangordon> and, there is no PA source, only sink
<freemangordon> also, I run out of time
<freemangordon> :)
<freemangordon> however, there is no obvious error
<freemangordon> maybe I shall try in VM, to ignore d4 audio issues
<Wizzup> no pa source would be a big problem
<Wizzup> freemangordon: so you also didn't install extra gst plugins?
<freemangordon> yes
<Wizzup> you can try vm but make sure it's not some double/trip NAT thing
<freemangordon> btw, how to disable sphone rotation?
<Wizzup> see leste-config-n900
<Wizzup> leste-config-n900/usr/share/sphone/sphone.ini.d/landscape-call.ini.leste
<Wizzup> [Gui]
<Wizzup> LandscapeCall=1
<freemangordon> thanks
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<freemangordon> well, does not work
<freemangordon> still rotates
<sicelo> LandscapeCalls ...
<freemangordon> heh
<Wizzup> oh sorry
<Wizzup> I just pasted from the commit
<freemangordon> yeah, that works
<freemangordon> anyway, enough for today
<freemangordon> will just try to disable media role in vm
<Wizzup> ok
<freemangordon> hard to say, streams are not created
<freemangordon> this needs further debugging
<freemangordon> but not now, perhaps on Sunday/Monday
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<Wizzup> ok :)
<Wizzup> I wonder if it is related to stun or something, it doesn't work
<Wizzup> then there is no point to set up audio
<Wizzup> if it doesn't work
<sicelo> stun shouldn't be always needed. it's mainly required for NAT issues
<Wizzup> right
<Wizzup> but most networks are nat
<sicelo> yeah, 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 stun
<arno11> btw i've been able to make a sip call with no stun, same issue (no sound)
<Wizzup> sdp?
<freemangordon> I have no stun configured in fremantle
<freemangordon> and was able to make call to android linphone
<Wizzup> ok
<Wizzup> then it's probably not stun
<freemangordon> I'll put some more traces in vcm
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<Wizzup> freemangordon: 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?