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b0bben	2014-07-04 , 19:52
Posts: 1 \| Thanked: 1 time \| Joined on Jul 2014 @ Gothenburg, Sweden	#61

@MSameer: would you be willing to help me get HLS going so I can do a native Plex app for Sailfish?

My GStreamer foo is weak

//b0bben

PS:
Long-time (like 3 years back) lurker on TMO, developed apps on N9 and Sailfish. Author of Plex on AppleTV and many other contributions to Plex. Friends with Plex team since day 1. Jolla fan.
Just registered

Last edited by b0bben; 2014-07-04 at 19:54.

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hemiwi	2015-10-21 , 18:56
Posts: 67 \| Thanked: 184 times \| Joined on Dec 2013 @ Helsinki	#62

Hello all,

i already asked in the Jolla Tablet experience Thread but unfortunately did not got an answer.

Is there anything i have to consider with installing additional codecs to Jolla tablet? Has anyone tried alreadyl?

Thanks very much in advance

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hemiwi	2015-11-21 , 18:43
Posts: 67 \| Thanked: 184 times \| Joined on Dec 2013 @ Helsinki	#63

^^ reup

Can someone who has the tablet already help me with the 1st steps to install additional codecs on the tablets i486 architecture?

Please

Pretty please

Last edited by hemiwi; 2015-11-21 at 18:53.

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faustomiglioresi	2015-12-22 , 22:55
Posts: 17 \| Thanked: 4 times \| Joined on Jun 2010	#64

how can i install the codecs?ii did not understand

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TheGrave	2016-02-23 , 17:04
Posts: 3 \| Thanked: 1 time \| Joined on Oct 2015	#65

I've been busting my head for a few days trying to figure out how to make telepathy-rakia to see all gstreamer codecs. My particular interest is G.729. Do you have any clue whether some config file has to be modified? farstream seems to have it enabled but Wireshark says it's not offered by the phone:

cat /usr/share/farstream/0.1/fsrtpconference/default-codec-preferences
################
# Audio codecs #
################

[audio/SPEEX:8000]
clock-rate=8000

[audio/SPEEX:16000]
clock-rate=16000

[audio/AMR]

[audio/G729]

[audio/ILBC]

Codec seems to be installed:

gst-inspect-0.10 | grep 729
rtp: rtpg729depay: RTP G.729 depayloader
rtp: rtpg729pay: RTP G.729 payloader

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llelectronics	2016-02-24 , 22:34
Posts: 426 \| Thanked: 1,812 times \| Joined on Dec 2013	#66

gst-inspect-0.10 | grep 729

SailfishOS now uses gstreamer 1.0 so the 0.10 command only shows the codecs support by 0.10 not the ones from 1.0

__________________
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TheGrave	2016-02-25 , 10:20
Posts: 3 \| Thanked: 1 time \| Joined on Oct 2015	#67

Originally Posted by llelectronics

SailfishOS now uses gstreamer 1.0 so the 0.10 command only shows the codecs support by 0.10 not the ones from 1.0

Not on 2.0.1.7. Are you running some later pre-release?

Anyway, what should be done in theory with either gstreamer 0.10 or 1.0 to get G.729 in Telepathy?

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llelectronics	2016-02-25 , 11:00
Posts: 426 \| Thanked: 1,812 times \| Joined on Dec 2013	#68

Since 2.0 SailfishOS uses gstreamer 1.2
I am not sure what telepathy utilizes

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	willi6868	2016-03-01 , 20:44
	Posts: 548 \| Thanked: 562 times \| Joined on Aug 2011 @ Germany	#69

There are some codecs compiled for the Tablet and avialiable on Openrepos now: https://openrepos.net/user/945/programs

__________________
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pasko	2016-03-03 , 18:39
Posts: 290 \| Thanked: 385 times \| Joined on Jan 2012 @ Madrid, Spain	#70

Originally Posted by TheGrave

I've been busting my head for a few days trying to figure out how to make telepathy-rakia to see all gstreamer codecs. My particular interest is G.729. Do you have any clue whether some config file has to be modified? farstream seems to have it enabled but Wireshark says it's not offered by the phone:

cat /usr/share/farstream/0.1/fsrtpconference/default-codec-preferences
################
# Audio codecs #
################

[audio/SPEEX:8000]
clock-rate=8000

[audio/SPEEX:16000]
clock-rate=16000

[audio/AMR]

[audio/G729]

[audio/ILBC]

Codec seems to be installed:

gst-inspect-0.10 | grep 729
rtp: rtpg729depay: RTP G.729 depayloader
rtp: rtpg729pay: RTP G.729 payloader

Hi.
I'm struggling to make VOIP calls work. I'm not able to make the sip client offer any codecs:

Code:

Content-Type: application/sdp
Content-Disposition: session
Content-Length: 124

v=0
o=- 2847608483721697075 2098627415485425572 IN IP4 172.26.0.16
s=-
c=IN IP4 172.26.0.16
t=0 0
m=audio 0 RTP/AVP 0

, and my SIP server drops the call just when I answer. I followed instrucctions in the Jolla Forum, but when I try to restart pulseaudio service I always get a "file not found" error.
These are the codecs offered by my Asterisk PBX:

Code:

Content-Type: application/sdp
Content-Length: 339

v=0
o=root 1205889408 1205889408 IN IP4 172.26.0.2
s=Asterisk PBX 11.7.0~dfsg-1ubuntu1
c=IN IP4 172.26.0.2
t=0 0
m=audio 14532 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Am I missing something? This is my 2nd day tinkering with the device, BTW

Regards.