Daniel Gultsch
59ea66ca78
make sure VideoSourceWrapper is stored in property
2022-11-19 14:19:07 +01:00
Daniel Gultsch
27d8da2ab4
refactor WebRTCWrapper to allow for track adds
2022-11-19 13:03:34 +01:00
Daniel Gultsch
6ececb4d2b
refactor webrtc video source + capture code
2022-11-12 13:37:56 +01:00
Stephen Paul Weber
78048bbd3d
Enable WebRTC-BindUsingInterfaceName/Enabled/
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This makes 464XLAT networks (such as T-Mobile LTE) work.
https://bugs.chromium.org/p/webrtc/issues/detail?id=10707
2022-03-10 16:29:00 +01:00
Daniel Gultsch
b6442c0bd4
add Samsung S4 to hardware aec blacklist
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fixes #4267
2022-01-18 11:30:23 +01:00
Daniel Gultsch
a508a81553
externalize rtc config generation into seperate method
2021-11-17 11:33:15 +01:00
Daniel Gultsch
abb671616c
synchronize setDescription calls
2021-11-16 15:17:12 +01:00
Daniel Gultsch
297a843b9c
use implicit rollback (needed to be enabled on libwebrtc)
2021-11-16 13:17:10 +01:00
Daniel Gultsch
5b80c62a63
treat transport-info w/o candidates and changed credentials as offer
2021-11-14 18:22:18 +01:00
Daniel Gultsch
717c83753f
delay discovered ice candidates until JingleRtpConnection is ready to receive
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otherwise setLocalDescription and the arrival of first candidates might race
the rtpContentDescription being set
2021-11-11 21:02:17 +01:00
Daniel Gultsch
b6dee6da6a
reverse: webrtc: include oldState in onConnectionChange
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turns out we don’t need it and a better way is for RtpConnection to keep track of *all*
states in the current generation
2021-11-11 17:05:36 +01:00
Daniel Gultsch
9843b72f6f
always use Camera2Enumerator
2021-11-11 15:23:45 +01:00
Daniel Gultsch
61851e5f84
do not automacially hang up failed webrtc sessions
2021-11-11 14:40:15 +01:00
Daniel Gultsch
4ec0996dff
webrtc: include oldState in onConnectionChange
2021-11-11 11:19:37 +01:00
Daniel Gultsch
fda45a7c86
use implicit descriptions for WebRTC
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using the parameter-free form of setLocalDescription() solves some potential race conditions
that can occur - especially once we introduce restartIce()
2021-11-10 16:40:24 +01:00
Daniel Gultsch
bae9fc45c2
make rtcpMux optional
2021-10-31 10:20:58 +01:00
Daniel Gultsch
6d91551f59
use onAddTrack instead of deprecated onAddStream
2021-05-03 13:06:42 +02:00
Daniel Gultsch
30c9e7399e
log track class in onAddTrack
2021-03-29 10:57:56 +02:00
Daniel Gultsch
1822a71c2a
Do not crash when receiving video call on device w/o camera
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Upon accepting a video call on a device that can not establish a video track on
its own (for example by not having a camera), displaying the video enable/disable
button would fail. This commit defaults this button to disabled.
2021-03-26 12:54:26 +01:00
Daniel Gultsch
3baacf8862
switch to unified plan
2021-03-16 18:52:38 +01:00
Daniel Gultsch
637c0cb15a
fixed rare race condition when receiving transport info right after WebRTCWrapper closes
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fixes #3849
2020-08-01 14:18:03 +02:00
Daniel Gultsch
bf85a55930
catch NPE when detecting camera facing. fixes #3820
2020-07-09 20:11:09 +02:00
Daniel Gultsch
6a6c9fb3bf
ignore race condition when toggling fixes #3822
2020-07-09 19:14:28 +02:00
Daniel Gultsch
dddb7ece25
show app failure instead of crashing when egl fails to init. fixes #3795
2020-06-18 20:37:56 +02:00
Daniel Gultsch
1c625e55a0
set candidate gathering to continually. fixes #3719
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This should be good enough to survive some network switches where both networks are online at the same time to allow for some handover
(for example when enabling wifi the 3G connection will usually (probably depends on OS) live on for a moment
2020-05-25 11:11:29 +02:00
Daniel Gultsch
685e01e83f
give TonManager control over audio mode to play dial tones on earpiece. fixes #3738
2020-05-21 15:39:59 +02:00
Daniel Gultsch
a2a7256682
disable hardware AEC on some devices. fixes #3734
2020-05-21 11:13:46 +02:00
Daniel Gultsch
f4247379bd
catch UnsatisfiedLinkError when trying to init libwebrtc. fixes #3707
2020-05-09 19:48:54 +02:00
Daniel Gultsch
3577afea4e
fixed crash caused by race when dedecting if mic is on
2020-05-03 11:54:58 +02:00
Daniel Gultsch
e70b6eec98
do not mirror back camera. fixes #3693
2020-05-03 11:54:58 +02:00
Daniel Gultsch
63ddd97b6b
add button to switch cameras during video call
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RIP symmetry :-(
fixes #3683
2020-05-02 17:15:50 +02:00
Daniel Gultsch
8a586527c4
check if setting local description was succesful
2020-04-29 15:32:27 +02:00
Daniel Gultsch
333f509e53
synchronize public WebRTCWrapper methods so closes don’t race
2020-04-29 09:10:15 +02:00
Daniel Gultsch
f93bac6d73
catch ISE around peerconnection.dispose()
2020-04-28 20:15:23 +02:00
Daniel Gultsch
80cac3bd69
disable tcp candidates
2020-04-23 19:51:58 +02:00
Daniel Gultsch
5b12e23382
improve logging for throws from native callbacks
2020-04-21 12:00:13 +02:00
Daniel Gultsch
7898ba65cd
extend extended webrtcwrapper logging
2020-04-20 17:05:27 +02:00
Daniel Gultsch
1cc0dfad84
move sdp logging to different tag
2020-04-20 15:57:31 +02:00
Daniel Gultsch
7dfd47a5c4
better crash than leave WebRTCWrapper unclosed
2020-04-20 15:57:30 +02:00
Daniel Gultsch
16d34c2ba0
parse turns and stuns (regression from earlier commit)
2020-04-20 15:57:30 +02:00
Daniel Gultsch
45d5d1f635
capture in ~1920 resolution when available
2020-04-20 15:57:30 +02:00
Daniel Gultsch
01a9a52990
show enable/disable video in video calls
2020-04-20 15:57:30 +02:00
Daniel Gultsch
445009c558
request camera permissions
2020-04-20 15:57:30 +02:00
Daniel Gultsch
d057ae3439
transmit media from proposal to actual session
2020-04-20 15:57:30 +02:00
Daniel Gultsch
1489dba44f
release resource. stop caputuring when webrtc ends
2020-04-20 15:57:30 +02:00
Daniel Gultsch
339bdaea06
rudimentary video caputuring
2020-04-20 15:57:30 +02:00
Daniel Gultsch
e16e0d895e
cancle ongoing jingle sessions on xmpp rebind
2020-04-20 15:57:30 +02:00
Daniel Gultsch
ef22071bd1
turn proximity wake lock and/off depending on speaker configuration
2020-04-20 15:57:30 +02:00
Daniel Gultsch
981aeaf264
make mute and speaker button work
2020-04-20 15:57:30 +02:00
Daniel Gultsch
b924a63d01
copy audio manager from AppRTCDemo
2020-04-20 15:57:30 +02:00