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58 commits 44 branches 1508 tags 428 MiB
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10 commits

Author SHA1 Message Date
Jan Alexander Steffens (heftig)
fd1d53b04a on_server_message: Do not unref message GBytes 
We don't own the reference. Since GLib 2.58, the g_bytes_unref that
follows the signal emission in libsoup loudly complains about the
attempt to underflow the refcount.
 2018-09-21 13:12:43 +00:00
Eloi Bail
d6741c1f80 mp-webrtc-sendrecv.c: add missing comma in the list of package required 
A comma is missing in the list of package required. Thus the package
'srtprtpmanager' is checked instead of packages srtp and rtpmanager.
 2018-04-03 15:04:57 +00:00
Nirbheek Chauhan
82314cabbb Don't use strict ssl certificate checking for localhost 
When using localhost signalling servers, we don't want to use
strict ssl because it's probably using a self-signed certificate
and there's no need to do certificate checking over localhost anyway.
 2018-03-31 10:27:05 +05:30
Nirbheek Chauhan
0e1be2a63f Add Makefiles for all C demos 2018-03-23 19:00:37 +05:30
Nirbheek Chauhan
2d2bc0fe0e Fix compiler warnings in all C demos 2018-03-23 19:00:37 +05:30
Nirbheek Chauhan
55e86469d9 Check for all necessary plugins at startup 
People seem to be having problems ensuring that they have all the
right plugins built, so make it a bit easier for them.
 2018-03-10 01:54:48 +05:30
Nirbheek Chauhan
fa2adc717b Fix crash on Windows by delimiting option entries with NULL 
Also use more verbose forms of g_assert which print values on failure
 2018-03-08 20:10:55 +05:30
Mathieu Duponchelle
e5c5767298 Update to new promise API 2017-11-22 22:28:55 +10:00
Nirbheek Chauhan
0c5e799952 multiparty sendrecv: Add a queue before the audio sink 
Missed this, fixes the bug where removing a peer causes the pipeline to
get stuck. However, when peers leave, there is still a chance that the
pipeline will get stuck.
 2017-10-30 13:24:21 +05:30
Nirbheek Chauhan
9b1a0e5389 WIP: Add a new multiparty sendrecv gstreamer demo 
You can join a room and an audio-only call will be started with all
peers in that room. Currently uses audiotestsrc only.

BUG: With >2 peers in a call, if a peer leaves, the pipeline stops
     outputting data from the remaining peers to the (audio) sink.

TODO: JS code to allow a browser to join the call
TODO: Cleanup pipeline when a peer leaves
TODO: Add ICE servers to allow calls over the Internet
TODO: Perhaps setup a TURN server as well
 2017-10-30 09:14:29 +05:30