23 commits

Author	SHA1	Message	Date
Marvin W	bec73ed371	RTP: Fix GStreamer version check	2021-05-14 10:17:51 +02:00
Marvin W	3bfd407843	Calls: Use vp8depay.wait-for-keyframe only with GStreamer 1.16+	2021-05-11 22:11:44 +02:00
Marvin W	8044b546d0	Support voice processing on GStreamer 0.14	2021-05-02 18:03:03 +02:00
fiaxh	7d2e647690	Improve call wording, cleanup	2021-05-01 21:51:24 +02:00
Marvin W	0409f55426	Fix webcam framerate selection	2021-05-01 17:27:55 +02:00
Marvin W	d388525fc6	Correctly handle missing webrtc-audio-processing	2021-05-01 16:00:37 +02:00
Marvin W	23ffd37dde	Echo Cancellation	2021-05-01 15:48:51 +02:00
fiaxh	5d85b6cdb0	Handle non-existant call support	2021-04-29 16:13:25 +02:00
Marvin W	3880628de4	Video optimizations	2021-04-29 15:53:59 +02:00
Marvin W	fe160d94ba	Handle broken VAPI in older vala	2021-04-11 16:28:59 +02:00
Marvin W	4edab3c8d6	Fix custom vapi integration	2021-04-11 15:13:22 +02:00
Marvin W	6ebdec1d78	GStreamer compat	2021-04-11 12:31:03 +02:00
Marvin W	c5ab4fed87	Fix bug in legacy SRTP decryption	2021-04-01 11:51:35 +02:00
Marvin W	c5cb43350a	Remove unnecessary debug code	2021-04-01 11:51:12 +02:00
Marvin W	5e58f29883	Migrate to libsrtp2	2021-03-29 13:20:12 +02:00
Marvin W	9520a81b81	Don't reuse PTs for different media types	2021-03-29 13:14:37 +02:00
Marvin W	fc3263d49e	Fix device manager usage for GStreamer 1.16	2021-03-26 15:18:04 +01:00
fiaxh	ec35f95e13	Add initial support for DTLS-SRTP	2021-03-25 14:45:54 +01:00
Marvin W	4b230808b9	Move SRTP implementation into crypto library for reuse	2021-03-23 20:04:28 +01:00
Marvin W	b01f6f9ef7	Resample audio data for common 48k sample rate	2021-03-23 15:11:00 +01:00
Marvin W	b393d41601	Add support for SRTP	2021-03-23 15:11:00 +01:00
Marvin W	cde1e38f5d	RTP: Backport gst_caps_copy_nth from GStreamer 1.16	2021-03-21 15:43:54 +01:00
Marvin W	ef2e3c774c	Add RTP implementation as plugin	2021-03-21 12:41:38 +01:00