Commit graph

72 commits

Author SHA1 Message Date
fiaxh e8c162eae3 Fix misc compiler warnings 2021-10-12 19:43:57 +02:00
Marvin W 686035ca1e
RTP: Handle missing rtp pay/depay elements 2021-05-15 19:55:44 +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 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
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