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979 commits 1 branch 0 tags 6.6 MiB
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39 commits

Author SHA1 Message Date
Marvin W 9aeff4bf9e SRTP: Do not continue processing data after encrypt/decrypt failed 
RTP: Copy less
 2021-12-18 21:45:36 +01:00
Marvin W 4f80a9f5cc RTP: Correctly handle timestamp after re-enabling a stream 2021-12-18 21:43:12 +01:00
fiaxh 2b3d150949 Improve call details dialog + small multi-party call fixes 2021-11-15 13:29:13 +01:00
Marvin W ec65415186
Optimize encoder for low cpu usage 2021-11-15 23:49:48 +01:00
Marvin W 0b828a0ae5
Add maximum bitrate and adjust video resolution based on bitrate 2021-11-15 22:49:44 +01:00
Marvin W 9958cfbe7b
Log probe for decode QOS 2021-11-11 22:49:48 +01:00
Marvin W 9e5a3895ae
Limit REMB target bitrate to 2x maximum actually seen value 2021-11-11 22:35:45 +01:00
fiaxh e205743f0c Display target bitrates in connection details UI 2021-11-11 21:54:55 +01:00
Marvin W 1b157a20ab
Fix REMB calculation 2021-11-10 23:13:33 +01:00
Marvin W cfe43de5d5
Make elements sync to get proper qos data 2021-11-10 23:13:33 +01:00
Marvin W f398135bc8 RTP: Make opus mono-channel 2021-11-10 11:05:34 +01:00
Marvin W ea19a9c5cb RTP: Only start gstreamer pipeline once needed 2021-11-10 11:05:34 +01:00
Marvin W b593aa05ef RTP: Encode with device 2021-11-10 11:05:34 +01:00
Marvin W 083f73b0ca Split payloader off encoder chain 2021-11-10 11:05:34 +01:00
Marvin W 72569ea52f Improve codec support 2021-11-10 11:05:34 +01:00
Marvin W aae13b9ea6 Crop video to match widget ratio 2021-11-10 11:05:34 +01:00
fiaxh 237081e573 Fix compiler warnings ('Switch does not handle .. of enum ..') 2021-10-12 19:43:57 +02:00
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