aboutsummaryrefslogtreecommitdiff
path: root/plugins/rtp
Commit message (Expand)AuthorAgeFilesLines
* Support voice processing on GStreamer 0.14Marvin W2021-05-023-19/+27
* Improve call wording, cleanupfiaxh2021-05-012-8/+8
* Fix webcam framerate selectionMarvin W2021-05-013-10/+35
* Correctly handle missing webrtc-audio-processingMarvin W2021-05-012-3/+6
* Echo CancellationMarvin W2021-05-015-16/+356
* Handle non-existant call supportfiaxh2021-04-293-3/+20
* Video optimizationsMarvin W2021-04-298-116/+1014
* Handle broken VAPI in older valaMarvin W2021-04-111-2/+19
* Fix custom vapi integrationMarvin W2021-04-111-2/+0
* GStreamer compatMarvin W2021-04-113-10/+30
* Fix bug in legacy SRTP decryptionMarvin W2021-04-011-1/+6
* Remove unnecessary debug codeMarvin W2021-04-011-4/+0
* Migrate to libsrtp2Marvin W2021-03-292-30/+23
* Don't reuse PTs for different media typesMarvin W2021-03-291-4/+4
* Fix device manager usage for GStreamer 1.16Marvin W2021-03-261-2/+12
* Add initial support for DTLS-SRTPfiaxh2021-03-251-0/+1
* Move SRTP implementation into crypto library for reuseMarvin W2021-03-235-1034/+12
* Resample audio data for common 48k sample rateMarvin W2021-03-232-10/+16
* Add support for SRTPMarvin W2021-03-236-66/+1103
* RTP: Backport gst_caps_copy_nth from GStreamer 1.16Marvin W2021-03-211-2/+10
* Add RTP implementation as pluginMarvin W2021-03-219-0/+1748
generated by cgit v1.2.3-70-g09d2 (git 2.47.0) at 2024-11-03 11:32:42 +0000