aboutsummaryrefslogtreecommitdiff
path: root/plugins/rtp/src
Commit message (Expand)AuthorAgeFilesLines
* RTP: Handle missing rtp pay/depay elementsMarvin W2021-05-152-14/+16
* Calls: Use vp8depay.wait-for-keyframe only with GStreamer 1.16+Marvin W2021-05-111-1/+3
* Support voice processing on GStreamer 0.14Marvin W2021-05-022-16/+23
* 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 W2021-05-014-14/+338
* Handle non-existant call supportfiaxh2021-04-293-3/+20
* Video optimizationsMarvin W2021-04-296-115/+386
* Handle broken VAPI in older valaMarvin W2021-04-111-2/+19
* GStreamer compatMarvin W2021-04-112-10/+24
* Fix bug in legacy SRTP decryptionMarvin W2021-04-011-1/+6
* Remove unnecessary debug codeMarvin W2021-04-011-4/+0
* Migrate to libsrtp2Marvin W2021-03-291-29/+22
* 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
* Move SRTP implementation into crypto library for reuseMarvin W2021-03-234-1031/+10
* Resample audio data for common 48k sample rateMarvin W2021-03-232-10/+16
* Add support for SRTPMarvin W2021-03-235-63/+1098
* RTP: Backport gst_caps_copy_nth from GStreamer 1.16Marvin W2021-03-211-2/+10
* Add RTP implementation as pluginMarvin W2021-03-218-0/+1712
generated by cgit v1.2.3-54-g00ecf (git 2.45.2) at 2024-07-03 15:41:15 +0000