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path: root/plugins/rtp/src
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* 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 (git 2.39.1) at 2026-05-20 22:44:47 +0000