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path: root/plugins/rtp
AgeCommit message (Expand)Author
2021-11-10Make elements sync to get proper qos dataMarvin W
2021-11-10RTP: Make opus mono-channelMarvin W
2021-11-10RTP: Only start gstreamer pipeline once neededMarvin W
2021-11-10RTP: Encode with deviceMarvin W
2021-11-10Split payloader off encoder chainMarvin W
2021-11-10Improve codec supportMarvin W
2021-11-10Crop video to match widget ratioMarvin W
2021-10-12Fix compiler warnings ('Switch does not handle .. of enum ..')fiaxh
2021-10-12Fix misc compiler warningsfiaxh
2021-05-15RTP: Handle missing rtp pay/depay elementsMarvin W
2021-05-14RTP: Fix GStreamer version checkMarvin W
2021-05-11Calls: Use vp8depay.wait-for-keyframe only with GStreamer 1.16+Marvin W
2021-05-02Support voice processing on GStreamer 0.14Marvin W
2021-05-01Improve call wording, cleanupfiaxh
2021-05-01Fix webcam framerate selectionMarvin W
2021-05-01Correctly handle missing webrtc-audio-processingMarvin W
2021-05-01Echo CancellationMarvin W
2021-04-29Handle non-existant call supportfiaxh
2021-04-29Video optimizationsMarvin W
2021-04-11Handle broken VAPI in older valaMarvin W
2021-04-11Fix custom vapi integrationMarvin W
2021-04-11GStreamer compatMarvin W
2021-04-01Fix bug in legacy SRTP decryptionMarvin W
2021-04-01Remove unnecessary debug codeMarvin W
2021-03-29Migrate to libsrtp2Marvin W
2021-03-29Don't reuse PTs for different media typesMarvin W
2021-03-26Fix device manager usage for GStreamer 1.16Marvin W
2021-03-25Add initial support for DTLS-SRTPfiaxh
2021-03-23Move SRTP implementation into crypto library for reuseMarvin W
2021-03-23Resample audio data for common 48k sample rateMarvin W
2021-03-23Add support for SRTPMarvin W
2021-03-21RTP: Backport gst_caps_copy_nth from GStreamer 1.16Marvin W
2021-03-21Add RTP implementation as pluginMarvin W
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