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path: root/plugins/rtp/src
Commit message (Collapse)AuthorAgeFilesLines
* Correctly handle missing webrtc-audio-processingMarvin W2021-05-012-3/+6
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* Echo CancellationMarvin W2021-05-014-14/+338
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* Handle non-existant call supportfiaxh2021-04-293-3/+20
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* Video optimizationsMarvin W2021-04-296-115/+386
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* Handle broken VAPI in older valaMarvin W2021-04-111-2/+19
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* GStreamer compatMarvin W2021-04-112-10/+24
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* Fix bug in legacy SRTP decryptionMarvin W2021-04-011-1/+6
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* Remove unnecessary debug codeMarvin W2021-04-011-4/+0
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* Migrate to libsrtp2Marvin W2021-03-291-29/+22
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* Don't reuse PTs for different media typesMarvin W2021-03-291-4/+4
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* Fix device manager usage for GStreamer 1.16Marvin W2021-03-261-2/+12
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* Move SRTP implementation into crypto library for reuseMarvin W2021-03-234-1031/+10
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* Resample audio data for common 48k sample rateMarvin W2021-03-232-10/+16
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* Add support for SRTPMarvin W2021-03-235-63/+1098
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* RTP: Backport gst_caps_copy_nth from GStreamer 1.16Marvin W2021-03-211-2/+10
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* Add RTP implementation as pluginMarvin W2021-03-218-0/+1712