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Commit message (Collapse)AuthorAgeFilesLines
* RTP: Fix GStreamer version checkMarvin W2021-05-141-1/+1
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* Calls: Use vp8depay.wait-for-keyframe only with GStreamer 1.16+Marvin W2021-05-111-1/+3
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* Support voice processing on GStreamer 0.14Marvin W2021-05-023-19/+27
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* Improve call wording, cleanupfiaxh2021-05-012-8/+8
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* Fix webcam framerate selectionMarvin W2021-05-013-10/+35
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* Correctly handle missing webrtc-audio-processingMarvin W2021-05-012-3/+6
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* Echo CancellationMarvin W2021-05-015-16/+356
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* Handle non-existant call supportfiaxh2021-04-293-3/+20
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* Video optimizationsMarvin W2021-04-298-116/+1014
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* Handle broken VAPI in older valaMarvin W2021-04-111-2/+19
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* Fix custom vapi integrationMarvin W2021-04-111-2/+0
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* GStreamer compatMarvin W2021-04-113-10/+30
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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 W2021-03-292-30/+23
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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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* Add initial support for DTLS-SRTPfiaxh2021-03-251-0/+1
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* Move SRTP implementation into crypto library for reuseMarvin W2021-03-235-1034/+12
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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 W2021-03-236-66/+1103
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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 W2021-03-219-0/+1748
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