aboutsummaryrefslogtreecommitdiff
path: root/plugins/rtp
Commit message (Collapse)AuthorAgeFilesLines
* Optimize encoder for low cpu usageMarvin W2021-11-151-5/+5
|
* Add maximum bitrate and adjust video resolution based on bitrateMarvin W2021-11-154-12/+111
|
* Log probe for decode QOSMarvin W2021-11-111-0/+40
|
* Limit REMB target bitrate to 2x maximum actually seen valueMarvin W2021-11-111-32/+48
|
* Display target bitrates in connection details UIfiaxh2021-11-111-9/+8
|
* Fix REMB calculationMarvin W2021-11-101-2/+5
|
* Make elements sync to get proper qos dataMarvin W2021-11-102-3/+3
|
* RTP: Make opus mono-channelMarvin W2021-11-101-2/+2
|
* RTP: Only start gstreamer pipeline once neededMarvin W2021-11-101-48/+66
|
* RTP: Encode with deviceMarvin W2021-11-102-174/+413
|
* Split payloader off encoder chainMarvin W2021-11-101-4/+39
|
* Improve codec supportMarvin W2021-11-101-4/+7
|
* Crop video to match widget ratioMarvin W2021-11-101-29/+45
|
* Fix compiler warnings ('Switch does not handle .. of enum ..')fiaxh2021-10-121-0/+2
|
* Fix misc compiler warningsfiaxh2021-10-121-1/+1
|
* RTP: Handle missing rtp pay/depay elementsMarvin W2021-05-152-14/+16
|
* RTP: Fix GStreamer version checkMarvin W2021-05-141-1/+1
|
* Calls: Use vp8depay.wait-for-keyframe only with GStreamer 1.16+Marvin W2021-05-111-1/+3
|
* Support voice processing on GStreamer 0.14Marvin W2021-05-023-19/+27
|
* 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 W2021-05-015-16/+356
|
* Handle non-existant call supportfiaxh2021-04-293-3/+20
|
* Video optimizationsMarvin W2021-04-298-116/+1014
|
* Handle broken VAPI in older valaMarvin W2021-04-111-2/+19
|
* Fix custom vapi integrationMarvin W2021-04-111-2/+0
|
* GStreamer compatMarvin W2021-04-113-10/+30
|
* Fix bug in legacy SRTP decryptionMarvin W2021-04-011-1/+6
|
* Remove unnecessary debug codeMarvin W2021-04-011-4/+0
|
* Migrate to libsrtp2Marvin W2021-03-292-30/+23
|
* 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
|
* Add initial support for DTLS-SRTPfiaxh2021-03-251-0/+1
|
* Move SRTP implementation into crypto library for reuseMarvin W2021-03-235-1034/+12
|
* Resample audio data for common 48k sample rateMarvin W2021-03-232-10/+16
|
* Add support for SRTPMarvin W2021-03-236-66/+1103
|
* RTP: Backport gst_caps_copy_nth from GStreamer 1.16Marvin W2021-03-211-2/+10
|
* Add RTP implementation as pluginMarvin W2021-03-219-0/+1748