aboutsummaryrefslogtreecommitdiff
path: root/plugins/rtp/src
Commit message (Collapse)AuthorAgeFilesLines
* SRTP: Do not continue processing data after encrypt/decrypt failedMarvin W2021-12-181-37/+82
| | | | RTP: Copy less
* RTP: Correctly handle timestamp after re-enabling a streamMarvin W2021-12-182-3/+14
|
* Improve call details dialog + small multi-party call fixesfiaxh2021-11-151-1/+1
|
* 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
|
* Calls: Use vp8depay.wait-for-keyframe only with GStreamer 1.16+Marvin W2021-05-111-1/+3
|
* Support voice processing on GStreamer 0.14Marvin W2021-05-022-16/+23
|
* 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 W2021-05-014-14/+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 v1.2.3-70-g09d2 (git 2.47.0) at 2024-11-14 21:21:47 +0000