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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
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* Improve call details dialog + small multi-party call fixesfiaxh2021-11-151-1/+1
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* Optimize encoder for low cpu usageMarvin W2021-11-151-5/+5
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* Add maximum bitrate and adjust video resolution based on bitrateMarvin W2021-11-154-12/+111
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* Log probe for decode QOSMarvin W2021-11-111-0/+40
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* Limit REMB target bitrate to 2x maximum actually seen valueMarvin W2021-11-111-32/+48
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* Display target bitrates in connection details UIfiaxh2021-11-111-9/+8
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* Fix REMB calculationMarvin W2021-11-101-2/+5
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* Make elements sync to get proper qos dataMarvin W2021-11-102-3/+3
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* RTP: Make opus mono-channelMarvin W2021-11-101-2/+2
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* RTP: Only start gstreamer pipeline once neededMarvin W2021-11-101-48/+66
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* RTP: Encode with deviceMarvin W2021-11-102-174/+413
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* Split payloader off encoder chainMarvin W2021-11-101-4/+39
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* Improve codec supportMarvin W2021-11-101-4/+7
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* Crop video to match widget ratioMarvin W2021-11-101-29/+45
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* Fix compiler warnings ('Switch does not handle .. of enum ..')fiaxh2021-10-121-0/+2
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* Fix misc compiler warningsfiaxh2021-10-121-1/+1
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* RTP: Handle missing rtp pay/depay elementsMarvin W2021-05-152-14/+16
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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 W2021-05-022-16/+23
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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 W2021-05-014-14/+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
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