fiaxh
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7d2e647690
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Improve call wording, cleanup
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2021-05-01 21:51:24 +02:00 |
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Marvin W
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0409f55426
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Fix webcam framerate selection
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2021-05-01 17:27:55 +02:00 |
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Marvin W
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d388525fc6
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Correctly handle missing webrtc-audio-processing
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2021-05-01 16:00:37 +02:00 |
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Marvin W
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23ffd37dde
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Echo Cancellation
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2021-05-01 15:48:51 +02:00 |
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fiaxh
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5d85b6cdb0
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Handle non-existant call support
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2021-04-29 16:13:25 +02:00 |
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Marvin W
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3880628de4
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Video optimizations
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2021-04-29 15:53:59 +02:00 |
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Marvin W
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fe160d94ba
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Handle broken VAPI in older vala
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2021-04-11 16:28:59 +02:00 |
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Marvin W
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6ebdec1d78
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GStreamer compat
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2021-04-11 12:31:03 +02:00 |
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Marvin W
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c5ab4fed87
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Fix bug in legacy SRTP decryption
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2021-04-01 11:51:35 +02:00 |
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Marvin W
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c5cb43350a
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Remove unnecessary debug code
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2021-04-01 11:51:12 +02:00 |
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Marvin W
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5e58f29883
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Migrate to libsrtp2
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2021-03-29 13:20:12 +02:00 |
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Marvin W
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9520a81b81
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Don't reuse PTs for different media types
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2021-03-29 13:14:37 +02:00 |
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Marvin W
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fc3263d49e
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Fix device manager usage for GStreamer 1.16
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2021-03-26 15:18:04 +01:00 |
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Marvin W
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4b230808b9
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Move SRTP implementation into crypto library for reuse
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2021-03-23 20:04:28 +01:00 |
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Marvin W
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b01f6f9ef7
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Resample audio data for common 48k sample rate
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2021-03-23 15:11:00 +01:00 |
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Marvin W
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b393d41601
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Add support for SRTP
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2021-03-23 15:11:00 +01:00 |
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Marvin W
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cde1e38f5d
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RTP: Backport gst_caps_copy_nth from GStreamer 1.16
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2021-03-21 15:43:54 +01:00 |
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Marvin W
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ef2e3c774c
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Add RTP implementation as plugin
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2021-03-21 12:41:38 +01:00 |
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