| Age | Commit message (Collapse) | Author | |
|---|---|---|---|
| 2022-02-12 | Calls: Fix device selector for multi-party calls, allow picking device ↵ | Marvin W | |
| before call started | |||
| 2022-02-11 | RTP: Make codec and hardware support compile-time configurable | Marvin W | |
| 2022-02-08 | Calls: Device picker | Marvin W | |
| 2021-12-19 | RTP: Improve screen resolution update logic | Marvin W | |
| 2021-12-18 | RTP: Less log spam | Marvin W | |
| 2021-12-18 | RTP: Correctly handle timestamp after re-enabling a stream | Marvin W | |
| 2021-11-15 | Add maximum bitrate and adjust video resolution based on bitrate | Marvin W | |
| 2021-11-10 | RTP: Encode with device | Marvin W | |
| 2021-05-01 | Fix webcam framerate selection | Marvin W | |
| 2021-05-01 | Correctly handle missing webrtc-audio-processing | Marvin W | |
| 2021-05-01 | Echo Cancellation | Marvin W | |
| 2021-04-29 | Video optimizations | Marvin W | |
| 2021-04-11 | GStreamer compat | Marvin W | |
| 2021-03-23 | Resample audio data for common 48k sample rate | Marvin W | |
| 2021-03-21 | RTP: Backport gst_caps_copy_nth from GStreamer 1.16 | Marvin W | |
| 2021-03-21 | Add RTP implementation as plugin | Marvin W | |
