12 Commits

Author SHA1 Message Date
Alessandro Ros
2024783eec webrtc: fix clippy audio when reading Opus (#3878) (#5047)
Opus timestamp is now recomputed from scratch.
2025-09-30 10:35:56 +02:00
Alessandro Ros
cd80814009 do not include recorder and HLS muxer in sent bytes (#4380) (#5039)
in API (/paths/list, /paths/get) and metrics (paths_bytes_sent), the
amount of sent bytes was increased even in case of writes to the
recorder and HLS muxer, which are not generating network traffic. This
fixes the issue.
2025-09-29 09:28:04 +02:00
Alessandro Ros
e3b8af8933 switch to gortsplib/v5 (#4978) 2025-09-16 13:10:34 +02:00
Alessandro Ros
1083eea307 make RTP packet size compatible with RTSP/SRTP (#4692)
when RTSP encryption is enabled, maximum RTP packet size is slightly
decreased to make room for SRTP.
2025-07-05 15:42:58 +02:00
Alessandro Ros
986e270862 count and log all discarded frames, decode errors, lost packets (#4363)
Discarded frames, decode errors and lost packets were logged
individually, then there was a mechanism that prevented more than 1 log
entry per second from being printed, resulting in inaccurate reports.

Now discarded frames, decode errors and lost packets are accurately
counted, and their count is printed once every second.
2025-03-25 21:59:58 +01:00
Alessandro Ros
b329c4bbe8 replace New* with Initialize() (#4345) 2025-03-16 15:34:53 +01:00
Alessandro Ros
72a8b3ca8a webrtc: support publishing and reading H265 tracks (#4003) 2024-12-02 23:55:54 +01:00
Alessandro Ros
2586782031 fix race condition in tests (#3834) 2024-10-05 21:54:11 +02:00
Alessandro Ros
4c3ac34425 fix memory leak in case of errors during initialization of a reader (#3831) 2024-10-05 00:49:44 +02:00
Alessandro Ros
658848f8c8 log track ID when skipping tracks (#3798) 2024-09-26 14:42:48 +02:00
Alessandro Ros
471019f606 warn users about skipped tracks when reading or publishing (#3753) 2024-09-15 19:28:05 +02:00
Alessandro Ros
6a38c87a5b hls, webrtc: add FromStream / ToStream (#3752) 2024-09-09 12:59:23 +02:00