Alessandro Ros and GitHub
0eb7089ed6
use safer atomic structs instead of atomic functions ( #5622 )
2026-03-31 11:30:50 +02:00
Alessandro Ros and GitHub
3381b196fb
solve codec labels internally ( #5603 )
...
stop using gortsplib format.Format.Codec() for getting codec names.
2026-03-21 18:22:49 +01:00
Alessandro Ros and GitHub
c8e07b55d5
api: add new stats ( #5582 )
...
- RTSPSession.outboundRTPPacketsDiscarded
- Path.inboundFramesInError
- SRTConn.outboundFramesDiscarded
- WebRTCSession.outboundFramesDiscarded
- RTMPConn.outboundFramesDiscarded
- HLSMuxer.outboundFramesDiscarded
2026-03-17 17:22:15 +01:00
Alessandro Ros and GitHub
adf45596a3
api: rename WebRTC stats to match RTSP ones ( #5581 )
2026-03-16 23:29:25 +01:00
Alessandro Ros and GitHub
dfce20fefa
api: add RTSP reported lost packets ( #5198 ) ( #5579 )
...
The new outboundRtpPacketsReportedLost property allows to track RTP
packets that have been reported lost by readers. Furthermore, stats now
have a "inbound" or "outbound" prefix to improve readability.
2026-03-16 22:10:18 +01:00
Alessandro Ros and GitHub
72832369f6
inherit MPEG-4 audio type from alwaysAvailableFile ( #5539 )
2026-03-01 10:12:35 +01:00
Alessandro Ros and GitHub
05ab631996
refactor alwaysAvailable tests ( #5537 )
2026-03-01 00:20:20 +01:00
Alessandro Ros and GitHub
2f91c8198c
fix audio from alwaysAvailableFile not being streamed ( #5535 )
2026-02-28 21:46:25 +01:00
Alessandro Ros and GitHub
10e0271755
fix corrupted video with alwaysAvailableFile ( #5534 )
...
when alwaysAvailableFile points to a file with a H265 or H264 track,
server is started, an online stream is published and then closed, video
was getting corrupted since the online video was overriding the
parameters of the offline video.
2026-02-28 21:06:07 +01:00
Alessandro Ros and GitHub
4ebfcde8e2
fix MPEG-4 audio configuration not matching error ( #5468 ) ( #5533 )
...
This happened when using alwaysAvailableFile and a MPEG-4 audio track.
2026-02-28 20:17:44 +01:00
Alessandro Ros and GitHub
5e80554d20
improve playback precision of alwaysAvailable offline segment ( #5530 )
2026-02-28 18:39:52 +01:00
Alessandro Ros and GitHub
549300cbd4
prevent using alwaysAvailableFile and alwaysAvailableTracks together ( #5529 )
2026-02-28 18:31:41 +01:00
Alessandro Ros and GitHub
4974cacb94
improve video/audio sync of alwaysAvailable ( #5443 ) ( #5508 )
...
store elapsed time once for the entire stream and start PTS of sub
streams from there.
2026-02-22 18:25:32 +01:00
2e429158a8
log packet sizes if UDPMaxPayloadSize is exceeded ( #4668 )
...
Co-authored-by: Alessandro Ros <aler9.dev@gmail.com >
2026-02-22 13:42:12 +01:00
bluenviron-bot and GitHub
a07526d57c
bump Golang to 1.26 ( #5460 )
2026-02-21 14:48:53 +01:00
Alessandro Ros and GitHub
336714157f
fix codec parameters not being loaded from alwaysAvailableFile ( #5484 )
2026-02-18 21:29:24 +01:00
Alessandro Ros and GitHub
ae21240c76
add additional checks on always-available streams ( #5408 )
...
make sure that published streams match MPEG-4 audio, G711 and LPCM
parameters declared in configuration or used by the offline file.
2026-02-06 16:06:24 +01:00
Alessandro Ros and GitHub
a56408db19
add always available streams ( #5335 )
...
When the publisher or source of a stream is offline, the server can be
configured to fill gaps in the stream with a video that is played on
repeat until a publisher comes back online. This allows readers to stay
connected regardless of the state of the stream. The offline video and
any future online stream are concatenated without decoding or
re-encoding packets, using the original codec.
2026-01-31 14:44:58 +01:00
Alessandro Ros and GitHub
02fe2e2b42
tests: fix race condition ( #5354 )
2026-01-17 23:21:34 +01:00
Alessandro Ros and GitHub
653d10fb75
use the same filtering process for every codec ( #5324 )
...
this allows to apply features that were previously implemented for
single codecs (like RTP packet resizing), to any codec, and simplifies
future development.
2026-01-16 14:03:13 +01:00
bluenviron-bot and GitHub
320bd3f4ae
bump golangci-lint to 2.8.0 ( #5350 )
2026-01-15 22:43:32 +01:00
Alessandro Ros and GitHub
eba3b828a6
print last processing error ( #5323 )
2026-01-06 21:48:36 +01:00
Alessandro Ros and GitHub
cd34de8770
print last decode error ( #5319 )
...
decode errors are still grouped together, but the last one is now
printed.
2026-01-05 15:28:24 +01:00
Alessandro Ros and GitHub
0cdae40fe3
estimate absolute timestamp more precisely ( #5078 )
...
When the absolute timestamp of incoming frames was not available, it
was filled with the current timestamp, which is influenced by latency
over time.
This mechanism is replaced by an algorithm that detects when latency is
the lowest, stores the current timestamp and uses it as reference
throughout the rest of the stream.
2025-10-12 11:02:14 +02:00
Alessandro Ros and GitHub
f5f03562d3
rewrite Unit definition ( #5079 )
...
Stream units now share the same struct, with a specialized payload.
2025-10-11 12:18:51 +02:00
Alessandro Ros and GitHub
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 and GitHub
e3b8af8933
switch to gortsplib/v5 ( #4978 )
2025-09-16 13:10:34 +02:00
Alessandro Ros and GitHub
57021467e9
rename formatprocessor into codecprocessor ( #4962 )
2025-09-12 10:24:11 +02:00
Alessandro Ros and GitHub
d423a71aaa
update linter settings ( #4790 )
2025-07-26 16:44:32 +02:00
Alessandro Ros and GitHub
9ddcbf5c97
recorder: limit maximum part size ( #4674 ) ( #4760 )
...
this prevents RAM exhaustion.
2025-07-20 19:16:33 +02:00
Alessandro Ros and GitHub
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 and GitHub
b564dc9ff1
rtsp: emit log message when remuxing RTP packets ( #4422 )
2025-04-13 10:21:46 +02:00
Alessandro Ros and GitHub
14eb9f6175
improve tests ( #4423 )
2025-04-13 10:00:52 +02:00
Alessandro Ros and GitHub
0d46cf3f74
rename stream decode errors into processing errors ( #4370 )
...
Stream errors include both errors from decoding RTP packets into
frames, and errors from encoding frames into RTP packets. "processing
errors" is more fit.
2025-03-26 15:14:08 +01:00
Alessandro Ros and GitHub
f851cb6961
add missing Start() call; fix wrong sprintf ( #4369 )
2025-03-26 12:47:09 +01:00
Alessandro Ros and GitHub
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 and GitHub
416ac1357e
bump mediacommon and gortsplib ( #4364 )
2025-03-24 17:25:18 +01:00
Alessandro Ros and GitHub
b329c4bbe8
replace New* with Initialize() ( #4345 )
2025-03-16 15:34:53 +01:00
Alessandro Ros and GitHub
5cb5dc4e12
remove useless checks ( #4213 )
2025-01-31 20:57:50 +01:00
Alessandro Ros and GitHub
57addb1cc2
api, metrics: add additional RTSP statistics ( #3312 ) ( #4073 )
2024-12-25 15:43:41 +01:00
Alessandro Ros and GitHub
23002d9f5f
route original timestamps without converting to durations ( #3839 )
...
This improves timestamp precision.
2024-10-07 17:59:32 +02:00
Alessandro Ros and GitHub
2586782031
fix race condition in tests ( #3834 )
2024-10-05 21:54:11 +02:00
Alessandro Ros and GitHub
4c3ac34425
fix memory leak in case of errors during initialization of a reader ( #3831 )
2024-10-05 00:49:44 +02:00
Alessandro Ros and GitHub
ad58efe47d
move RTMP tests into internal/servers/rtmp ( #3035 )
2024-02-16 23:27:27 +01:00
Alessandro Ros and GitHub
514036d41a
treat different RTSP formats as different tracks in logs and API ( #2907 )
2024-01-15 12:08:14 +01:00
4bf0d10079
metrics: add paths_bytes_sent, srt_conns, srt_conns_bytes_received, srt_conns_bytes_sent ( #2620 ) ( #2619 ) ( #2629 )
...
* add missing Prometheus exports (#2620 , #2619 ):
paths_bytes_sent, srt_conns, srt_conns_bytes_received, srt_conns_bytes_sent
* protect Stream.BytesSent()
* add tests
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2023-11-08 11:20:16 +01:00
Alessandro Ros and GitHub
95ab9375c7
support recording to MPEG-TS ( #2505 )
2023-10-14 22:52:10 +02:00
Alessandro Ros and GitHub
73ddb21e63
implement native recording ( #1399 ) ( #2255 )
...
* implement native recording (#1399 )
* support saving VP9 tracks
* support saving MPEG-1 audio tracks
* switch segment when codec parameters change
* allow to disable recording on a path basis
* allow disabling recording cleaner
* support recording MPEG-1/2/4 video tracks
* add microseconds to file names
* add tests
2023-09-16 17:27:07 +02:00
Alessandro Ros and GitHub
c4cb4200ff
split RTP packet handling from data handling ( #2337 )
2023-09-16 17:16:33 +02:00
Alessandro Ros and GitHub
ffa34428fc
fix bug that prevented multiple readers from accessing the same stream ( #2281 ) ( #2282 )
2023-09-01 23:15:27 +02:00
