github-actions[bot]
6250d6c485
bump hls.js to v1.6.0 ( #4374 )
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2025-03-28 08:48:10 +01:00
Alessandro Ros
f851cb6961
add missing Start() call; fix wrong sprintf ( #4369 )
2025-03-26 12:47:09 +01:00
Alessandro Ros
986e270862
count and log all discarded frames, decode errors, lost packets ( #4363 )
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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
416ac1357e
bump mediacommon and gortsplib ( #4364 )
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2025-03-24 17:25:18 +01:00
Alessandro Ros
b329c4bbe8
replace New* with Initialize() ( #4345 )
2025-03-16 15:34:53 +01:00
Alessandro Ros
371b9585a6
change error message when stream is not found ( #4301 )
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Switch from 'no one is publishing to path' to 'no stream is available
on path' since the stream might be provided from a static source too.
2025-03-03 18:09:22 +01:00
Alessandro Ros
c692f3b78c
webrtc: rewrite WHIP client ( #4299 )
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2025-03-01 17:01:57 +01:00
Alessandro Ros
5c6cf58d75
webrtc: fix connecting to TCP-only sources ( #4293 )
2025-03-01 11:07:50 +01:00
Alessandro Ros
386be42784
rtsp: rewrite authentication around ServerConn.VerifyCredentials ( #4267 )
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2025-02-18 17:54:13 +01:00
Alessandro Ros
d822c2a200
rtsp: support using relative URLs in sourceRedirect ( #4260 )
2025-02-18 15:23:07 +01:00
Alessandro Ros
244da930a1
switch to mediacommon/v2 ( #4259 )
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2025-02-17 14:54:58 +01:00
Jean-Philippe Bergeron
e8297478f3
Configurable webrtcSTUNGatherTimeout ( #4221 )
2025-02-07 16:34:17 +01:00
Alessandro Ros
1a1da5edda
fix freeze when getting metrics and a RTSP connection is being created ( #4179 ) ( #4216 )
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2025-02-02 14:03:17 +01:00
Alessandro Ros
a1c6da84dc
webrtc: fix detecting closure of some sessions ( #4204 ) ( #4212 )
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2025-02-01 13:43:57 +01:00
Alessandro Ros
5cb5dc4e12
remove useless checks ( #4213 )
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2025-01-31 20:57:50 +01:00
dependabot[bot]
c6a9d81d4c
build(deps): bump github.com/bluenviron/mediacommon ( #4193 )
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Bumps [github.com/bluenviron/mediacommon](https://github.com/bluenviron/mediacommon ) from 1.13.4-0.20250119173005-20b296ab1174 to 1.13.4.
- [Commits](https://github.com/bluenviron/mediacommon/commits/v1.13.4 )
---
updated-dependencies:
- dependency-name: github.com/bluenviron/mediacommon
dependency-type: direct:production
update-type: version-update:semver-patch
...
Signed-off-by: dependabot[bot] <support@github.com>
Co-authored-by: dependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
2025-01-24 11:04:08 +01:00
github-actions[bot]
f40f491b09
bump hls.js to v1.5.20 ( #4185 )
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2025-01-22 07:15:45 +01:00
Alessandro Ros
c1e1c74f6e
webrtc: fix setting video width, height and rate when publishing a camera with browser ( #4167 ) ( #4171 )
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2025-01-18 18:57:56 +01:00
Alessandro Ros
8f6267deb8
bump pion/webrtc to v4 ( #4145 )
2025-01-13 23:19:29 +01:00
Alessandro Ros
8f04264fe5
webrtxc: fix MTX_QUERY not set when reading or publishing ( #4138 ) ( #3937 ) ( #4141 )
2025-01-11 17:29:48 +01:00
github-actions[bot]
5077fb2238
bump hls.js to v1.5.19 ( #4140 )
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2025-01-11 10:25:29 +01:00
aler9
1342431c15
webrtc: fix reconnecting to server after a failure ( #4022 )
2025-01-06 13:51:08 +01:00
Alessandro Ros
d4c29f8283
webrtc: switch to recvonly transceivers ( #4059 ) ( #4108 )
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This fixes compatibility with devices that support decoding AV1 but
don't support encoding it.
This was previously impossible to achieve due to a bug that prevented
video from being displayed when recvonly transceivers were in use and
audio was not present.
2025-01-04 16:36:03 +01:00
Alessandro Ros
b49acb1e00
accept durations expressed as days (i.e. '1d') ( #4094 )
2025-01-02 12:44:15 +01:00
Alessandro Ros
8cbbbc05c3
rtmp: support additional Enhanced-RTMP features ( #3685 ) ( #4088 )
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new features:
* support publishing Opus and AC-3 tracks
* support publishing more than 2 tracks. This is compatible with OBS multitrack video and OBS VOD audio track
2025-01-02 12:44:01 +01:00
Alessandro Ros
6425bd5a8e
srt: fix compatibility with Blackmagic Design hardware ( #3701 ) ( #4087 )
2025-01-02 12:43:37 +01:00
Alessandro Ros
dfb792eb90
rtsp: rename encryption, protocols, serverKey, serverCert into rtspEncryption, rtspProtocols, rtspServerKey, rtspServerCert ( #4078 )
2024-12-25 18:26:14 +01:00
Alessandro Ros
8cb584b514
api, metrics: fix RTC lowercase in webRTCSession and webRTCSource ( #4076 )
2024-12-25 17:47:24 +01:00
Alessandro Ros
28970b323e
fix MTX_CONN_TYPE value with RTSP connections ( #3967 ) ( #4075 )
2024-12-25 17:42:19 +01:00
Alessandro Ros
8a808ac0f6
api: add 'session' field to RTSPConn ( #3974 ) ( #4074 )
2024-12-25 16:17:53 +01:00
Alessandro Ros
57addb1cc2
api, metrics: add additional RTSP statistics ( #3312 ) ( #4073 )
2024-12-25 15:43:41 +01:00
github-actions[bot]
d23fb08e2d
bump hls.js to v1.5.18 ( #4052 )
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Co-authored-by: mediamtx-bot <bot@mediamtx>
2024-12-18 11:22:53 +01:00
Alessandro Ros
72a8b3ca8a
webrtc: support publishing and reading H265 tracks ( #4003 )
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2024-12-02 23:55:54 +01:00
Alessandro Ros
576a117c7c
bump Golang to 1.23 ( #3702 )
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2024-11-04 14:02:05 +01:00
Andres Uribe
f8b366c604
webrtc: restart ICE only on failed connection states ( #3899 )
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* webrtc: Restart ICE only on failed connection states
* rename "connected" into "ready" since WebRTC can emit the "connected" state multiple times
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
2024-10-31 12:47:35 +01:00
Alessandro Ros
5ff4f90ff9
webrtc: fix using JavaScript classes from outside the server ( #3923 )
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2024-10-30 21:29:50 +01:00
Alessandro Ros
20ad88f013
webrtc: move JavaScript reader and publisher into portable classes ( #3921 )
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2024-10-28 18:48:11 +01:00
Alessandro Ros
51b21c8adb
webrtc: include query parameters in session URL ( #3920 )
2024-10-28 18:35:55 +01:00
github-actions[bot]
773116a647
bump hls.js to v1.5.17 ( #3885 )
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Co-authored-by: mediamtx-bot <bot@mediamtx>
2024-10-18 17:42:35 +02:00
github-actions[bot]
a09b27f421
bump hls.js to v1.5.16 ( #3853 )
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Co-authored-by: mediamtx-bot <bot@mediamtx>
2024-10-09 20:07:13 +02:00
Alessandro Ros
1a6547f13b
restore pprof service ( #3848 ) ( #3851 )
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2024-10-08 17:06:47 +02:00
Alessandro Ros
d13dc1074a
make better use of Gin middleware system ( #3849 )
2024-10-08 17:02:16 +02:00
Alessandro Ros
23002d9f5f
route original timestamps without converting to durations ( #3839 )
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This improves timestamp precision.
2024-10-07 17:59:32 +02:00
Alessandro Ros
ae3a0e4560
remove deprecated test ( #3836 )
2024-10-06 13:58:28 +02:00
Alessandro Ros
c71694741e
show right server version when server is compiled from source ( #3409 )
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2024-10-05 23:10:14 +02:00
Alessandro Ros
2586782031
fix race condition in tests ( #3834 )
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2024-10-05 21:54:11 +02:00
Alessandro Ros
534b637bc7
support using JWT in Authorization header with API, Metrics, PProf ( #3630 ) ( #3795 )
2024-10-05 21:15:21 +02:00
Alessandro Ros
4c3ac34425
fix memory leak in case of errors during initialization of a reader ( #3831 )
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2024-10-05 00:49:44 +02:00
Alessandro Ros
4d0ce87f09
hls: support generating streams with multiple audio tracks ( #2728 ) ( #3793 )
2024-10-03 19:38:54 +02:00
Alessandro Ros
337d3d04ff
hls: fix delay when window is minimized ( #3759 ) ( #3772 )
2024-09-15 22:53:04 +02:00
