e6fec31516
bump hls-js to v1.4.14 ( #2825 )
2 years ago
fe4f436bd3
rtmp: change default RTMPS port to 1936 when it is omitted in source ( #2821 )
2 years ago
4e8b8a2014
rtmp, srt: fix output of 'is reading from' log entry ( #2820 )
2 years ago
94953f5d22
webrtc: support Chrome versions older than M72 ( #2621 ) ( #2814 )
2 years ago
89560c19a0
rtmp: fix parsing error caused by extended timestamps ( #2393 ) ( #2556 ) ( #2384 ) ( #1550 ) ( #2564 ) ( #2808 )
2 years ago
0c131a2e92
rpi: pass log level to libcamera ( #2617 ) ( #2811 )
2 years ago
22414bb38c
rtmp: return error in case of extended chunk stream IDs ( #2806 )
2 years ago
d261bfe773
support static sources in paths with regular expressions ( #824 ) ( #2799 )
...
This allows to proxy requests to other servers by using regular
expressions.
2 years ago
15e6c199b4
always use 'all_others' or 'all' after every other path configuration ( #2798 )
2 years ago
11988249df
move servers into internal/servers ( #2792 )
2 years ago
3891e7eb26
bump hls-js to v1.4.13 ( #2791 )
...
Co-authored-by: mediamtx-bot <bot@mediamtx>
2 years ago
b8dd7b9e52
add RTMPS metrics ( #2783 )
2 years ago
ed72fa7db1
fix crash when requesting metrics and RTMP, SRT or WebRTC servers are not present anymore ( #2782 )
2 years ago
20123fd5b9
record: rename segmentPath into path ( #2777 )
2 years ago
7c8e593b0b
record: normalize decoding and encoding of segment paths ( #2775 )
2 years ago
5da2ded815
record: rename recFormat into format ( #2776 )
2 years ago
47bd7352f0
support unix timestamp in filenames of recorded segments ( #2774 )
...
* added support for %s to use unix timestamp in filenames
* fix lint errors
* add tests
* update documentation
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
2 years ago
3e12f83732
hls, webrtc: fix appending slash to paths that contain slashes ( #2773 )
2 years ago
aade2eedb9
hls, webrtc: prevent XSS attack when appending slash to paths ( #2766 ) ( #2767 ) ( #2772 )
2 years ago
4ccb245feb
webrtc: fix propagating query parameters when appending slash to paths ( #2771 )
2 years ago
50e7b038c4
hls: fix appending slash when behind a reverse proxy ( #2770 )
2 years ago
0e5e3324a7
rpicamera: allow to change bitrate and IDR period dynamically ( #2703 ) ( #2764 )
2 years ago
ce45498769
move hooks into dedicated package ( #2746 )
2 years ago
81c803ba64
webrtc: fix bad quality when converting Opus from WebRTC to LL-HLS ( #2499 ) ( #2740 )
2 years ago
dc453620fe
record: fix invalid segment duration in case of offsets between video/audio ( #2477 ) ( #2715 )
2 years ago
f335fc67f9
test: use `t.Setenv` to set env vars in tests ( #2722 )
...
This commit replaces `os.Setenv` with `t.Setenv` in tests. The
environment variable is automatically restored to its original value
when the test and all its subtests complete.
Reference: https://pkg.go.dev/testing#T.Setenv
Signed-off-by: Eng Zer Jun <engzerjun@gmail.com>
2 years ago
c3d9be322d
allow using 'fallback' with static sources ( #2606 ) ( #2706 )
2 years ago
4fa6c16c85
Add query as optional block for SRT publish and read ( #2656 )
2 years ago
3277765851
apidocs: add missing recording settings to PathConf ( #2705 )
2 years ago
a760a98bcc
webrtc: fix crash when reading from path 'publish' ( #2688 ) ( #2692 )
2 years ago
7a1e155531
update dependencies ( #2687 )
2 years ago
687d8685ef
webrtc: improve connectivity mechanism ( #2686 )
2 years ago
83a3086de7
webrtc: prevent pion/webrtc from writing log lines ( #2681 )
2 years ago
2fe31a2421
record: allow using special characters in recording path ( #2674 )
2 years ago
d51baa04f7
record: fix deleting segments in case of relative paths ( #2526 ) ( #2673 )
2 years ago
ef19552632
fix setting 'recordDeleteAfter' to zero ( #2670 ) ( #2671 )
2 years ago
cb04d28f6e
metrics: expose state of WebRTC sessions ( #2662 )
2 years ago
f82c6e3654
tests: fix race condition in TestPathRunOnDemand ( #2663 )
2 years ago
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>
2 years ago
813611057d
add runOnUnDemand hook ( #2645 )
2 years ago
1d1d64cb89
rename sourceProtocol into rtspTransport, sourceAnyPortEnable into rtspAnyPort ( #2644 )
2 years ago
faf7218fda
webrtc: restore compatibility when running behind a proxy, in a subpath ( #2638 )
2 years ago
4f876ed207
fix crash when publishing to a path with 'runOnDemand' from outside 'runOnDemand' ( #2636 ) ( #2637 )
2 years ago
d41e4025de
hls: format code ( #2634 )
2 years ago
39a239caba
hls: bugfix, stability and resilience improvements in the webpage
...
- STABILITY: await MEDIA_ATTACHED event before performing "loadSource" on HLS
- STABILITY: await MANIFEST_PARSED event before performing "play" on video element
- RESILIENCE: on "MEDIA_ERROR" event perform "recoverMediaError" on HLS
2 years ago
15f1c73314
hls: apply readTimeout to HLS sources ( #2627 )
2 years ago
68b862c7f6
webrtc: allow using sourceFingerprint with WebRTC sources ( #2618 ) ( #2626 )
2 years ago
107a9a4281
webrtc: add options (width, height, framerate, audio capture) and store parameters in URL ( #1837 ) ( #2167 ) ( #2584 )
2 years ago
43d41c070b
move static sources into dedicated package ( #2616 )
2 years ago
e9528c0917
api: return empty list when page is empty ( #2604 ) ( #2614 )
2 years ago
github-actions[bot]
Alessandro Ros
Neeraj Kanhere
Eng Zer Jun
Bohdan Cherniakh
Dr. Ralf S. Engelschall
Lukinoh