Alessandro Ros and GitHub
b49acb1e00
accept durations expressed as days (i.e. '1d') ( #4094 )
2025-01-02 12:44:15 +01:00
Alessandro Ros and GitHub
8cbbbc05c3
rtmp: support additional Enhanced-RTMP features ( #3685 ) ( #4088 )
...
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 and GitHub
770d72b638
rtmp: support Extended Metadata Frames ( #4006 ) ( #4014 ) ( #4018 )
2024-12-04 23:29:16 +01:00
Alessandro Ros and GitHub
72a8b3ca8a
webrtc: support publishing and reading H265 tracks ( #4003 )
2024-12-02 23:55:54 +01:00
f8b366c604
webrtc: restart ICE only on failed connection states ( #3899 )
...
* 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 and GitHub
d13dc1074a
make better use of Gin middleware system ( #3849 )
2024-10-08 17:02:16 +02:00
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
4d0ce87f09
hls: support generating streams with multiple audio tracks ( #2728 ) ( #3793 )
2024-10-03 19:38:54 +02:00
Alessandro Ros and GitHub
1e30dcb555
webrtc: fix crash in case of congestion ( #3813 ) ( #3815 )
2024-09-29 09:56:06 +02:00
Alessandro Ros and GitHub
658848f8c8
log track ID when skipping tracks ( #3798 )
2024-09-26 14:42:48 +02:00
Alessandro Ros and GitHub
783fede2cb
remove debug lines ( #3771 )
2024-09-15 22:52:55 +02:00
Alessandro Ros and GitHub
471019f606
warn users about skipped tracks when reading or publishing ( #3753 )
2024-09-15 19:28:05 +02:00
Alessandro Ros and GitHub
6a38c87a5b
hls, webrtc: add FromStream / ToStream ( #3752 )
2024-09-09 12:59:23 +02:00
Alessandro Ros and GitHub
e6653857aa
rtmp: support ingesting AV1, VP9, H265, MP3, PCM from other servers ( #3751 )
2024-09-09 12:26:35 +02:00
Alessandro Ros and GitHub
c5059fa7a0
move RPI Camera component into dedicated repository ( #3656 )
2024-08-14 23:24:17 +02:00
Alessandro Ros and GitHub
1e7a7d1674
rpi camera: update font and link it to IBM repository ( #3647 )
2024-08-11 23:34:12 +02:00
1055be99c0
automatically reload TLS certificates when they change ( #3598 )
...
* Dynamically refresh tls certs for all servers
* make sure that CertLoader is always closed
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2024-08-04 14:12:08 +02:00
Alessandro Ros and GitHub
c9a938a501
improve fuzz tests ( #3596 )
2024-08-01 17:01:56 +02:00
Alessandro Ros and GitHub
0230aa0a9c
record, hls: fix panic with MPEG-4 audio tracks without config ( #3590 ) ( #3594 )
2024-08-01 16:34:40 +02:00
Alessandro Ros and GitHub
afec0cd86c
rtmp: fix publishing from DJI FlightHub Sync ( #3301 ) ( #3504 )
2024-07-01 16:03:12 +02:00
Alessandro Ros and GitHub
a1dc9f45f5
webrtc: support publishing H265 tracks ( #3435 ) ( #3492 )
...
IMPORTANT NOTE: this doesn't allow to read H265 tracks with WebRTC,
just to publish them. The inability to read H265 tracks with WebRTC is
not in any way related to the server but depends on browsers and on the
fact that they are not legally entitled to embed a H265 decoder inside
them.
2024-06-19 21:02:08 +02:00
Alessandro Ros and GitHub
80a133afc9
bump dependencies ( #3457 )
2024-06-11 22:43:27 +02:00
3a2594d610
rtmp: fix error when publishing AAC audio tracks ( #3414 )
...
* fix single aac file rtmp publish error.
* add tests
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2024-06-11 13:26:45 +02:00
Alessandro Ros and GitHub
427249877c
webrtc: fix error "Failed to setup RTCP mux" on some readers ( #3381 ) ( #3449 )
2024-06-10 15:43:52 +02:00
Alessandro Ros and GitHub
095921dc26
webrtc: on browsers, display error messages from server ( #3448 )
2024-06-10 15:41:05 +02:00
Alessandro Ros and GitHub
5fe2819546
webrtc: set fmtp of outgoing VP9 and multiopus tracks ( #3446 )
2024-06-10 09:54:08 +02:00
Alessandro Ros and GitHub
511b276b4d
webrtc: support reading G711 16khz tracks ( #2848 ) ( #3445 )
2024-06-10 00:57:26 +02:00
Alessandro Ros and GitHub
44953c8e05
webrtc: fix supported AV1 profiles ( #3442 )
2024-06-09 23:09:55 +02:00
Alessandro Ros and GitHub
d7bc304e52
webrtc: speed up gathering of incoming tracks ( #3441 )
2024-06-09 22:58:40 +02:00
Alessandro Ros and GitHub
eaf47e6598
webrtc: support reading, publishing, proxying LPCM tracks ( #3437 )
2024-06-09 22:51:16 +02:00
Alessandro Ros and GitHub
16d0bb7200
webrtc: fix proxying PCMU tracks ( #3427 )
2024-06-05 12:46:55 +02:00
Alessandro Ros and GitHub
ca1638976b
webrtc: support reading and publishing multichannel Opus ( #3371 ) ( #3375 )
2024-06-03 00:26:32 +02:00
Alessandro Ros and GitHub
ca6e1259fb
webrtc: support reading and proxying stereo PCMU/PCMA tracks ( #3402 )
2024-06-02 23:08:54 +02:00
Jason Walton and GitHub
1c2f95f609
webrtc: allow configuring timeouts ( #3404 ) ( #3406 )
...
* webrtc: allow configuring timeouts (#3404 )
* fix from code inspect
2024-05-30 13:36:58 +02:00
Alessandro Ros and GitHub
407702380a
webrtc: in answer, include codecs that are actually in use ( #3374 )
2024-05-19 19:41:42 +02:00
Alessandro Ros and GitHub
e283725cde
support routing multichannel Opus between RTSP, SRT, HLS, UDP and recording in MPEG-TS and fMP4 ( #3355 ) ( #3368 )
2024-05-19 14:38:57 +02:00
Alessandro Ros and GitHub
5d0ebdef56
webrtc: fix error when charset is in Content-Type ( #3126 ) ( #3318 )
2024-05-05 10:47:12 +02:00
xjr20042 and GitHub
8b51ca9c8b
rtmp: fix error when reading metadata ( #3302 )
...
some rtmp client will send set chunk size after publish stream
2024-04-29 21:49:38 +02:00
Alessandro Ros and GitHub
1204f76eb7
update golangci-lint ( #3300 )
2024-04-28 18:20:08 +02:00
Alessandro Ros and GitHub
85dd81698b
support HTTPS, Allow-Origin and trusted proxies in API, playback server, metrics server and pprof server ( #2658 ) ( #2491 ) ( #3235 ) ( #3280 )
2024-04-21 17:10:35 +02:00
Alessandro Ros and GitHub
bfa83fcd33
webrtc: support WHIP/WHEP endpoints without PATCH support ( #3273 ) ( #3277 )
2024-04-20 23:55:35 +02:00
FuseTim and GitHub
4ed1a39da2
webrtc: support charset in Content-Type of WHIP offer responses ( #3274 )
2024-04-20 23:24:19 +02:00
2bd8ac7e19
fix support for JWT authentication in API, metrics, playback, pprof ( #3253 )
...
Co-authored-by: Rafael Scheidt <rafaelscheidt@Rafaels-MacBook-Air.local >
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2024-04-18 23:55:48 +02:00
Alessandro Ros and GitHub
665e11a376
webrtc, hls: use absolute paths in Location header ( #3195 ) ( #3177 ) ( #3240 )
2024-04-14 12:04:51 +02:00
Alessandro Ros and GitHub
a18bebfa58
webrtc: unexport WHIP primitives ( #3233 )
2024-04-13 11:08:30 +02:00
Alessandro Ros and GitHub
4354535a94
webrtc: fix crash when marshaling ICE fragments ( #3231 )
2024-04-13 10:47:26 +02:00
8aa26d6fb3
rtmp: fix streaming errors by handling AMF0 strict arrays ( #3188 ) ( #3189 )
...
* add amf0 strict array decode
* add amf0 long string decode
* support amf0 types undefined, unsupported, xmldocument and date
* fix error and add tests
* modify tests to follow original way to test
* do not skip markerUnsupported / markerUndefined
* implement StrictArray marshaling; remove handling of long strings, XML documents, date
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2024-04-12 16:13:57 +02:00
Alessandro Ros and GitHub
a50642c669
improve record tests ( #3216 )
2024-04-09 09:36:38 +02:00
