112 commits

Author	SHA1	Message	Date
Alessandro Ros	823697210e	webrtc: fix race condition after #4558 (#4564)	2025-05-27 15:17:31 +02:00
Alessandro Ros	5d203b4d98	webrtc: prevent routine leaks (#4558) ... wait for all routines to exit before assuming a WebRTC connection is closed.	2025-05-26 11:33:46 +02:00
Alessandro Ros	fff3b9b315	record: fix loss of audio samples during segment switch (#4556)	2025-05-25 21:29:33 +02:00
Alessandro Ros	fad64a87bd	bump golangci-lint (#4550)	2025-05-23 15:10:21 +02:00
Alessandro Ros	1b9dfbd367	rtmp: support connecting to sources that require standard credentials (#4530)	2025-05-15 14:23:03 +02:00
Alessandro Ros	3b3b898358	fix error message (#4522)	2025-05-12 22:10:13 +02:00
Alessandro Ros	c17a6de2a6	parse HTTP username and password OR token, not both (#4517)	2025-05-11 10:20:57 +02:00
Alessandro Ros	f97213ae6e	support passing JWTs through the password field (#4516) ... This is safer than passing JWTs through query parameters, unfortunately support is limited.	2025-05-10 22:54:24 +02:00
Alessandro Ros	a85d8f1391	rtmp: fix connect command when reading (#4512) ... when reading, the "connect" command should not contain fpad, capabilities, audioCodecs, videoCodecs, videoFunction.	2025-05-10 15:26:48 +02:00
Alessandro Ros	15fff95246	ensure unmarshaled entities can be marshaled back (#4484)	2025-05-04 11:34:05 +02:00
Alessandro Ros	77a3c7ae6e	webrtc: route original absolute timestamp of packets (#1300) (#4415)	2025-04-12 11:34:27 +02:00
Alessandro Ros	49bcd35afd	bump gortsplib (#4416)	2025-04-12 11:29:37 +02:00
Alessandro Ros	8b98c02903	rtsp: route original absolute timestamp of packets (#1300) (#4378)	2025-03-29 19:08:42 +01:00
Alessandro Ros	4bbedc4955	hls: fix reading streams with non-standard clock rates (#4376)	2025-03-29 17:14:02 +01:00
Alessandro Ros	ee8a47844a	hls: fix race condition when getting absolute time (#4375)	2025-03-29 17:10:31 +01:00
Alessandro Ros	60cabeea92	hls: support routing absolute timestamps (#1300) (#4372)	2025-03-27 22:18:13 +01:00
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	416ac1357e	bump mediacommon and gortsplib (#4364)	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	c692f3b78c	webrtc: rewrite WHIP client (#4299)	2025-03-01 17:01:57 +01:00
Alessandro Ros	aa101c680c	webrtc: make client always provide UDP candidates (#4298)	2025-03-01 16:52:59 +01:00
Alessandro Ros	5c6cf58d75	webrtc: fix connecting to TCP-only sources (#4293)	2025-03-01 11:07:50 +01:00
Alessandro Ros	244da930a1	switch to mediacommon/v2 (#4259)	2025-02-17 14:54:58 +01:00
dependabot[bot]	b66efd66da	build(deps): bump github.com/bluenviron/gortsplib/v4 (#4236) ... Bumps [github.com/bluenviron/gortsplib/v4](https://github.com/bluenviron/gortsplib) from 4.12.2 to 4.12.3. - [Commits](https://github.com/bluenviron/gortsplib/compare/v4.12.2...v4.12.3) --- updated-dependencies: - dependency-name: github.com/bluenviron/gortsplib/v4 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>	2025-02-07 20:19:14 +01:00
Jean-Philippe Bergeron	e8297478f3	Configurable webrtcSTUNGatherTimeout (#4221)	2025-02-07 16:34:17 +01:00
Alessandro Ros	a1c6da84dc	webrtc: fix detecting closure of some sessions (#4204) (#4212)	2025-02-01 13:43:57 +01:00
dependabot[bot]	c6a9d81d4c	build(deps): bump github.com/bluenviron/mediacommon (#4193) ... 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
Alessandro Ros	5bb751a4c8	hls: fix reading AV1 from OBS+WHIP (#3886) (#4177)	2025-01-19 18:40:18 +01:00
Alessandro Ros	e86a7a8217	webrtc: disable UDP when not needed (#4176)	2025-01-19 15:43:58 +01:00
Alessandro Ros	8f6267deb8	bump pion/webrtc to v4 (#4145)	2025-01-13 23:19:29 +01:00
Alessandro Ros	070ea5fc19	rtmp: fix compatibility with DJI Osmo Action 4 (#3802) (#4116)	2025-01-07 17:36:31 +01:00
Alessandro Ros	6d7d0e6e21	fix documentation (#4110)	2025-01-05 12:02:04 +01:00
Alessandro Ros	d4c29f8283	webrtc: switch to recvonly transceivers (#4059) (#4108) ... 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) ... 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	770d72b638	rtmp: support Extended Metadata Frames (#4006) (#4014) (#4018)	2024-12-04 23:29:16 +01:00
Alessandro Ros	72a8b3ca8a	webrtc: support publishing and reading H265 tracks (#4003)	2024-12-02 23:55:54 +01:00
Andres Uribe	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	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) ... This improves timestamp precision.	2024-10-07 17:59:32 +02:00
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	4d0ce87f09	hls: support generating streams with multiple audio tracks (#2728) (#3793)	2024-10-03 19:38:54 +02:00
Alessandro Ros	1e30dcb555	webrtc: fix crash in case of congestion (#3813) (#3815)	2024-09-29 09:56:06 +02:00
Alessandro Ros	658848f8c8	log track ID when skipping tracks (#3798)	2024-09-26 14:42:48 +02:00
Alessandro Ros	783fede2cb	remove debug lines (#3771)	2024-09-15 22:52:55 +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
Alessandro Ros	e6653857aa	rtmp: support ingesting AV1, VP9, H265, MP3, PCM from other servers (#3751)	2024-09-09 12:26:35 +02:00
Alessandro Ros	c5059fa7a0	move RPI Camera component into dedicated repository (#3656)	2024-08-14 23:24:17 +02:00