Alessandro Ros and GitHub
a4ad182e4e
format imports ( #6041 )
2026-08-04 22:13:15 +00:00
Alessandro Ros and GitHub
4bb896afe0
webrtc: reset recomputed audio PTS if it drifts too much ( #6021 )
2026-08-01 16:24:53 +00:00
d843a0bf21
webrtc: fix packet corruption when reading G722 ( #6000 )
...
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2026-07-29 13:31:02 +02:00
c11810c2fc
build(deps): bump github.com/bluenviron/gortsplib/v5 from 5.5.3 to 5.5.4 ( #5822 )
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* build(deps): bump github.com/bluenviron/gortsplib/v5 from 5.5.3 to 5.5.4
Bumps [github.com/bluenviron/gortsplib/v5](https://github.com/bluenviron/gortsplib ) from 5.5.3 to 5.5.4.
- [Commits](https://github.com/bluenviron/gortsplib/compare/v5.5.3...v5.5.4 )
---
updated-dependencies:
- dependency-name: github.com/bluenviron/gortsplib/v5
dependency-version: 5.5.4
dependency-type: direct:production
update-type: version-update:semver-patch
...
Signed-off-by: dependabot[bot] <support@github.com >
* additional changes
---------
Signed-off-by: dependabot[bot] <support@github.com >
Co-authored-by: dependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2026-06-02 21:50:15 +02:00
Alessandro Ros and GitHub
7ab03cdc58
webrtc: rename incoming/outgoing into inbound/outbound ( #5816 )
2026-06-02 09:15:49 +02:00
Alessandro Ros and GitHub
0f5a76e253
modernize code ( #5814 )
2026-05-31 20:33:48 +02:00
8568d8c57c
webrtc: fix random absolute timestamps with Opus, G711 and LPCM ( #5597 )
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When rewriting audio RTP timestamps in WebRTC egress, NTP was
derived using regenerated packet timestamps minus the incoming
RTP base timestamp.
That mixed timestamp domains and could shift absolute time by an
arbitrary offset while still exposing mapping as available.
Fix by using a consistent outgoing RTP domain in rewritten audio
paths:
- snapshot outgoing base timestamp before rewriting each unit
- compute NTP from (outgoing packet timestamp - outgoing base
timestamp)
This keeps RTP<->NTP mapping coherent for sender reports and prevents
random absolute-time offsets in WebRTC loopback with
useAbsoluteTimestamp.
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2026-03-21 19:46:39 +01:00
Alessandro Ros and GitHub
3381b196fb
solve codec labels internally ( #5603 )
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stop using gortsplib format.Format.Codec() for getting codec names.
2026-03-21 18:22:49 +01:00
bluenviron-bot and GitHub
a07526d57c
bump Golang to 1.26 ( #5460 )
2026-02-21 14:48:53 +01:00
8901ac4e9c
webrtc: support reading KLV tracks ( #4722 )
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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2026-01-16 23:40:21 +01:00
62effa79ef
build(deps): bump github.com/bluenviron/gortmplib from 0.1.2 to 0.2.0 ( #5300 )
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Bumps [github.com/bluenviron/gortmplib](https://github.com/bluenviron/gortmplib ) from 0.1.2 to 0.2.0.
- [Commits](https://github.com/bluenviron/gortmplib/compare/v0.1.2...v0.2.0 )
---
updated-dependencies:
- dependency-name: github.com/bluenviron/gortmplib
dependency-version: 0.2.0
dependency-type: direct:production
update-type: version-update:semver-minor
...
Signed-off-by: dependabot[bot] <support@github.com >
Co-authored-by: dependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
2025-12-28 17:29:27 +01:00
Alessandro Ros and GitHub
9e82b5233d
fix 1-minute offset when recording a H264 WebRTC track ( #5275 ) ( #5278 )
2025-12-19 13:32:14 +01:00
Alessandro Ros and GitHub
d02fb4ed8e
move constant ( #5190 )
2025-11-12 18:09:53 +01:00
Alessandro Ros and GitHub
f5f03562d3
rewrite Unit definition ( #5079 )
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Stream units now share the same struct, with a specialized payload.
2025-10-11 12:18:51 +02:00
Alessandro Ros and GitHub
691d7290fb
webrtc: fix packet corruption when reading G711 ( #5048 )
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when a reader is reading a G711 track, stream units are modified
globally, affecting every other reader. This fixes the issue.
2025-09-30 10:42:24 +02:00
Alessandro Ros and GitHub
2024783eec
webrtc: fix clippy audio when reading Opus ( #3878 ) ( #5047 )
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Opus timestamp is now recomputed from scratch.
2025-09-30 10:35:56 +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
e0f4748839
modernize code ( #4947 )
2025-09-07 16:08:47 +02:00
Alessandro Ros and GitHub
d423a71aaa
update linter settings ( #4790 )
2025-07-26 16:44:32 +02:00
Alessandro Ros and GitHub
77a3c7ae6e
webrtc: route original absolute timestamp of packets ( #1300 ) ( #4415 )
2025-04-12 11:34:27 +02:00
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
244da930a1
switch to mediacommon/v2 ( #4259 )
2025-02-17 14:54:58 +01:00
Alessandro Ros and GitHub
8f6267deb8
bump pion/webrtc to v4 ( #4145 )
2025-01-13 23:19:29 +01:00
Alessandro Ros and GitHub
72a8b3ca8a
webrtc: support publishing and reading H265 tracks ( #4003 )
2024-12-02 23:55:54 +01:00
Alessandro Ros and GitHub
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 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
658848f8c8
log track ID when skipping tracks ( #3798 )
2024-09-26 14:42:48 +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
