mirrors/gst-plugins-rs
mirrors/gst-plugins-rs
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git synced 2025-02-24 16:46:25 +00:00
Code Issues Projects Releases Wiki Activity
3581 commits 14 branches 107 tags 58 MiB
Commit graph

905 commits

Author SHA1 Message Date
Tim-Philipp Müller
40d33805d8 webrtcsink: chain up in signaller ::constructed funcs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2094>
 2025-02-20 12:53:20 +00:00
eipachte
1b866222eb hlssink3: Write playlists atomically 
We want to try to ensure that playlist files are written completely in a
single shot, to avoid the possibility of serving up a patially written
playlist.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2098>
 2025-02-20 11:23:50 +00:00
Sebastian Dröge
ee330f13ae Fix some new clippy 1.84 warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2097>
 2025-02-19 19:45:04 +02:00
Yaakov Selkowitz
6f4a4c392f Add missing copies of license files 
This should fix the crates that are missing license files.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2052>
 2025-01-28 11:34:15 +00:00
Sebastian Dröge
b70ad7895f mpegtslivesrc: Handle zero-byte adaption fields 
Simply skip over them instead of handling them as parse error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2048>
 2025-01-21 13:50:46 +00:00
Tim-Philipp Müller
0b0c1cbf50 rtp: ac3: validate depayloaded AC-3 data in unit tests 
Check for valid frame header.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2046>
 2025-01-20 19:32:48 +00:00
Tim-Philipp Müller
6292916f94 rtp: tests: add run_test_pipeline variants with data validation 
So we can actually check the content of depayloaded buffers too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2046>
 2025-01-20 19:32:48 +00:00
Tim-Philipp Müller
82167453b4 rtpac3depay2: fix handling of non-fragmented payloads 
The frames of a non-fragmented payload would contain
an extra two bytes before the frame sync and then
missing two bytes at the end which which would cause
decoding errors on the last block and/or frame crc
check failures.

This happened because we didn't take into account
the 2-byte packet payload header when creating output
sub-buffers, as the offsets we were using were in
relation to the payload data after the headers.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/645

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2046>
 2025-01-20 19:32:47 +00:00
Sebastian Dröge
148608e1d5 rtp: basepay: Only forward buffers if we have a segment 
If there are pending buffers without a segment then they must come from
the caps only and should be forwarded at a later time, if any.

Also reject any incoming buffers if no segment was received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2042>
 2025-01-15 18:10:39 +00:00
Sebastian Dröge
49d6b9b1c2 rtp: basedepay: Only forward buffers if we have a segment 
If there are pending buffers without a segment then they must come from
the caps only and should be forwarded at a later time, if any.

Also reject any incoming buffers if no segment was received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2042>
 2025-01-15 18:10:39 +00:00
Sebastian Dröge
36cdb85e77 mpegtslivesrc: Parse PES packets and check for reasonable PTS/DTS 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2011>
 2024-12-24 11:30:23 +00:00
Sebastian Dröge
f610a0c9d8 mpegtslivesrc: Refactor section parser 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2011>
 2024-12-24 11:30:23 +00:00
Sebastian Dröge
ff547205ea mpegtslivesrc: Reset rate to 1/1 on disconts and flush observations 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2010>
 2024-12-23 19:59:06 +00:00
Sanchayan Maity
e871e5f74d aws: Add next-file support to putobjectsink 
Add `next-file` support to `awss3putobjectsink` on similar lines to
the `next-file` support in `multifilesink`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1928>
 2024-12-23 17:26:31 +00:00
Sebastian Dröge
feb1f45836 raptorqdec: Stop using deprecated API 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2007>
 2024-12-23 18:31:07 +02:00
Sebastian Dröge
230afc11ea Fix / silence various new Rust 1.83 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2007>
 2024-12-23 18:27:38 +02:00
Thibault Saunier
4764058efa webrtcsrc: Add a 'connect-to-first-producer' property 
This is an helper property which allows to avoid requiring to know
peer IDs, which is very useful during development.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/386
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2000>
 2024-12-20 10:51:25 +02:00
Mathieu Duponchelle
c2c8423e7f gstwebrtc-api: example: use http by default 
That way the webpage connects with ws:/ to the signaller.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/589
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1992>
 2024-12-16 15:14:41 +00:00
Mathieu Duponchelle
0ab583c3aa webrtc-api example: do not rely on webpack / npm proxying websocket 
Instead simply use the desired address directly from the reference
example, this makes it work out of the box without placing expectations
on the web server.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1991>
 2024-12-16 13:59:51 +00:00
Mathieu Duponchelle
9e848d3b75 webrtcsink: add nvh265enc support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1981>
 2024-12-11 13:13:42 +00:00
Taruntej Kanakamalla
6cf39d4a00 webrtc: use the nick to set enum type properties on openh264enc 
The properties `rate-control` and `complexity` are of enum types and passing
a gint value is resulting in a panic. So pass the corresponding nick of the enum
value instead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1971>
 2024-12-05 17:17:51 +00:00
Guillaume Desmottes
625eeccf25 webrtc: janus: handle slowlink event 
Fix this warning:

webrtc-janusvr-signaller imp.rs:426:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x7f317009b4d0> Unknown message from server: {
   "janus": "slowlink",
   "session_id": 980554280060589,
   "sender": 5867141593320621,
   "mid": "video0",
   "media": "video",
   "uplink": false,
   "lost": 15
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1967>
 2024-12-04 13:05:26 +00:00
Guillaume Desmottes
8434126a58 webrtc: janus: handle slow_link videoroom event 
Fix this warning:

webrtc-janusvr-signaller imp.rs:426:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x7f317009b4d0> Unknown message from server: {
   "janus": "event",
   "session_id": 980554280060589,
   "sender": 5867141593320621,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "slow_link",
         "current-bitrate": 0
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1967>
 2024-12-04 13:05:26 +00:00
Mathieu Duponchelle
f8178fbd21 webrtscink: fix deadlock when answering 
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/637
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1959>
 2024-12-02 16:49:18 +00:00
Ruben Gonzalez
0821c39640 webrtcsink: add openh264enc support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1957>
 2024-11-29 23:08:45 +00:00
Sebastian Dröge
3cccede612 webrtcsink: Set caps-change-mode=delayed on encoder capsfilter 
Otherwise when changing the target caps (e.g. for reducing quality)
there is a race condition between buffers between the converter elements
and renegotiation.

For example, videoconvertscale might've output a 1920x1080 buffer, then
the capsfilter is configured to 1280x720, the buffer arrives in
videorate, videorate notices that renegotiation is pending, tries to
renegotiate and ends up with EMPTY caps because it can only change the
framerate but not the resolution.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1951>
 2024-11-29 12:39:20 +00:00
Matthew Waters
0537734aee webrtcsink: don't block the tokio runtime while holding state lock in unprepare() 
It is possible that in unprepare(), waiting for a task to complete while
holding the state lock, that task may be waiting to acquire the state lock and
result in a deadlock.

This is quick to reproduce when starting and stopping webrtcsink in very quick
succession.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1932>
 2024-11-28 16:44:05 +11:00
Xavier Claessens
52f48d7279 webrtcsink: Ignore more fields in caps change 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1940>
 2024-11-27 12:18:33 +00:00
Sebastian Dröge
d032c52ff5 mpegtslivesrc: Fix mismatch between internal / external time usage 
Previously the internal time was stored as base offset for calculating
the external time from the PCR, which resulted in disconts being
detected wrongly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1934>
 2024-11-21 12:55:36 +00:00
Sebastian Dröge
5fe4d948bf mpegtslivesrc: Rename variables to make it clear which time domain they refer to 
We have the internal time domain (monotonic clock) and the external time
domain (scaled monotonic clock in the rate of the PCR).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1934>
 2024-11-21 12:55:36 +00:00
Sebastian Dröge
be2b4952d2 Update to async-tungstenite 0.28 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1905>
 2024-11-02 12:34:43 +02:00
Sebastian Dröge
1afe165fce Update async-tungstenite dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1905>
 2024-11-02 12:34:11 +02:00
Xavier Claessens
634ec19e24 janusvr_signaller: Do not block in end_session() 
Only stop() is allowed to block, wait there.

Fixes #603

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1903>
 2024-11-01 15:11:17 +00:00
Mathieu Duponchelle
4d75091e91 awstranscriber: post warning message with details when item is late 
When the latency is configured to a value that is too low, items will be
pushed out with an adjusted timestamp, thus affecting synchronization.

It can be useful for the application to receive details about those
adjustments.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1897>
 2024-10-29 11:36:07 +00:00
Sebastian Dröge
da634989d9 Update to quick-xml 0.37 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1893>
 2024-10-29 06:51:38 +00:00
Sebastian Dröge
d9664d77b0 mpegtslivesrc: Parse PAT/PMT and only handle PCRs from the first program 
This matches default behaviour of tsdemux and makes sure we're not
jumping between different PCRs if there are multiple.

At a later time, program selection could be implemented.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1894>
 2024-10-28 13:25:43 +00:00
Jerome Colle
05e3c4a5a3 rtpbasedepay2: fix reference timestamp meta duplicates 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1892>
 2024-10-28 12:05:54 +00:00
Sanchayan Maity
199995201c net/quinn: Specify crypto provider explicitly 
rustls allows the choice of ring or aws-lc-rs as the cryptographic
library implementation. This is enabled/selected via Cargo feature
flags. We have plugins directly or indirectly depending on rustls
like quinn, aws and spotify. In the presence of multiple plugins,
selecting different implementations as the default, rustls can
panic.

The safest way to avoid this is by using builder_with_provider
and selecting a provider explicitly.

See below issues for further discussion and clarifications.
https://github.com/rustls/rustls/issues/1877
https://github.com/seanmonstar/reqwest/pull/2225

While at it, also specify features explicitly for quinn and rustls.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1884>
 2024-10-28 11:10:03 +00:00
Sebastian Dröge
d10a7224e1 Update to quick-xml 0.36 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1885>
 2024-10-28 09:33:11 +02:00
Sebastian Dröge
07f7730632 aws: Allow a deprecated BehaviourVersion for now 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1873>
 2024-10-22 20:21:55 +00:00
Sebastian Dröge
347b5987bd Fix a couple of type hierarchy bugs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1873>
 2024-10-22 20:21:55 +00:00
Sebastian Dröge
6c0bfd3ffc webrtc: Silence two new Rust 1.82 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1874>
 2024-10-22 19:29:28 +00:00
Sanchayan Maity
9c640c8bc3 net/quinn: Fix panic due to unset default crypto provider 
Fix CI failure that we see after the upgrade of rustls from
0.23.13 to 0.23.15.

Related docs/PR
https://docs.rs/rustls/latest/rustls/crypto/struct.CryptoProvider.html#using-the-per-process-default-cryptoprovider
https://github.com/quinn-rs/quinn/pull/1882

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1866>
 2024-10-21 10:31:36 +00:00
Sebastian Dröge
e824ac31f4 reqwesthttpsrc: Allow a server error after a seek 
There might be a server error because a seek would immediately close the
old connection without allowing for clean shutdown.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/527

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1863>
 2024-10-18 17:48:42 +00:00
Mathieu Duponchelle
2790fb41b2 webrtcsink: fix session not in place errors 
The InPlace/Taken logic was introduced to avoid using an extra lock
around the session, but it places expectations that are not always
obvious to meet around when a session is expected to be taken or not.

Any code that expects to have access to the sessions at all times thus
needs either extra logic in the session wrapper, or to maintain the
state of the session outside of the session (eg mids).

This commit removes the logic, and wraps sessions in Arc<Mutex>>.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1859>
 2024-10-18 10:53:16 +00:00
Chris Bainbridge
f5b90ba261 custom-signaller: add missing manual-sdp-munging property 
All signallers must now implement this property

Fixes #611

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1858>
 2024-10-18 09:33:42 +00:00
Guillaume Desmottes
57234522ec webrtc: janus: add 'janus-state' property to the sink 
This property can be used by applications to track the state of the
signaller, especially to know when the stream is up.

Fix #510

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1849>
 2024-10-10 16:56:25 -04:00
Guillaume Desmottes
6cc9945d4e webrtc: janus: fix typo in doc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1849>
 2024-10-10 16:43:23 -04:00
Mathieu Duponchelle
5958e342c7 webrtcsink: fix naming of error dot files for discovery pipelines 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1844>
 2024-10-03 16:45:42 +01:00
Guillaume Desmottes
87697609a2 webrtc: allow PAR change in webrtcsink input caps 
We are already allowing resolution changes which can lead to change in
pixel-aspect-ratio.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1833>
 2024-10-01 17:42:21 +01:00