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Author SHA1 Message Date
Jordan Petridis
f30cb2b56c video/gtk4: Add --features to the flatpak example 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1608>
 2024-06-07 10:54:05 +00:00
Mathieu Duponchelle
7cec628c43 transcriberbin: make sure to always record pad property changes 
When the pad isn't parented yet we should still record user choices,
either in our settings or in our state.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1593>
 2024-06-06 15:42:21 +00:00
Mathieu Duponchelle
0e85973e94 transcriberbin: fix regression with > 1 translation languages 
By making sure to expose uniquely named pads on the inner transcription
bins.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/552
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1593>
 2024-06-06 15:42:21 +00:00
Sebastian Dröge
30252a1b2e ndi: Add support for loading NDI SDK v6 
The library name and environment variable name have changed but the ABI
is completely compatible.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1607>
 2024-06-06 14:51:09 +00:00
Matthew Waters
1e964233c6 ci: run tests with RUST_BACKTRACE=1 
Produces backtraces which would allow some initial debugging on hard to
find issues.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1606>
 2024-06-06 14:02:55 +00:00
Angelo Verlain
c9ac553cfe gtk4: update flatpak integration code 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1600>
 2024-06-06 13:08:19 +00:00
Matthew Waters
260b04a1cf rtpbin2: protoct against adding with overflow 
If jitter is really bad, then this calculation may overflow.  Protect
against that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1605>
 2024-06-06 11:43:26 +00:00
Sebastian Dröge
ba70bb1154 deny: Add override for older tungstenite 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
 2024-06-06 10:34:12 +00:00
Sebastian Dröge
85c38107cf webrtc: Update to async-tungstenite 0.26 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
 2024-06-06 10:34:12 +00:00
Sanchayan Maity
8171a00943 net/quinn: Fix pad template naming typo 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1601>
 2024-06-05 13:44:40 +05:30
Tim-Philipp Müller
ab2f5e3d8d rtp: ac3: add some unit tests 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
 2024-06-01 12:43:27 +00:00
Tim-Philipp Müller
2b68920f82 rtp: tests: add possibility to make input live 
.. for payloaders that behave differently with live
and non-live inputs (e.g. audio payloaders which by
default will pick different aggregation modes based
on that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
 2024-06-01 12:43:27 +00:00
Tim-Philipp Müller
6597ec84eb rtp: tests: add possibility to check duration of depayloaded buffers 
.. and clarify an expect panic message.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
 2024-06-01 12:43:27 +00:00
Tim-Philipp Müller
6b628485c5 rtp: Add AC-3 RTP payloader/depayloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
 2024-06-01 12:43:27 +00:00
Tamas Levai
802ff6a67c net/quinn: Make QUIC role configurable 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
 2024-05-31 23:20:38 +02:00
Francisco Javier Velázquez-García
8fc652f208 webrtcsink: Refactor value retrieval to avoid lock poisoning 
When setting an incorrect property name in settings,
start_stream_discovery_if_needed would panic because it attempts to
unwrap a poisoned lock for settings.

This refactor avoids that situation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
 2024-05-31 08:10:23 +00:00
Francisco Javier Velázquez-García
568e8533fa webrtcsink: Fix typo in property name for av1enc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
 2024-05-31 08:10:23 +00:00
Sebastian Dröge
91bc39367b deny: Add another override for librespot for nix 2024-05-31 10:06:14 +03:00
Arun Raghavan
04e9e5284c webrtc: signaller: A couple of minor doc fixups 
The expectation is `Returns:`, not `Return:`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
 2024-05-30 22:16:46 +03:00
Arun Raghavan
1c54c77840 webrtcsink: Add a mechanism for SDP munging 
Unfortunately, server implementations might have odd SDP-related quirks,
so let's allow clients a way to work around these oddities themselves.
For now, this means that a client can fix up the H.264 profile-level-id
as required by Twitch (whose media pipeline is more permissive than the
WHIP implementation).

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/516
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
 2024-05-30 22:16:46 +03:00
Taruntej Kanakamalla
83f76280f5 net/webrtc: Example for whipserver 
rudimentary sample to test multiple WHIP client connections

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
 2024-05-29 21:03:27 +00:00
Taruntej Kanakamalla
712d4757c3 net/webrtc/whip_signaller: multiple client support in the server 
- generate a new session id for every new client
use the session id in the resource url

- remove the producer-peer-id property in the WhipServer signaler as it
is redundant to have producer id in a session having only one producer

- read the 'producer-peer-id' property on the signaller conditionally
if it exists else use the session id as producer id

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
 2024-05-29 21:03:27 +00:00
Taruntej Kanakamalla
de726ca8d2 net/webrtc: multi producer support in webrtcsrc 
- Add a new structure Session
  - manage each producer using a session
  - avoid send EOS when a session terminates, instead keep running
    waiting for any new producer to connect

- Maintain a bin element per session
  - each session bin encapsulates webrtcbin and the decoder if needed
    as well as the parser and filter if requested by the application
    (through request-encoded-filter)
  - this will be helpful to cleanup the session's respective elements
    when the corresponding producer terminates the session

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
 2024-05-29 21:03:27 +00:00
Seungha Yang
ebdcc403cf transcriberbin: Fix mux-method=cea708 
* Update "translation-languages" property to include G_PARAM_CONSTRUCT
so that it can be applied to initial state.

* Change default "translation-languages" value to be None instead of
cea608 specific one. Transcriberbin will be able to configure initia
state depending on selected mux method if "translation-languages" is
unspecified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1589>
 2024-05-30 04:40:09 +09:00
Matthew Waters
45800d7636 tttocea708: ensure periodic sync points in roll up mode 
Otherwise, without the relevant DefineWindow, then a receiver cannot
begin to display the captions from the middle of a stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1591>
 2024-05-29 11:15:10 +00:00
Sebastian Dröge
a7418fb483 rtp: Use released version of rtcp-types 2024-05-29 10:30:40 +03:00
Matthew Waters
df32e1ebfa rtpsend: ensure only a single rtcp pad push 
Otherwise, it can occur that multiple rtcp packets may be produced out
of order.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
525179f666 rtpbin2: handle ssrc collisions 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Nirbheek Chauhan
9485265769 rtspsrc2: Update rtpbin2 support to use rtprecv and rtpsend 
USE_RTPBIN2 is now USE_RTP2 because there is no "rtpbin2" now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
1600d3b055 rtpbin2: split send and receive halves into separate elements 
There is now two elements, rtpsend and rtprecv that represent the two
halves of a rtpsession.  This avoids the potential pipeline loop if two
peers are sending/receiving data towards each other.  The two halves can
be connected by setting the rtp-id property on each element to the same
value and they will behave like a combined rtpbin-like element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
0121d78482 rtpbin2: expose session signals for new/bye ssrc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
d480c6c2d3 rtpbin2/config: add stats to session GObject 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
7d5789032a rtpbin2/config: add a new-ssrc signal 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
06f40e72cb rtpbin2: implement a session configuration object 
Currently only contains pt-map

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
48e7a2ed06 jitterbuffer: handle flush-start/stop 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Matthew Waters
66306e32f2 jitterbuffer: remove mpsc channel for every packet 
It is very slow.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Mathieu Duponchelle
327f563e80 jitterbuffer: implement support for serialized events / queries 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Mathieu Duponchelle
74ec83a0ff rtpbin2: implement and use synchronization context 
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 19:58:09 +10:00
Mathieu Duponchelle
1865899621 rtpbin2: implement jitterbuffer 
The jitterbuffer implements both reordering and duplicate packet
handling.

Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
2b4ec75bc5 rtpbin2: Add support for receiving rtcp-mux packets 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
e09ad990fa rtpbin2: Implement support for reduced size RTCP (RFC 5506) 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
1e4a966c92 rtpbin2: Add support for sending NACK/PLI and FIR 
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
66c9840ad8 rtpbin2: Add handling for receiving NACK/PLI and FIR 
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Matthew Waters
2c86f18a99 rtpbin2: add support for RFC 4585 (RTP/AVPF) 
Implements the timing rules for RTP/AVPF

Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Matthew Waters
27ad26c258 rtp: Initial rtpbin2 element 
Can receive and recevie one or more RTP sessions containing multiple
pt/ssrc combinations.

Demultiplexing happens internally instead of relying on separate
elements.

Co-Authored-By: François Laignel <francois@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
 2024-05-28 17:35:41 +10:00
Sebastian Dröge
984a9fe5ff rtp: Don't restrict payload types for payloaders 
WebRTC uses payload types 35-63 as dynamic payload types too to be able
to place more codec variants into the SDP offer.

Instead of allowing just certain payload types, completely remove any
restrictions and let the user decide. There's technically nothing wrong
with using any payload type, especially when using the encoding-name.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1587>
 2024-05-27 09:34:16 +00:00
Liam
b4fd6cf362 aws: Add system-defined metadata options to both sinks 
Add to awss3sink and awss3putobjectsink elements the following
paramerters which are set on the uploaded S3 objects:

* cache-control;
* content-encoding; and
* content-language

Bugfix: Set the content-type and content-disposition values in the S3
putobject call. Previously the params were defined on the element but
unused.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1585>
 2024-05-27 10:25:22 +03:00
Tim-Philipp Müller
4f74cb7958 rtp: klv: add test for fragmented payloads with packet loss 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
b6e24668a7 rtp: klv: add unit test with some packet loss 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
92a1e222f4 rtp: tests: add functionality to drop RTP packets after payloading 
Add ExpectedPacket::drop() to flag RTP packets that should not
be forwarded to the depayloader.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-26 12:34:44 +03:00