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Author SHA1 Message Date
Mathieu Duponchelle
612279f421 gst_plugins_cache.json: generate tracer objects documentation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2007>
 2024-12-23 18:37:55 +02:00
Sanchayan Maity
6e67e7c378 docs: Fix CI failure due to missing GRAY10_LE16 entry 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2007>
 2024-12-23 16:42:44 +02:00
Thibault Saunier
e0bef7d179 docs: Allow updating the plugins_cache.json files without generating documentation 
There is no dependencies between the 2 things, and it is often useful to be able
to update the cache even if we are not generating the documentation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2000>
 2024-12-20 10:51:25 +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
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
Sebastian Dröge
7b4a2daed0 rtpav1depay: Add wait-for-keyframe and request-keyframe properties 
These behave the same as the properties in other depayloaders. Keyframe
detection is based on the N flag in the aggregation header.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/598

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1825>
 2024-09-28 10:51:05 +01:00
Mathieu Duponchelle
9331824479 webrtcsrc: expose MSID property on source pad 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1796>
 2024-09-21 00:27:57 +02:00
Jerome Colle
18771be680 dav1ddec: add properties for film grain synthesis and in-loop filters 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1771>
 2024-09-09 18:29:31 +01:00
Sanchayan Maity
988c58de43 whepsrc: Fix incorrect default caps 
add-transceiver needs application/x-rtp caps and not raw caps. We were
providing raw caps which is incorrect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1756>
 2024-08-27 09:51:05 +01:00
Piotr Brzeziński
bd5154ebe4 cmafmux: Add opus support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1742>
 2024-08-23 08:11:46 +00:00
Guillaume Desmottes
5f234d734e gtk4: add window-{width,height} property 
Allow the application to pass the actual rendering size so overlays can
be rendered accordingly.

Fix #562

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1697>
 2024-08-07 10:37:04 +01:00
François Laignel
170cb76458 rav1enc: document bitrate property unit 
See:

e34e772e47/src/rate.rs (L365)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1667>
 2024-07-16 12:47:33 +03:00
Sebastian Dröge
8522c8a445 gtk4: Add support for rotations / flipping 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/284

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1590>
 2024-07-07 07:43:49 +00:00
Sebastian Dröge
6e974cf4b9 gtk4: Document paintable properties correctly 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1655>
 2024-07-06 11:36:55 +00:00
Philippe Normand
eee93aea52 rtp2: Fix typo on auto-header-extension property name 
The rtp (de)pay elements use auto-header-extension so the new elements should do
the same.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1646>
 2024-07-02 09:35:39 +01:00
Edward Hervey
95ae67752f net: New mpegtslive element 
This element allows wrapping an existing live "mpeg-ts source" (udpsrc,
srtsrc,...) and providing a clock based on the actual PCR of the stream.

Combined with `tsdemux ignore-pcr=True` downstream of it, this allows playing
back the content at the same rate as the (remote) provider **and** not modify
the original timestamps.

Co-authored-by: Sebastian Dröge <slomo@coaxion.net>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1640>
 2024-07-01 15:29:22 +02:00
Sebastian Dröge
960529d90d livesync: Add sync property for allowing to output buffers as soon as they arrive 
By default livesync will wait for each buffer on the clock. If sync is
set to false, it will output buffers immediately once they're available
and only waits on the clock for outputting gap filler buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1635>
 2024-06-26 16:21:42 +00:00
Sanchayan Maity
0bd98e2c34 net/quinn: Allow dropping buffers when buffer size exceeds maximum datagram size 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
 2024-06-25 20:15:40 +05:30
Sanchayan Maity
e00ebca63f net/quinn: Add stats property for connection statistics 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
 2024-06-25 20:15:40 +05:30
Sanchayan Maity
cf7172248c net/quinn: Allow setting some parameters from TransportConfig 
As of now, we expose the below four properties from `TransportConfig`.
- Initial MTU
- Minimum MTU
- Datagram receive buffer size
- Datagram send buffer size

Maximum UDP payload size from `EndpointConfig` and upper bound from
`MtuDiscoveryConfig` are also exposed as properties.

See the below documentation for further details.
- https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html
- https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html
- https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html

While at it, also clean up passing function parameters to the functions
in utils.rs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
 2024-06-25 20:15:40 +05:30
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
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
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
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
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
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
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
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
566e6443f4 rtp: Add KLV RTP payloader/depayloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
 2024-05-25 20:21:50 +03:00
cdelguercio
f5a7de9dc3 webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
 2024-05-23 10:16:59 +03:00
Tim-Philipp Müller
2585639054 rtp: Add Opus RTP payloader/depayloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
 2024-05-18 09:29:29 +00:00
Martin Nordholts
9a7f37e2b7 rtpgccbwe: Support linear regression based delay estimation 
In our tests, the slope (found with linear regression) on a
history of the (smoothed) accumulated inter-group delays
gives a more stable congestion control. In particular,
low-end devices becomes less sensitive to spikes in
inter-group delay measurements.

This flavour of delay based bandwidth estimation with Google
Congestion Control is also what Chromium is using.

To make it easy to experiment with the new estimator, as
well as add support for new ones in the future, also add
infrastructure for making delay estimator flavour selectable
at runtime.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1566>
 2024-05-14 16:25:48 +00:00
Tamas Levai
71cd80f204 net/quinn: Enable client to keep QUIC conn alive 
Co-authored-by: Felician Nemeth <nemethf@tmit.bme.hu>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1568>
 2024-05-11 08:51:00 +02:00
Rafael Caricio
5549dc7a15 fmp4mux: Support AV1 packaging in the fragmented mp4 plugin 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1544>
 2024-05-10 20:59:49 +00:00
Sebastian Dröge
f265c3197b Update plugins cache JSON for new CI GStreamer version 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
 2024-05-10 14:14:51 +03:00
Sebastian Dröge
7e09481adc rtp: Add JPEG RTP payloader/depayloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1543>
 2024-05-10 11:12:49 +03:00
Sebastian Dröge
b4576a0074 gtk4: Fix description of the plugin 
A paintable is not a widget and that aspect does not belong in the short
description anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1563>
 2024-05-07 20:21:03 +03:00
Sanchayan Maity
80f8664564 net/quinn: Use camel case acronym 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1558>
 2024-05-02 16:39:29 +00:00
Sanchayan Maity
096538989b docs: Add documentation for gst-plugin-quinn 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
 2024-05-01 22:30:23 +05:30
François Laignel
16b0a4d762 rtp: add mp4gpay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
 2024-04-29 13:33:42 +00:00
François Laignel
b588ee59bc rtp: add mp4gdepay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
 2024-04-29 13:33:42 +00:00
François Laignel
5466cafc24 rtp: add mp4apay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
 2024-04-29 13:33:42 +00:00
François Laignel
812fe0a9bd rtp: add mp4adepay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
 2024-04-29 13:33:42 +00:00