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Author SHA1 Message Date
Mathieu Duponchelle
c51a65d973 awstranscriber, speechmatics: store language tags on translation source pads 
In order to do so we need to activate the pad as soon as it is added,
which means we can no longer start the task at this point, instead wait
for stream-start to do so now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2029>
 2025-01-20 14:27:05 +00:00
Sebastian Dröge
8e62e54cc9 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/2037>
 2025-01-15 09:43:12 +00:00
Sebastian Dröge
536f4db5c1 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/2037>
 2025-01-15 09:43:12 +00:00
Sebastian Dröge
7b4665c793 Fix some new clippy 1.84 warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2032>
 2025-01-10 10:08:38 +02:00
Sebastian Dröge
81ff664666 rtp: Add AMR NB/WB RTP payloader/depayloader 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2016>
 2025-01-02 16:42:14 +00:00
Sebastian Dröge
38e8134edd Update to itertools 0.14 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2018>
 2025-01-01 12:46:07 +02:00
Sanchayan Maity
3aa1fa81b5 net/quinn: Update QUIC multiplexing examples for WebTransport 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1966>
 2024-12-30 09:40:43 +05:30
Sanchayan Maity
59cc4af3ba net/quinn: Support stream multiplexing in quinnwtclientsrc 
While at it, drop the use-datagram property since the data handler
thread receives data for both streams and datagram irrespective of
the property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1966>
 2024-12-30 09:40:43 +05:30
Sanchayan Maity
a02296eb95 net/quinn: Support stream multiplexing in quinnwtserversink 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1966>
 2024-12-30 09:40:43 +05:30
Ruben Gonzalez
ebfa0fb890 deps: update itertools to 0.13 
same used in gstreamer-rs

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/2002>
 2024-12-20 16:23:52 +00:00
Sanchayan Maity
e21e07c46a net/quinn: Fix ChildProxy implementation for muxer & demuxer 
The demuxer did not need the ChildProxy implementation while
the muxer was missing the call to child_added, child_removed
and the interface entry in ObjectSubclass.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1998>
 2024-12-20 17:45:50 +05:30
Thibault Saunier
1e3eef253b 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/1996>
 2024-12-19 14:32:16 +00:00
Sebastian Dröge
7d4ddc7eb9 webrtc: Specify to use playbin3 instead of playbin 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1995>
 2024-12-18 07:31:17 +00:00
Sebastian Dröge
248b7ac059 webrtcsink: Configure custom host/port on the signaller when running signalling server internally 
Otherwise it just tries connecting to the default URL, which doesn't
work if either the host or the port are changed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1994>
 2024-12-17 16:22:41 +02:00
Sebastian Dröge
6a8f1bdc61 mpegtslivesrc: Parse PES packets and check for reasonable PTS/DTS 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1977>
 2024-12-13 13:11:16 +00:00
Sebastian Dröge
44978159a3 mpegtslivesrc: Refactor section parser 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1977>
 2024-12-13 13:11:16 +00:00
Mathieu Duponchelle
8886cceaf0 webrtcsink: add nvh265enc support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1980>
 2024-12-11 08:07:15 +00:00
Mathieu Duponchelle
be00ae7999 aws/polly: expose property for overflow control 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1965>
 2024-12-10 14:19:30 +00:00
Andoni Morales Alastruey
1ba2468a05 quinn: fix clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Andoni Morales Alastruey
b020ae6fc2 quinn: fix racy tests 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Andoni Morales Alastruey
2d6f084596 quinn: ignore the test using the hostname 
Ignore the test for now, since the CI runners only resolve to
an IPv6 address which are not handled correctly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Andoni Morales Alastruey
a791cfff2b quinn: allow unsecure connections in WebTransport elements 
WebTransport requires a secure connection, but certificates
can have a validity of 2 weeks. For testing, a new property
is added to allow unsecure connections.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Sanchayan Maity
be02c0e388 net/quinn: Move quinnwtclientsrc to PushSrc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Sanchayan Maity
850331699a net/quinn: Use LazyLock instead of once_cell::Lazy 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Andoni Morales Alastruey
d80c4c4351 quinn: add tests for WebTransport 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Ruben González
6fed4acf53 quinn: add a new WebTransport server sink 
Co-authored-by: Andoni Morales Alastruey <amorales@fluendo.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Andoni Morales Alastruey
ef21a6aa3b quinn: add a new WebTransport client element 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Andoni Morales Alastruey
62e49b3ed5 quinn: add support for Sec1 keys 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Andoni Morales Alastruey
cf8b49b257 quinn: make private key optional for clients 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Andoni Morales Alastruey
4104ebca25 quinn: cleanup transport config creation 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1867>
 2024-12-09 12:26:48 +00:00
Taruntej Kanakamalla
c9a0731e61 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/1970>
 2024-12-05 17:28:09 +05:30
Sebastian Dröge
050e582366 mpegtslivesrc: Reset rate to 1/1 on disconts and flush observations 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1964>
 2024-12-03 10:38:48 +02:00
Guillaume Desmottes
45519a7d85 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/1929>
 2024-12-02 15:38:24 +00:00
Guillaume Desmottes
867c2b78b6 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/1929>
 2024-12-02 15:38:24 +00:00
Sebastian Dröge
6ee745edee Update for GLib signal accumulator API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1954>
 2024-11-30 15:10:06 +02:00
Sebastian Dröge
6aeb3f2af2 Fix / silence various new Rust 1.83 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1954>
 2024-11-30 14:57:24 +02:00
Mathieu Duponchelle
9c844acba5 aws/transcriber: fix unsynced_translate_src_%u presence 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1930>
 2024-11-29 22:09:37 +00:00
Mathieu Duponchelle
f16f8f69d5 aws/transcriber: don't adjust late item duration 
It makes for a better user experience to simply adjust the pts of a late
item, but to preserve its duration: for instance a speech synthesis
element might use the duration as a hint for speeding up the audio.

Future late items may also be similarly offset anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1930>
 2024-11-29 22:09:37 +00:00
Mathieu Duponchelle
9972c83c60 aws/transcriber: put posting of warning messages behind property 
Repeated warning messages are fairly noisy with gst-launch, better make
this behavior opt-in.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1930>
 2024-11-29 22:09:37 +00:00
Mathieu Duponchelle
4d45ae0e44 aws/polly: expose ssml-set-max-duration property 
With standard voices, AWS polly supports passing a max-duration
attribute.

When the element gets raw text passed in, it can wrap it as SSML and set
the max duration attribute, this to make sure synthesized speech
doesn't overlap.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1930>
 2024-11-29 22:09:37 +00:00
Mathieu Duponchelle
c57b74e269 awstranscriber: release matching unsynced pad along request pads 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1930>
 2024-11-29 22:09:37 +00:00
Mathieu Duponchelle
4720b575b6 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/1955>
 2024-11-29 18:52:41 +01:00
Ruben Gonzalez
f646504fce webrtcsink: add openh264enc support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1948>
 2024-11-29 13:44:11 +00:00
Sebastian Dröge
f4d2bd1a5d 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/1949>
 2024-11-28 21:14:43 +00:00
Sanchayan Maity
c3de9e5927 net/quinn: Add examples for QUIC multiplexing & RoQ 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1937>
 2024-11-28 17:52:18 +00:00
Xavier Claessens
e5f3ab4053 webrtcsink: Ignore more fields in caps change 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1838>
 2024-11-26 15:49:21 -05:00
Diego Nieto
362216f40b net/webrtc: add whipclient example 
Add a simple example producing both audio and video to make it
work with the whipserver example

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1938>
 2024-11-25 20:29:43 +01:00
Diego Nieto
0135aea9e4 net/webrtc: whipserver example 
extend the example to support both audio and video conversions

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1938>
 2024-11-25 20:29:43 +01:00
Sebastian Dröge
347bee16d4 Update for GLib signal API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1936>
 2024-11-22 15:52:41 +02:00
François Laignel
a8146f333f all: use builder conditional setters where applicable 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1926>
 2024-11-21 12:57:16 +00:00
François Laignel
4262a8aafe all: update due to new has_property signature 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1926>
 2024-11-21 12:57:16 +00:00
Sebastian Dröge
160f08889f 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/1933>
 2024-11-21 11:40:24 +00:00
Sebastian Dröge
d32d499856 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/1933>
 2024-11-21 11:40:24 +00:00
Matthew Waters
25bb2a12f1 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/1931>
 2024-11-21 17:15:44 +11:00
Mathieu Duponchelle
b5bd7d047c awstranscribe: output original transcripts to separate pad 
When the transcriber is used in a live situation, it can be useful
to save a transcript for editing after the fact when producing a
VOD.

Each source pad now gets an "unsynced_" pendant. That unsynced pad
is pushed to from the context of the "live" source pad task. Flow
returns from the unsynced pads are ignored, we simply check the
last flow return before attempting to push the next transcript.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1915>
 2024-11-18 17:30:54 +00:00
Sanchayan Maity
28e66e150f net/quinn: Use aggregator as base class for quinnroqmux 
While at it, also update and fix the docs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1775>
 2024-11-18 11:46:20 +05:30
Sanchayan Maity
accb6b02ea net/quinn: Add muxer and demuxer for RTP over QUIC 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1775>
 2024-11-16 11:46:13 +05:30
Sanchayan Maity
d5425c5225 net/quinn: Fix test using QUIC Stream 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
 2024-11-15 23:14:13 +00:00
Sanchayan Maity
5bf44b6187 net/quinn: Enable log feature 
This is required if and when we do need to capture logs from quinn for
debugging.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
 2024-11-15 23:14:13 +00:00
Sanchayan Maity
324f3531be net/quinn: Use aggregator as base class for quinnquicmux 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
 2024-11-15 23:14:13 +00:00
Sanchayan Maity
5c829e6ca8 net/quinn: Add quinnquicdemux to support stream demultiplexing 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
 2024-11-15 23:14:13 +00:00
Sanchayan Maity
f4ecf3873b net/quinn: Handle multiple stream connections in quinnquicsrc 
While at it, use PushSrc as base class. quinnquicsrc never supported
seeking and only ever operated in push mode. Length and offset for
create from BaseSrc was also never really honoured. Use PushSrc as
the base class which is more appropriate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
 2024-11-15 23:14:13 +00:00
Sanchayan Maity
babb6f360b net/quinn: Support stream multiplexing in quinnquicsink 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
 2024-11-15 23:14:13 +00:00
Sanchayan Maity
1cc2682b55 net/quinn: Add quinnquicmux to support stream multiplexing 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
 2024-11-15 23:14:13 +00:00
Sanchayan Maity
0eb3f52356 net/quinn: Add helper for queries 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
 2024-11-15 23:14:13 +00:00
Sanchayan Maity
0e89a79727 net/quinn: Add helper for adding stream id as meta to buffers 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
 2024-11-15 23:14:13 +00:00
Jerome Colle
f88c88ddb3 webrtcsink: set rtpgccbwe min bitrate 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1896>
 2024-11-07 18:00:12 +00:00
Sebastian Dröge
ef39046e18 Update to thiserror 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1911>
 2024-11-06 11:02:41 +02:00
Mathieu Duponchelle
5f8e8b4873 aws: add wrapper for the polly text to speech API 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1899>
 2024-11-05 08:46:48 +00:00
Xavier Claessens
372c44655a 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/1848>
 2024-10-30 12:36:01 +00:00
Mathieu Duponchelle
79845fd99a 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/1793>
 2024-10-28 13:13:31 +00:00
Sebastian Dröge
41ddbd8706 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/1887>
 2024-10-28 12:15:11 +00:00
Chris Bainbridge
5010ee872d webrtc: Fix Python custom signaller receiving SDP offer 
The GstWebRTC API web interface defaults to receiving an SDP offer and
generating an answer, but this can be overridden by entering "offer
options" before clicking to open the remote stream. The Python
webrtcsink-custom-signaller.py example failed in this mode as it was
coded to only generate an offer and receive an answer. Fix this by
implementing support for receiving an offer and sending an answer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1883>
 2024-10-28 11:23:32 +00:00
Chris Bainbridge
e30d80c71e webrtc: README: add webrtcsink-custom-signaller.py 
Document the Python webrtcsink custom signaller example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1888>
 2024-10-28 10:19:25 +00:00
Sebastian Dröge
98a87fb8f2 Update to quick-xml 0.37 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1891>
 2024-10-28 09:12:13 +00:00
Jerome Colle
0df2b72ff2 rtpbasedepay2: fix reference timestamp meta duplicates 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1890>
 2024-10-25 18:23:07 +02:00
Sebastian Dröge
f73317510e Update to quick-xml 0.36 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1881>
 2024-10-24 10:09:18 +03:00
Sanchayan Maity
af54b2396b 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/1878>
 2024-10-23 21:24:22 +05:30
kingosticks
a81b7f380f net/quinn: Fix test panic due to unset default crypto provider 
If another dep in the workspace pulls in a different rustls crypto
provider then we need to explicitly specify our default provider.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1801>
 2024-10-23 12:37:05 +00:00
Benjamin Gaignard
cf757e6ad2 relationmeta: Add onvifmeta2relationmeta element 
Add onvifmeta2relationmeta wich convert ONVIF metas
into relation metas and add them to buffer.
Used ONVIFS metas are removed from buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1464>
 2024-10-23 10:27:31 +00:00
Benjamin Gaignard
fc3cefc38c relationmeta: Add relationmeta2onvifmeta element 
Add relationmeta2onvifmeta which convert relation metas
to ONVIF metas and add them to buffer.
Used relation metas are removed from buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1464>
 2024-10-23 10:27:31 +00:00
Benjamin Gaignard
de153222da onvif: Add onvifmetadataextractor element 
onvifmetadataextractor does the opposite operation than
onvifmetadatacombiner, it extracts ONVIF metadatas from the
stream buffer and export them as buffers which could be
used by rtponvifpay element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1464>
 2024-10-23 10:27:31 +00:00
Benjamin Gaignard
43a4468263 onvif: onvifmetadataoverlay: Add support of Transformation node 
Transformation node allows to modify the coordinate system for
individual nodes of XML tree.
Parse the XML to get translation and scaling vectors values and
apply them when computing bounding box coordinates.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1464>
 2024-10-23 10:27:31 +00:00
Benjamin Gaignard
eddf443a8b onvif: Use CustomMeta::is_registered function 
Use CustomMeta::is_registered() to avoid registering twice
OnvifXMLFrameMeta type.
It will be useful later when adding relation meta elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1464>
 2024-10-23 10:27:31 +00:00
Sebastian Dröge
4abc5c7a48 Be stricter with Impl-trait bounds to enforce type hierarchies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1871>
 2024-10-22 13:43:12 +00:00
Sebastian Dröge
7e59c3f0fd Remove once_cell dependency 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1868>
 2024-10-21 17:53:18 +00:00
Sanchayan Maity
d6e7031799 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/1865>
 2024-10-21 09:27:33 +00:00
Sebastian Dröge
0e3d019e24 aws: Don't unnecessarily clone AWS behaviour version 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Sebastian Dröge
00a4398aee aws: Allow a deprecated BehaviourVersion for now 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Sebastian Dröge
120c62964d Update to bitstream-io 2.5 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Sebastian Dröge
d057488a20 aws: Update to test-with 0.14 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Sebastian Dröge
b43a778a8e Fix a couple of type hierarchy bugs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Sebastian Dröge
ec8759ae44 Fix various new clippy warnings due to MSRV bump 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1864>
 2024-10-20 19:53:15 +00:00
Sebastian Dröge
3e040c65f1 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/1861>
 2024-10-18 15:59:41 +00:00
Sebastian Dröge
54bc7a898e webrtc: Silence two new Rust 1.82 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1860>
 2024-10-17 21:38:10 +00:00
Mathieu Duponchelle
959463ff65 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/1852>
 2024-10-17 12:29:53 +00:00
Mathieu Duponchelle
ef06421a25 webrtcsrc: make updated transceiver retrieval backward compatible 
In 1.24 and before transceivers for remote sendonly medias are only
created at answer time. If that is the case, we can add the transceiver
ourself, it will get associated when creating the answer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1853>
 2024-10-16 14:48:20 +00:00
Mathieu Duponchelle
82d0eaf438 webrtcsrc: fix debug message on offer created 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1853>
 2024-10-16 14:48:20 +00:00
Mathieu Duponchelle
3d257b4819 webrtcsink: improve debut message when start session failed 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1853>
 2024-10-16 14:48:20 +00:00
Chris Bainbridge
785209cc7f 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/1854>
 2024-10-16 15:45:50 +02:00
Mathieu Duponchelle
5f0ca7acde webrtcsink: fix custom_signaller hanging 
Since 6a23ae168f, the chain function
of webrtcsink adds a custom meta on input buffers.

That custom meta was registered only by the class_init of the subclasses
of BaseWebRTCSink, but the custom signaller example uses
BaseWebRTCSink::with_signaller() directly.

Fix by registering the meta in BaseWebRTCSink::class_init()

Fixes: #610
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1854>
 2024-10-16 15:25:09 +02:00
Mathieu Duponchelle
5e49f1d10e webrtcsrc: address non-compliant transceiver creation 
Instead of adding transceivers explicitly then setting the remote
description, expecting the manually added transceivers to get picked
up, we pass a promise to set-remote-description-set, and set the
relevant properties on the automatically created transceivers at that
point.

We then call create-answer and proceed as before.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/596
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1829>
 2024-10-14 11:19:38 +00:00
Guillaume Desmottes
027eead86d 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/1505>
 2024-10-10 10:59:50 -04:00
Guillaume Desmottes
d8b9a7a486 webrtc: janus: fix typo in doc 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1505>
 2024-10-10 10:57:02 -04:00
Matthew Waters
d4fd21d197 rtp2/jitterbuffer: check for event query earlier 
If a serialized query arrives (e.g. allocation) and the jitterbuffer has never
received a packet, then jitterbuffer would never forward the serialized query
resulting in a hang.

Fix by forwarding queries/events before the conditions that require the first
packet to arrive.

Also update unit test to check for this scenario.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1846>
 2024-10-09 16:21:13 +00:00
Mathieu Duponchelle
b3ace3678b webrtcsink: fix naming of error dot files for discovery pipelines 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1843>
 2024-10-03 14:35:45 +00:00
Guillaume Desmottes
d9e8f4054c 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/1830>
 2024-09-30 14:40:48 +02:00
Sebastian Dröge
dcb072ee23 webrtc: livekit: Set connection earlier during setup 
Otherwise it's not available yet when handling the initial participants
that are already in the session when joining.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1794>
 2024-09-30 13:04:24 +03:00
Sebastian Dröge
cd2b641321 livekitwebrtcsrc: Add API for disabling/enabling a track 
A disabled track is still negotiated but no data is sent for it
temporarily until it is enabled again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1794>
 2024-09-30 13:04:24 +03:00
Sebastian Dröge
27dc76826e livekitwebrtcsrc: Add pad properties for various LiveKit participant / track metadata 
The content of the TrackInfo and ParticipantInfo structs is exposed as
gst::Structure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1794>
 2024-09-30 13:04:24 +03:00
Sebastian Dröge
ceb88d960f 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/1823>
 2024-09-27 12:25:16 +03:00
Mathieu Duponchelle
87c6719e1d webrtcsink: add define-encoder-bitrates signal 
When congestion control is used for a session with multiple encoders,
the default implementation simply divides the overall bitrate equally
between encoders.

This is not always desirable, and this patch exposes a new signal
that users can register to, with two arguments:

* The overall bitrate to allocate
* A structure with an encoder.stream_name -> bitrate mapping

Handlers should return a similar structure with a custom mapping.

An example is also provided.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1792>
 2024-09-25 15:19:44 +00:00
François Laignel
f532d523b2 webrtcsink: fix RFC7273 attributes 
RFC7273 related attributes are set in the SDP offer by passing them via the
transceiver `codec-preferences` signal. These attributes are intended to be set
at the media level so they must be prefixed by `a-` in the `Caps` argument to
the signal. Otherwise they end up under `a=fmtp`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1810>
 2024-09-25 09:30:48 +00:00
Mathieu Duponchelle
5c66d8c107 webrtcsrc: ensure source pad has msid when added 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1800>
 2024-09-24 14:50:30 +00:00
Mathieu Duponchelle
f70482d9bc webrtcsrc: fix default msid property value 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1800>
 2024-09-24 14:50:30 +00:00
Mathieu Duponchelle
a85b0cb72e webrtcsrc: expose MSID property on source pad 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1789>
 2024-09-20 09:31:57 +03:00
Jan Schmidt
7905626a5f onvifmetadatapay: Set output caps earlier 
As soon as input caps arrive, we can set output
caps. This means upstream can send gap events earlier,
before there is any actual metadata to send

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1779>
 2024-09-19 20:45:43 +10:00
Mathieu Duponchelle
6a23ae168f webrtcsink: implement mechanism to forward metas over control channel 
It may be desirable for the frontend to receive ancillary information
over the control channel.

Such information includes but is not limited to time code metas, support
for other metas (eg custom meta) might be implemented in the future, as
well as downstream events.

This patch implements a new info message, probes buffers that arrive at
nicesink to look up timecode metas and potentially forwards them to the
consumer when the `forward-metas` property is set appropriately.

Internally, a "dye" meta is used to trace the media identifier the
packet we are about to send over relates to, as rtpfunnel bundles all
packets together.

The example frontend code also gets a minor update and now logs info
messages to the console.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1749>
 2024-09-19 08:41:47 +00:00
Mathieu Duponchelle
db026ad535 gstwebrtc-api: expose API on consumer-session for munging stereo 
We cannot do that by default as this is technically non-compliant,
so we need to expose API to let the user opt into it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1754>
 2024-09-19 07:37:23 +00:00
Seungha Yang
1675e517b3 hlscmafsink: Add playlist-root-init property 
Adding a property to allow setting base path for init fragment to be
written in manifest file

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1773>
 2024-09-11 03:36:08 +09:00
Sebastian Dröge
c505d9a418 Update to async-tungstenite 0.28 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1772>
 2024-09-10 09:19:18 +03:00
Sebastian Dröge
24003a79f6 mpegtslivesrc: Make sure to use the object as context for all debug logs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
c32cb20906 mpegtslivesrc: Check if old compared to new PCR clock estimation is too far off 
It the difference between the two estimations is more than 1s then
consider this a discontinuity too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
c5b1ebc7d8 mpegtslivesrc: Fix order of parameters passed to add_observation() 
The first one should be the internal time, i.e. the monotonic clock time
in our case, and the second one the external time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
44f64fb3f6 mpegtslivesrc: Scale monotonic time on PCR disconts to allow for continuous clock times 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
453b3014e6 mpegtslivesrc: Set DISCONT flag on buffers at PCR discontinuities 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1767>
 2024-09-09 13:29:14 +00:00
Sebastian Dröge
a709eb96d9 Fix new Rust 1.81 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1768>
 2024-09-05 22:01:01 +03:00
Sebastian Dröge
295b9f01c2 ndisrc: Use correct receive time to re-initialize time tracking on disconts 
The base receive time should not be the monotonic system clock time, but
the monotonic system clock time adjusted by the current clock calibration.
For the first time this is equivalent as the clock calibration is the default,
but for further discontinuities it is not and would cause a
discontinuity in the clock times at this point.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1766>
 2024-09-05 10:18:48 +00:00
Mathieu Duponchelle
bfc32cc692 net/aws: fix spurious dispatch failures 
Since https://github.com/awslabs/aws-sdk-rust/discussions/956, the AWS
SDK errors out HTTP streams that do not transfer data for more than 5
seconds.

This probably should be an opt-in bhevior as it clearly not generically
useful, but as it is we need to opt out.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1760>
 2024-09-05 07:43:23 +00:00
Mathieu Duponchelle
65508cfe75 net/aws: don't discard errors from transcribe loop 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1760>
 2024-09-05 07:43:23 +00:00
Arun Raghavan
e72db57179 webrtc: Fix whipclientsink name in README 
The element name was changed, but the documentation wasn't updated to
match.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1764>
 2024-09-03 16:44:19 -04:00
Sebastian Dröge
871756bb70 ndisrc: Reset timestamp tracking if remote time goes backwards 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 20:53:13 +03:00
Sebastian Dröge
ee4416ee5f ndisrc: Add a clocked timestamp mode that provides a clock that follows the remote timecodes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 20:53:13 +03:00
Sebastian Dröge
ab3db748be ndisrc: Get rid of unnecessary AtomicRefCell dependency 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
0c4ec370cf ndisrc: Remove slope workaround in timestamping code 
This was needed for an old version of the NDI HX Camera iOS application
and is fixed since quite a while. Let's get rid of unnecessarily
complicated code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
57821cade4 ndisrc: Only calculate timecode/timestamp mappings if necessary 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
04da3b2047 ndisrc: receiver: Improve debug message when receiving frames 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
84fef267b5 ndisrc: receiver: Remove some code duplication 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
f2658eb773 ndisrc: Move from start/stop to change_state for slight code simplification 
All state change related code is in a single place now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1727>
 2024-08-29 16:32:51 +00:00
Sebastian Dröge
fc29ff7d8b hlssink3: Update to sprintf 0.2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1757>
 2024-08-27 21:06:52 +03:00
Mathieu Duponchelle
2f9bb62b6b gstwebrtc-api: create control data channel when offering 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1755>
 2024-08-27 07:52:12 +02:00
Sanchayan Maity
f3206c2e1a 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/1550>
 2024-08-26 19:56:34 +00:00
Sanchayan Maity
d274caeb35 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/1748>
 2024-08-26 19:44:37 +05:30
Mathieu Duponchelle
66727188cf net/aws: fix sanity check in transcribe loop 
When we receive a new alternative we want to avoid iterating out of
bounds, but the comparison between the current index and the length of
the alternative should not log an error when partial_index == length, as
Vec::drain(length..) is valid, and it is completely valid for AWS to
send us a new alternative with as many items as we have already
dequeued.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1751>
 2024-08-26 11:37:08 +02:00
Sanchayan Maity
320f36a462 hlssink3: Use fragment duration from splitmuxsink if available 
splitmuxsink now reports fragment offset and duration in the
splitmuxsink-fragment-closed message. Use this duration value
for the MediaSegment when available.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1728>
 2024-08-22 15:13:21 +00:00
Mathieu Duponchelle
4cf93ccbdb net/webrtc: Add missing npm command to README 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/589

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1746>
 2024-08-22 15:46:28 +02:00
Jerome Colle
dee0e32dde webrtcsink: add nvv4l2av1enc support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1735>
 2024-08-22 06:41:52 +00:00
Mathieu Duponchelle
8ad882bed5 gstwebrtc-api: address issues raised by mix matrix support 
1c48d7065d was mistakenly merged too
early, and there were concerns about the implementation and API design:

The fact that the frontend had to expose a text area specifically for
sending over a mix matrix, and had to manually edit in floats into the
stringified JSON was suboptimal.

Said text area was always present even when remote control was not
enabled.

The sendControlRequest API was made more complex than needed by
accepting an optional stringifier callback.

This patch addresses all those concerns:

The deserialization code in webrtcsink is now made more clever and
robust by first having it pick a numerical type to coerce to when
deserializing arrays with numbers, then making sure it doesn't allow
mixed types in arrays (or arrays of arrays as those too must share
the same inner value type).

The frontend side simply sends over strings wrapped with a request
message envelope to the backend.

The request text area is only shown when remote control is enabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1725>
 2024-08-22 05:54:46 +00:00
Piotr Brzeziński
c4bcdea830 hlscmafsink: Add new-playlist signal 
Allows you to switch output between folders without having to state change to READY to close the current playlist.
Closes the current playlist immediately and starts a new one at the currently set location.
Should be used after changing the relevant location properties.
Makes use of the send-headers signal in cmafmux.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1692>
 2024-08-22 02:06:51 +00:00
Mathieu Duponchelle
5dc2d56c0e webrtcsink: store mids per-session instead of globally 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1730>
 2024-08-21 21:20:40 +00:00