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Author SHA1 Message Date
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