Commit graph

969 commits

Author SHA1 Message Date
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 

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