Commit graph

217 commits

Author SHA1 Message Date
Mathieu Duponchelle
39a8db51de transcriberbin: register pad templates for unsynced source pads
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1963>
2024-12-06 09:38:39 +00:00
Mathieu Duponchelle
4e722d6dcc speechmatics: expose unsynced pads on transcriber
This can be used for storing original transcripts for editing after the
fact.

Modeled on the aws transcriber, to be usable from transcriberbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1963>
2024-12-06 09:38:39 +00:00
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
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
4406851ae7 transcriberbin: add support for speech synthesis
This commit adds a new "synthesis-languages" property. Users can set it
to define a map of languages (typically translations) that should then
be routed through a "synthesis" bin, with its description specifiable
as the value of the map.

The output of this bin is then exposed as a new pad on the top-level
bin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1930>
2024-11-29 22:09:37 +00:00
Mathieu Duponchelle
169f7b762e transcriberbin: remove proxy properties on element
Those properties are now exposed on the sink pads, and should be used
directly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1925>
2024-11-20 12:52:17 +01:00
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
8dc8aa6f55 Update docs for quinn elements
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1775>
2024-11-18 10:59:53 +05:30
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
46f1fb4430 Update docs for quinn elements
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1634>
2024-11-15 23:14:13 +00:00
Sanchayan Maity
9b5f4b48ca docs: Fix CI failure due to missing GRAY10_LE16 entry
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1920>
2024-11-15 20:00:09 +05:30
Mathieu Duponchelle
4284fe953a transcriberbin: expose lateness property
Directly mapped to the lateness property on the transcriber object.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1910>
2024-11-14 16:40:14 +00:00
Taruntej Kanakamalla
b16379d00b threadshare: udp: add multicast-iface property
similar to the non threadshare counterparts, the ts-udpsink can accept
only one multicast interface and the ts-udpsrc can accept a list of
interfaces to be listening on for the multicast.

Use the getifaddrs crate to get the available network interfaces and filter
the desired interfaces from the available interfaces

Reuse a custom api written for PTP helper to join and leave multicast group
for IPv4 based addresses. Continue to use the UdpSocket crate's _multicast_v6
to join/leave an IPv6 multicast group

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1420>
2024-11-14 00:02:04 +00:00
Mathieu Duponchelle
635a83a8d7 transcriberbin: notify passthrough at the appropriate time
We want to enable passthrough internally, and only notify that
internally it has been enabled once the transcriber has been unlinked.

This way applications connected to the notify handler can synchronously
update the properties and attempt to disable passthrough again.

Doing so properly requires a refactoring of the transition to the
passthrough state, with the currently set passthrough mode maintained
separately from the target passthrough state.

This commit also finishes the work left incomplete in
17d7997137 by moving the passthrough
property to the sink pad class, making each transcriber passthrough
state independent from the others.

Also adds an example to demonstrate the behavior

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1856>
2024-11-05 12:23:54 +00:00
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
Thomas Goodwin
2cbf83d7e2 rsinter: add 'event-types' property to intersink
The event-types property defaults to Eos.  Setting an
array of additional, serialized event types results in
calling the producer.set_forward_events with those types
so that the events will be forwarded to any consumers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1875>
2024-10-29 15:17:24 +00:00
Mathieu Duponchelle
0ea7e9a59c cea708mux: add support for force-live
This requires a bump of the gst-base dependency to 1.22 in order to
use the new set_force_live() API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1876>
2024-10-24 15:10:15 +00:00
Guillaume Desmottes
4e501f276b spotify: add spotifylyricssrc element
Retrieve synced lyrics from Spotify.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1880>
2024-10-24 12:27:58 +00:00
Guillaume Desmottes
9b4942c6dd spotify: replace username/password auth with access token.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1801>
2024-10-23 12:37:05 +00:00
Sebastian Dröge
13dd1b03c9 docs: Add new ST-2038 elements
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1777>
2024-10-23 11:42:37 +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
Alicia Boya García
f12bd41510 Add streamgrouper element
streamgrouper allows to construct simple gst-launch pipelines where
streams of different group-ids are merged to use the same group-id.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1745>
2024-10-22 13:08:08 +00:00
Thibault Saunier
e630aab769 fmp4mux: Add support to write edts to handle audio priming
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1434>
2024-10-18 12:33:43 +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 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1505>
2024-10-10 10:59:50 -04:00
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
Jendrik Weise
d5a9c7a940 fmp4: Add split-at-running-time signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1761>
2024-09-20 12:35:24 +03:00
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
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
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
Jerome Colle
fef6601094 dav1ddec: add properties for film grain synthesis and in-loop filters
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1763>
2024-09-09 14:23:15 +00:00
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
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
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
Piotr Brzeziński
798936afc9 cmafmux: Add send-headers signal
Forces cmafmux to output headers for the init segment again, alongside the next chunk.
Needed for hlscmafsink to support changing output paths on the fly, without going back to READY.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1692>
2024-08-22 02:06:51 +00:00
Piotr Brzeziński
ad0a23fee7 cmafmux: Add opus support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1692>
2024-08-22 02:06:51 +00:00
Mathieu Duponchelle
170e769812 audio: add speechmatics transcriber
Element implemented around the Speechmatics API:

<https://docs.speechmatics.com/rt-api-ref>

The element also comes with translation support, and offers a similar
interface to the one exposed by `awstranscriber`.

The Speechmatics service has good accuracy, and can be deployed on
premises, offering an advantage over AWS transcribe.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1665>
2024-08-21 17:43:02 +00:00
Mathieu Duponchelle
1c48d7065d gstwebrtc-api example: add support for requesting mix matrix
This is one example of how a consumer might send over custom upstream
event requests to the producer.

As webrtcsink will deserialize numbers in priority as integers, we need
a custom stringifying function to ensure members of the matrix array are
indeed serialized with the floating point.

An optional stringifier parameter is thus added to the
sendControlRequest API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
2024-08-15 15:42:04 +00:00
Mathieu Duponchelle
9455e09d9f webrtcsink: expose properties for running web server
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-08-08 16:40:46 +02:00
Guillaume Desmottes
17910dd532 gtk4: add window-{width,height} property
Allow the application to pass the actual rendering size so overlays can
be rendered accordingly.

Fix 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1680>
2024-08-06 10:29:41 +00:00
Thibault Saunier
a05ab37b49 tracers: Add a tracer that dumps data flow into .pcap files
See documentation for more details

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/879>
2024-07-31 20:27:27 +00:00
François Laignel
34b791ff5e webrtc: add raw payload support
This commit adds support for raw payloads such as L24 audio to `webrtcsink` &
`webrtcsrc`.

Most changes take place within the `Codec` helper structure:

* A `Codec` can now advertise a depayloader. This also ensures that a format
  not only can be decoded when necessary, but it can also be depayloaded in the
  first place.
* It is possible to declare raw `Codec`s, meaning that their caps are compatible
  with a payloader and a depayloader without the need for an encoder and decoder.
* Previous accessor `has_decoder` was renamed as `can_be_received` to account
  for codecs which can be handled by an available depayloader with or without
  the need for a decoder.
* New codecs were added for the following formats:
  * L24, L16, L8 audio.
  * RAW video.

The `webrtc-precise-sync` examples were updated to demonstrate streaming of raw
audio or video.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1501>
2024-07-16 19:32:02 +00:00
Taruntej Kanakamalla
3a8462367e threadshare: udpsrc: add buffer-size property
Use buffer-size to set the receive buffer size
on the socket

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1636>
2024-07-15 12:13:41 +00:00
Taruntej Kanakamalla
276ec91cb2 threadshare: udpsrc: add loop property to set multicast loopback
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1636>
2024-07-15 12:13:41 +00:00