Commit graph

117 commits

Author SHA1 Message Date
François Laignel
c5e7e76e4d webrtcsrc: add do-retransmission property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1509>
2024-03-21 07:25:30 +00:00
Tim-Philipp Müller
ce3960f37f rtp: Add MPEG-TS RTP payloader
Pushes out pending TS packets on EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2024-03-16 10:07:37 +00:00
Tim-Philipp Müller
9f07ec35e6 rtp: Add MPEG-TS RTP depayloader
Can handle different packet sizes, also see:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1310

Has clock-rate=90000 as spec prescribes, see:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/691

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2024-03-16 10:07:37 +00:00
Nirbheek Chauhan
6f8fc5f178 meson: Disable docs completely when the option is disabled
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1496>
2024-03-14 15:30:17 +05:30
Guillaume Desmottes
03abb5c681 spotify: document how to use with non Facebook accounts
See discussion on #203.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1490>
2024-03-11 09:46:40 +01:00
Olivier Crête
15e7a63e7b originalbuffer: Pair of elements to keep and restore original buffer
The goal is to be able to get back the original buffer
after performing analysis on a transformed version. Then put the
various GstMeta back on the original buffer.

An example pipeline would be
.. ! originalbuffersave ! videoscale ! analysis ! originalbufferestore ! draw_overlay ! sink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1428>
2024-03-08 15:15:13 -05:00
Guillaume Desmottes
612f863ee9 webrtc: janusvrwebrtcsink: add 'use-string-ids' property
Instead of exposing all ids properties as strings, we now have two
signaller implementations exposing those properties using their actual
type. This API is more natural and save the element and application
conversions when using numerical ids (Janus's default).

I also removed the 'joined-id' property as it's actually the same id as
'feed-id'. I think it would be better to have a 'janus-state' property or
something like that for applications willing to know when the room has
been joined.
This id is also no longer generated by the element by default, as Janus
will take care of generating one if not provided.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1486>
2024-03-07 09:34:58 +01:00
Sebastian Dröge
2839e0078b rtp: Port RTP AV1 payloader/depayloader to new base classes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1472>
2024-03-06 09:40:35 +00:00
Jordan Yelloz
fa006b9fc9 webrtcsrc: Added LiveKit source element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-28 11:25:58 -07:00
Sebastian Dröge
f563f8334b rtp: Add PCMU/PCMA RTP payloader / depayloader elements
These come with new generic RTP payloader, RTP raw-ish audio payloader
and RTP depayloader base classes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2024-02-23 14:43:45 +02:00
Jordan Yelloz
67b7cf9764 webrtcsink: Added sinkpad with "msid" property
This forwards to the webrtcbin sinkpad's msid when specified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-02-12 15:04:44 +00:00
Nirbheek Chauhan
e59f3bbe58 rtspsrc2: Increase RTP timeout to 5 seconds, matching rtspsrc
Also fix some logging.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
437326ebfd rtspsrc2: Allocate a buffer pool for UDP RTP data
Control the size with a receive-mtu property

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
086ffd7aff New RTSP source plugin with live streaming support
GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI]

Features:
* Live streaming N audio and N video
  - With RTCP-based A/V sync
* Lower transports: TCP, UDP, UDP-Multicast
* RTP, RTCP SR, RTCP RR
* OPTIONS DESCRIBE SETUP PLAY TEARDOWN
* Custom UDP socket management, does not use udpsrc/udpsink
* Supports both rtpbin and the rtpbin2 rust rewrite
  - Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs)
* Properties:
  - protocols selection and priority (NEW!)
  - location supports rtsp[ut]://
  - port-start instead of port-range

Co-Authored-by: Tim-Philipp Müller <tim@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:18 +05:30
Eva Pace
80b58f3b45 net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller
The JanusVRWebRTCSink is a new plugin that integrates with the Video
Room plugin of the Janus Gateway, which simplifies WebRTC communication.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
2024-01-17 20:33:57 +00:00
Maksym Khomenko
17f0b61576 webrtcsink: add payloader-setup signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
2023-12-23 08:02:08 +00:00
Sebastian Dröge
df1f986239 Update plugin documentation cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
2023-12-22 11:41:01 +02:00
Arun Raghavan
06d96ec5a2 aws: Add plugin docs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 16:13:48 -05:00
Taruntej Kanakamalla
43ee6bfc1c net/webrtc: add whipserversrc
Implement new signaller WhipServerSignaller
 - an http server using 'warp'
 - handlers for the POST, OPTIONS, PATCH and DELETE
 - fixed path `/whip/endpoint` as the URI
 - fixed value 'whip-client' as the producer peer id
 - fixed resource url `/whip/resource/whip-client`

Derive whipserversrc element from BaseWebRTCSrc
 - implement constructed method for ObjectImpl to set
  non-default signaller, i.e., WhipServerSignaller
 - bind the properties stun-server and turn-servers to those on
   the Signaller

Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller
 - it will be emitted by the webrtcsrc when the webrtcbin element is ready
 - the closure for this signal will in turn connect to webrtcbin's ice-gathering-state
   and perform send with the answer sdp via the channel
 - the WhipServer will hold its HTTP response in POST handler until this signal
   is received or timeout which happens early

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
ed3aa740be net/webrtc: deprecate consumer-added on the signaller
add a new signal webrtcbin-ready in this place doing same
thing but can be used for both consumers and producers

Please note this change is only to the consumer-added
signal on the signaller interface.
The consumer-added signal on the webrtcsink is unchanged

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
a0638ec983 net/webrtc: Extract BaseWebRTCSrc
Define a Base for all the webrtcsrc type elements
so they can all be derived from it. Similar to base
element defined for webrtcsink type elements

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Maksym Khomenko
e5fd2c3568 webrtcsrc: add turn-servers property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1380>
2023-11-04 10:19:45 +00:00
François Laignel
50dd519c4f net/webrtcsrc: define signaller property as CONSTRUCT_ONLY
The "signaller" property used to be defined as MUTABLE_READY which meant that
the property was always set after `constructed()` was called.

Since `connect_signaller()` was called from `constructed()`, only the default
signaller was used.

This commit sets the "signaller" property as CONSTRUCT_ONLY. Using a builder,
this property will now be set before the call to `constructed()`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1324>
2023-10-12 17:38:09 +00:00
Stéphane Cerveau
68c2d27e8d fmp4mux: specify the fragment duration unit
The fragment duration is expressed in nanoseconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1348>
2023-10-04 12:47:15 +02:00
Sebastian Dröge
747d9bfc6e Update plugins cache for updated raw video caps 2023-10-03 11:12:39 +03:00
Seungha Yang
ed4181617a hlssink3: Update plugin docs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-25 21:34:59 +09:00
rajneeshksoni
a7fe24a294 hlssink3: Add property track-pipeline-clock-for-pdt.
This is required to take care of clock skew between
system time and pipeline time.
`track-pipeline-clock-for-pdt: true` mean utd time is
sampled for first segment and for subsequent segments
keep adding the time based on pipeline clock. difference
of segment duration and PDT time will match.
track-pipeline-clock-for-pdt: false` mean utd time is
sampled for each segment. system time may jump forward
or backward based on adjustments. If application needs
to synchronization of external events `false` is
recommended.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1145>
2023-09-20 13:54:48 +03:00
rajneeshksoni
4be24fdcaf hlssink3: Allow adding EXT-X-PROGRAM-DATE-TIME tag.
- connect to `format-location-full` it provide the first
sample of the fragment. preserve the running-time of the
first sample in fragment.
- on fragment-close message, find the mapping of running-time
to UTC time.
- on each subsequent fragment, calculate the offset of the
running-time with first fragment and add offset to base
utc time

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1145>
2023-09-20 13:54:48 +03:00
Sebastian Dröge
95a7a3c0ec gtk4: Only support RGBA/RGB in the GL code path
For all other component orderings a shader is necessary to re-order the
components for what GTK expects.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1312>
2023-09-20 13:22:41 +03:00
Seungha Yang
1de7754616 webrtcsink: Add support for d3d11 memory and qsvh264enc
Adding d3d11 memory and qsvh264enc support

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1322>
2023-09-15 00:26:04 +09:00
Sebastian Dröge
146a96686b Update docs for new order of raw video formats
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1308>
2023-08-29 19:45:04 +03:00
Taruntej Kanakamalla
de6d2e7f40 net/webrtc: rename whipwebrtcsink as whipclientsink
add a deprecation warning in whipsink to indicate it
should be used only for RTP content
add documentation in whipsink code regarding usage and
deprecation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1282>
2023-08-26 10:53:30 +05:30
Mathieu Duponchelle
e905299eba generic: expose inter plugin
This new plugin exposes two elements, intersink and intersrc. These act
as wormholes for data in the same process and can be used to forward
data from one pipeline to another.

The implementation makes use of gstreamer-utils' StreamProducer, and
supports dynamically adding and removing consumers, before and after
producers, and changing producer names while PLAYING, both on the sink
and the src.

This initial implementation comes with a small demo, and a few tests.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1257>
2023-08-14 08:13:12 +00:00
Andoni Morales Alastruey
3000b08ec7 webrtcsrc: add support for navigation events
This provides support GstNavigation events handling in webrtcsrc so that
a GStreamer client can be used to control remotely a GStreamer server,
similar to how the web client is capable of controlling a wpesrc.
This is part of a larger set of patches that require more work on the
sinks and sources.
server: d3d11screencapturesrc ! webrtcsink enable-data-channel-navigation=true
client: webrtcsrc enable-data-channel-navigation=true ! d3d11videosink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
2023-08-10 17:43:51 +00:00
Loïc Le Page
7af2ff0843 net/webrtc/signaller: advertise running producers in Listener mode
When starting a webrtcsrc-signaller client in Listener mode, only the producers
started after the client connection were advertised. All currently
running producers were ignored unlike the gstwebrtc-api behavior. This
commit now lists all running producers when the client Listener connects
and advertises them through the "producer-added" signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
2023-08-10 17:30:21 +02:00
Sebastian Dröge
983f990fe9 Update docs after GStreamer update on the CI 2023-07-06 13:48:59 +03:00
Olivier Crête
793ee66afa webrtcsink: Add LiveKit WebRTC sink and signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-07-05 21:43:17 +00:00
Mathieu Duponchelle
84a33ca7b9 webrtcsink: bring in signalling code from whipsink as a signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-06-16 00:32:56 +02:00
Vivia Nikolaidou
063871a1eb togglerecord: Add support for non-live inputs
Live input + is-live=false:
    While not recording, drop input
    When recording is started, offset to collapse the gap

Live input + is-live=true:
    While not recording, drop input
    Don't modify the offset

Non-live input + is-live=false:
    While not recording, block input
    Don't modify the offset

Non-live input + is-live=true:
    While not recording, block input
    When recording is started, offset to current running time

Co-authored-by: Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1206>
2023-06-14 15:58:04 +03:00
Guillaume Desmottes
4683291c1f fallbackswitch: add 'stop-on-eos' property
Fix the following use case:
- main input of fallbackswitch is finite (a media file)
- fallback input is infinite (videotestsrc)
- main input is done and send eos, which is propagated downstream
- fallbackswitch switches to fallback, sending STREAM_START which reset
  EOS downstream (aggregator does that)
- fallback input keeps pushing buffers forever.

Solve it by adding a 'stop-on-eos' property so fallbackswitch stops
pushing property once the main input is eos.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1242>
2023-06-13 14:49:06 +02:00
Sebastian Dröge
c65b3429ad Use MPL as license specifier for plugins only requiring GStreamer < 1.20
And use MPL-2.0 for all that require GStreamer 1.20 or newer. The new
string is only allowed in 1.20 or newer and using it in older versions
causes failure to load the plugin.

All affected plugins are of course still MPL-2.0 licensed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/374

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1235>
2023-06-07 19:13:55 +03:00
Mathieu Duponchelle
fda5aed89f webrtcsink: encoded streams: address last review comments
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 16:05:28 +02:00
Mathieu Duponchelle
a20855dfd9 webrtcsink: expose consumer-pipeline-created signal
This signal is emitted as soon as the pipeline for each consumer
is created, and can be used by applications that require a greater
level of control over webrtcsink's internals.

An example is also provided to demonstrate usage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1220>
2023-05-25 13:15:52 +02:00
Guillaume Desmottes
7ebf2d7a4f fallbackswitch: document the pad priority ordering
I just wasted lots of time trying to figure out why my higher priority
pad wasn't used...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1208>
2023-05-15 16:13:20 +02:00
Seungha Yang
773fcd0780 transcriberbin: Add "language-code" property
Proxy the child transcriber element's property so that transcriberbin
can apply the property with required state management

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1205>
2023-05-10 19:12:01 +00:00
François Laignel
680d5221db net/webrtc: src: add signal "request-encoded-filter"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
2023-05-09 12:02:15 +02:00
Arun Raghavan
aabfb61834 ffv1dec: Drop rank for now
We'll keep the rank lower than avdec_ffv1, at least until we're
comfortable with support for the entire range of possible inputs working
well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1174>
2023-04-13 15:58:49 +00:00
Mathieu Duponchelle
f1fd8d84c3 webrtc: extract a BaseWebRTCSink
For documentation purposes, AwsKVSWebRTCSink should not inherit from
another element.

+ Mark base class as plugin API and update plugin cache

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:06:59 +00:00
Mathieu Duponchelle
355f925954 tttocea608: specify raw 608 field
The element can only output field=0 raw 608 data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1166>
2023-04-11 09:26:24 +10:00
Guillaume Desmottes
403004a85e fix typos
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
2023-04-10 13:35:32 +02:00