Commit graph

141 commits

Author SHA1 Message Date
Tim-Philipp Müller
2585639054 rtp: Add Opus RTP payloader/depayloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2024-05-18 09:29:29 +00:00
Martin Nordholts
9a7f37e2b7 rtpgccbwe: Support linear regression based delay estimation
In our tests, the slope (found with linear regression) on a
history of the (smoothed) accumulated inter-group delays
gives a more stable congestion control. In particular,
low-end devices becomes less sensitive to spikes in
inter-group delay measurements.

This flavour of delay based bandwidth estimation with Google
Congestion Control is also what Chromium is using.

To make it easy to experiment with the new estimator, as
well as add support for new ones in the future, also add
infrastructure for making delay estimator flavour selectable
at runtime.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1566>
2024-05-14 16:25:48 +00:00
Tamas Levai
71cd80f204 net/quinn: Enable client to keep QUIC conn alive
Co-authored-by: Felician Nemeth <nemethf@tmit.bme.hu>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1568>
2024-05-11 08:51:00 +02:00
Rafael Caricio
5549dc7a15 fmp4mux: Support AV1 packaging in the fragmented mp4 plugin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1544>
2024-05-10 20:59:49 +00:00
Sebastian Dröge
f265c3197b Update plugins cache JSON for new CI GStreamer version
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
2024-05-10 14:14:51 +03:00
Sebastian Dröge
7e09481adc rtp: Add JPEG RTP payloader/depayloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1543>
2024-05-10 11:12:49 +03:00
Sebastian Dröge
b4576a0074 gtk4: Fix description of the plugin
A paintable is not a widget and that aspect does not belong in the short
description anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1563>
2024-05-07 20:21:03 +03:00
Sanchayan Maity
80f8664564 net/quinn: Use camel case acronym
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1558>
2024-05-02 16:39:29 +00:00
Sanchayan Maity
096538989b docs: Add documentation for gst-plugin-quinn
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 22:30:23 +05:30
François Laignel
16b0a4d762 rtp: add mp4gpay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
François Laignel
b588ee59bc rtp: add mp4gdepay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
François Laignel
5466cafc24 rtp: add mp4apay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
François Laignel
812fe0a9bd rtp: add mp4adepay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
Sebastian Dröge
3dd800ac77 gtk4paintablesink: Implement child proxy interface
This allows setting properties on the paintable from gst-launch-1.0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-26 12:29:10 +03:00
Sebastian Dröge
c92462b240 gtk4: Implement support for directly importing dmabufs
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/441

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1547>
2024-04-26 12:29:10 +03:00
Mathieu Duponchelle
17d7997137 transcriberbin: add support for consuming secondary audio streams
In some situations, a translated alternate audio stream for a content
might be available.

Instead of going through transcription and translation of the original
audio stream, it may be preferrable for accuracy purposes to simply
transcribe the secondary audio stream.

This MR adds support for doing just that:

* Secondary audio sink pads can be requested as "sink_audio_%u"

* Sometimes audio source pads are added at that point to pass through
  the audio, as "src_audio_%u"

* The main transcription bin now contains per-input stream transcription
  bins. Those can be individually controlled through properties on the
  sink pads, for instance translation-languages can be dynamically set
  per audio stream

* Some properties that originally existed on the main element still
  remain, but are now simply mapped to the always audio sink pad

* Releasing of secondary sink pads is nominally implemented, but not
  tested in states other than NULL

An example launch line for this would be:

```
$ gst-launch-1.0 transcriberbin name=transcriberbin latency=8000 accumulate-time=0 \
      cc-caps="closedcaption/x-cea-708, format=cc_data" sink_audio_0::language-code="es-US" \
      sink_audio_0::translation-languages="languages, transcript=cc3"
    uridecodebin uri=file:///home/meh/Music/chaplin.mkv name=d
      d. ! videoconvert ! transcriberbin.sink_video
      d. ! clocksync ! audioconvert ! transcriberbin.sink_audio
      transcriberbin.src_video ! cea608overlay field=1 ! videoconvert ! autovideosink \
      transcriberbin.src_audio ! audioconvert ! fakesink \
    uridecodebin uri=file:///home/meh/Music/chaplin-spanish.webm name=d2 \
      d2. ! audioconvert ! transcriberbin.sink_audio_0 \
      transcriberbin.src_audio_0 ! fakesink
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1546>
2024-04-25 11:56:01 +02:00
Sebastian Dröge
66030f36ad tracers: Add a pad push durations tracer
This tracer measures the time it takes for a buffer/buffer list push to return.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1506>
2024-04-17 16:20:43 +03:00
Sebastian Dröge
d6a855ff1b rtp: Add VP8/9 RTP payloader/depayloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1487>
2024-04-15 14:03:56 +00:00
François Laignel
83d70d3471 webrtc: add RFC 7273 support
This commit implements [RFC 7273] (NTP & PTP clock signalling & synchronization)
for `webrtcsink` by adding the "ts-refclk" & "mediaclk" SDP media attributes to
identify the clock. These attributes are handled by `rtpjitterbuffer` on the
consumer side. They MUST be part of the SDP offer.

When used with an NTP or PTP clock, "mediaclk" indicates the RTP offset at the
clock's origin. Because the payloaders are not instantiated when the offer is
sent to the consumer, the RTP offset is set to 0 and the payloader
`timstamp-offset`s are set accordingly when they are created.

The `webrtc-precise-sync` examples were updated to be able to start with an NTP
(default), a PTP or the system clock (on the receiver only). The rtp jitter
buffer will synchronize with the clock signalled in the SDP offer provided the
sender is started with `--do-clock-signalling` & the receiver with
`--expect-clock-signalling`.

[RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1500>
2024-04-12 14:18:09 +02:00
Guillaume Desmottes
61c9cbdc8f uriplaylistbin: allow to change 'iterations' property while playing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1492>
2024-04-11 11:13:20 +02:00
Matthew Waters
fbce73f6fc closedcaption: implement cea708overlay element
Can overlay any single CEA-708 service or any single CEA-608 channel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1519>
2024-04-10 06:55:34 +00:00
Sanchayan Maity
a3e30b499f aws: Introduce a property to use path-style addressing
AWS SDK switched to virtual addressing as default instead of path
style earlier. While MinIO supports virtual host style requests,
path style requests are the default.

Introduce a property to allow the use of path style addressing if
required.

For more information, see
https://github.com/minio/minio/blob/master/docs/config/README.md#domain
https://docs.aws.amazon.com/AmazonS3/latest/userguide/VirtualHosting.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1527>
2024-04-10 00:23:22 +00:00
Simonas Kazlauskas
5d939498f1 mp4/fmp4: support flac inside the iso (f)mp4 container
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1401>
2024-04-09 14:37:05 +03:00
Matthew Waters
e1cd52178e transcriberbin: also support 608 inside 708
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters
55b4de779c tttocea708: add support for writing 608 compatibility bytes
608 compatibility bytes are generated using the same functionality as
tttocea608.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters
df30d2fbd3 transcriberbin: add support for generating cea708 captions
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters
b0cf7e5c81 cea708mux: add element muxing multiple 708 caption services together
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters
756abbf807 tttocea708: add element converting from text to cea708 captions
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1406>
2024-03-28 13:46:28 +11:00
Matthew Waters
e868f81189 gopbuffer: implement element buffering of an entire GOP
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1349>
2024-03-26 15:29:48 +11:00
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
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