Commit graph

86 commits

Author SHA1 Message Date
Sebastian Dröge
99a1e30ab0 webrtchttp: Fix documentation JSON
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 12:47:04 +02:00
Sanchayan Maity
cc7419308b webrtchttp: whipsink: Add candidates when sending the offer
WHIP endpoint providers like Cloudflare do not support Trickle ICE
and need candidates to be send along with the initial offer. Instead
of sending the offer in create-offer promise, send it once the ICE
candidates have been gathered.

While at it add properties to set STUN and TURN server along with the
ICE transport policy as at least when testing the Cloudflare WHIP
endpoint seems unreachable without it. This has also been observed
with Cloudflare provided demos.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Sanchayan Maity
6be5796888 Add a WebRTC WHEP source element
This implements WHEP specification based on
https://datatracker.ietf.org/doc/html/draft-murillo-whep-00

and has been tested with Cloudflare.

Server offers are likely to be removed from the WHEP specification
in upcoming revisions, to avoid compatibility issues. None of the
commercial services implementing WHEP support server initiated offers.
So we only support client side initiated offers.

Follows session setup and tear down as covered in Figure 1, Section 3
of the specification.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Seungha Yang
1c145e2ba9 dav1ddec: Lower rank to primary
The rank of AOM av1dec was demoted as secondary, and thus
primary rank is sufficient.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/996>
2022-12-01 17:03:31 +00:00
Jordan Petridis
975f0141be video/gtk4: Implement support for GLTextures when possible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/588>
2022-11-29 21:18:46 +02:00
Thibault Saunier
6b11284e8a webrtcsink: Make the turn-server prop a turn-servers list
So that we can simply specify several turn servers at once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/973>
2022-11-16 14:48:16 +00:00
Sebastian Dröge
2b4fd40d62 mp4: Add ONVIF non-fragmented MP4 muxer
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/974>
2022-11-10 12:59:53 +02:00
François Laignel
29a490f6dc ts: introduce ts-audiotestsrc
This makes it easy to generate "listenable" signals and to evaluate
discontinuities.

When the `tuning` feature is activated and the `main-elem` property
is set, the element can log the parked duration in %, which is an
image of the CPU usage for the ts-context.

This commit adds a test mode to `udpsrc-benchmark-sender` which
generates default audio buffers from `ts-audiotestsrc`. The `rtp`
mode is modified so that it uses `ts-audiotestsrc`.
2022-11-09 07:55:04 +00:00
Sebastian Dröge
c2f403f998 gst-plugin-mp4: Add new MP4 plugin with a non-fragmented MP4 muxer 2022-11-08 19:08:47 +02:00
Sebastian Dröge
f062b7cf0d fmp4mux: Make media/trak timescales configurable
And refactor a bit of code for easier extensibility.
2022-11-07 18:06:29 +00:00
Sebastian Dröge
6706f3a4b4 fmp4mux: Add initial Opus support
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/239
2022-11-03 16:53:01 +02:00
Sebastian Dröge
9504e4d540 docs: Remove some stale entries of renamed elements 2022-11-03 15:09:20 +02:00
Matthew Waters
8c8384c711 fmp4: add support for muxing VP9 streams in cmaf, dash and iso fmp4
As specified in https://www.webmproject.org/vp9/mp4/
2022-10-25 18:33:42 +11:00
Sebastian Dröge
fe8e0a8bf8 Update docs 2022-10-23 21:29:14 +03:00
b97a855a51 videocompare: Update README with reference 2022-10-23 17:16:22 +03:00
Nick Steel
c6578c8699 spotifyaudiosrc: convert to PushSrc
Fixes #252
2022-10-21 09:37:25 +03:00
Thibault Saunier
cbdd3a7f26 webrtc: Enhance documentation 2022-10-20 12:04:43 +00:00
Thibault Saunier
4942a916a8 webrtc: Uniformise GType names 2022-10-20 13:32:31 +02:00
Thibault Saunier
39c0dcb0d4 Plug webrtc in 2022-10-20 11:51:58 +02:00
9180d348bf Add video comparison element
New video/image comparison element, find images in the stream and post
metadata of comparisons of the video frames to the application.
2022-10-18 13:24:05 +00:00
Guillaume Desmottes
a5ebefd736 spotifyaudiosrc: implement URI handler
Fix #204
2022-10-18 08:31:59 +00:00
Vivia Nikolaidou
f11b0fa5eb plugins, examples, tutorials: Use AudioCapsBuilder and VideoCapsBuilder
Simplify caps creation code
2022-10-13 19:24:57 +00:00
Sebastian Dröge
97e0852156 ndi: Add NDI plugin to the docs 2022-10-12 22:25:13 +03:00
Seungha Yang
3d317b976e jsontovtt: Add timeout property
As described in the spec D.4 Automatic Caption Blanking,
allows automatic clear if user specified timeout value
2022-10-10 22:16:15 +09:00
Sebastian Dröge
38753b08ac fallbacksrc: Implement support for fallback streams 2022-09-27 12:56:15 +03:00
Mac Thi Kieu Van
98fc0d5bd6 ts-jitterbuffer: Declare request-pt-map signal 2022-09-21 11:31:06 +00:00
Sebastian Dröge
28151f2011 onvifmetadataparse: Push buffers from a separate source pad task to guarantee latency and generally improve correctness 2022-09-16 14:54:33 +03:00
Thibault Saunier
21e1756168 docs: Remove cruft 2022-09-15 19:27:52 -03:00
Vivienne Watermeier
8d73b5008a Add RTP de/payloader elements for AV1
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/881
2022-09-12 18:14:39 +03:00
Thibault Saunier
f966b3a573 meson: Fix usage of python3 variable which was renamed recently 2022-09-09 10:01:12 +03:00
Thibault Saunier
528bbcf67e onvifmetadatacombiner: Do not classify as Muxer
It confuses `encodebin` and technically it is not really a muxer so
as agreed on IRC, I am proposing to remove that classification.
2022-09-09 10:01:12 +03:00
Thibault Saunier
664e2b75bd tsjitterbuffer: Fix latency type when getting property 2022-09-02 21:41:35 +00:00
Taruntej Kanakamalla
67e9ba8286 whipsink: A GstBin implementation for WHIP
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410

Created a new plugin 'webrtchttp' to implement all the
WebRTC HTTP protocols under /net/webrtc-http directory.

WhipSink wraps around 'webrtcbin' with HTTP capabilites
to exchange SDP offer/answer so an ICE/DTLS session can
be established between the encoder/media producer (WHIP client)
and the broadcasting ingestion endpoint (Media Server).

Once the ICE/DTLS session is set up, the media will
flow unidirectionally from the WHIP client to the
broadcasting ingestion endpoint (Media Server).
Spec:
https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
2022-09-03 00:18:59 +03:00
Sebastian Dröge
420f36251a onvif: Rename onvif(de)pay to rtponvifmetadata(de)pay and include the metadata specifier in the other element names too
This is more descriptive and avoids any future conflicts with other
kinds of ONVIF specific RTP (de)payloaders.
2022-08-31 13:00:53 +03:00
Thibault Saunier
16d804e761 doc: Mark request::user-agent as doc show default 2022-08-29 18:33:22 -04:00
Thibault Saunier
31a53bba8a Generate plugins documentation using hotdoc
Which will automatically be integrated in gstreamer documentation
2022-08-29 18:33:22 -04:00