Commit graph

58 commits

Author SHA1 Message Date
Stéphane Cerveau ac3ca76c9f 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/1387>
2023-11-10 17:47:41 +02:00
Maksym Khomenko 19597b3737 webrtcsrc: add turn-servers property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1387>
2023-11-10 17:46:28 +02:00
Sebastian Dröge 755f021a30 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/1328>
2023-09-20 13:52:34 +03:00
Seungha Yang 581787f651 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/1327>
2023-09-20 11:45:05 +03:00
Guillaume Desmottes 6f75243c8f 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/1249>
2023-06-19 19:02:33 +03:00
Sebastian Dröge c9b2c88469 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/1249>
2023-06-19 19:01:52 +03:00
Guillaume Desmottes 015edb7d37 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/1216>
2023-05-18 18:28:24 +03:00
Arun Raghavan 5462ed0e95 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/1190>
2023-04-22 12:10:28 +03:00
Thibault Saunier 4b867d27fe Add a webrtcsrc element
Updating the docker image to include:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3236

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1117>
2023-03-02 14:56:30 -03:00
rajneeshksoni 01d3b0f9da awss3sink: Add properties to set content-Type and content-disposition.
for uploaded object default content-type is set to binary/octet-stream,
which is correct.
metadata cannot be used to set content-type and content-disposition as
setting metadata add a prefix x-amz-meta to key
e.g. setting metadate "content-type=video/mp4" actually set value as
x-amz-meta-content-type. So these has to be seaprate property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
2023-02-09 21:43:57 +02:00
Simon Himmelbauer fffecca624 spotifyaudiosrc: Support configurable bitrate
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
2023-02-09 21:43:57 +02:00
rajneeshksoni f96b64e1c1 hlssink3: Allow setting i-frame-only playlist.
HLS allows manifest where all segments are single ifames.
This manifest requires `EXT-X-I-FRAMES-ONLY` tag in the
manifest.
I-FRAMES-ONLY playlist segments are video only segments.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
2023-02-09 21:43:57 +02:00
Sebastian Dröge dae1d8b5ef mp4/fmp4: Update docs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
2023-02-09 21:17:34 +02:00
Sebastian Dröge 17dec1cb26 rtpav1pay: Add support for tu/frame aligned input
In this case every buffer can be sent out immediately and makes up a
whole frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
2023-02-09 21:13:40 +02:00
Sebastian Dröge 3520fc67de rtpav1depay: Don't output full TUs but just OBUs as they come
Simplifies state tracking and potentially reduces latency as it's not
necessary to wait until all fragments of an OBU are received.

The last OBU of a TU is marked with the marker flag to allow parsers to
detect this without first seeing the beginning of the next TU.

Also use a simple `Vec` for collecting complete OBUs instead of a
`gst_base::Adapter` as this reduces the number of allocations.

And also handle invalid packets a little bit more gracefully.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/244

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
2023-02-09 21:13:06 +02:00
Sebastian Dröge a01437b675 fmp4mux: Add support for sub-fragments / chunking
Allow outputting sub-fragments (chunks in CMAF terms) that are shorter
than the fragment duration and don't usually start on a keyframe. By
this the buffering requirements of the element is reduced to one chunk
duration, as is the latency.

This is used for formats like low-latency / LL-HLS and DASH.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
2023-02-09 21:06:56 +02:00
Guillaume Desmottes b9e203d6c1 fmp4mux: add 'offset-to-zero' property
Add it only to 'isofmp4mux', the onvif variant already does this and
CMAF and DASH are always single-stream so you rely on inter-container
synchronization via the running-time.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1086>
2023-02-09 21:06:08 +02:00
Guillaume Desmottes c6158b7a4e livesync: fix late-threshold property min value
The code is handling 0 as "always over threshold" but it was not
possible to set the property to 0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1050>
2023-01-18 16:56:34 +02:00
rajneeshksoni 698ab100b3 awss3hlssink: Add stats property.
application can monitor the progress of hls segment generation
and upload progress.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1038>
2023-01-07 13:25:31 +02:00
Nirbheek Chauhan 8cdb30bd39 meson: Add options for all plugins
Required a slight rework of the build file, and how options are passed
to cargo_wrapper.py

Necessitated a bump of the required gstreamer version to 1.20, which
should be fine for the meson build since its primary function is to be
built as part of the gstreamer monorepo build to get a dev env.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1038>
2023-01-07 13:23:41 +02:00
Jan Alexander Steffens (heftig) 6596b6cdd1 Add livesync plugin
It attempts to produce a (nearly) gapless live stream by synchronizing
its output to the running time and forwarding the next input buffer if
its start is (nearly) flush with the end of the last output buffer.

If the input buffer is missing or too far in the future, it duplicates
the last output buffer with adjusted timestamps. If it is operating on a
raw audio stream, it will fill duplicate buffers with silence.

If an input buffer arrives too late, it is thrown away. If the last
input buffer was accepted too long ago (according to `late-threshold`),
a late input buffer is accepted anyway, but immediately considered a
duplicate. Due to the silence-filling, this has no effect on audio, but
video gets a "slideshow" effect instead of freezing completely.

The "many-repeats" property will be notified when this element has
recently duplicated a lot of buffers or recovered from such a state.

Co-authored-by: Vivia Nikolaidou <vivia@ahiru.eu>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1017>
2022-12-16 18:51:22 +02:00
Michiel Konstapel a1fd847f70 audiornnoise: add voice detection threshold
Add a property "voice-activity-threshold". Frames where the voice
detection score from the RNN is below the threshold will be completely
muted.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
2022-12-12 13:40:01 +02:00
Guillaume Desmottes b3e33e329b textahead: add settings to display previous buffers
I'll use this in Karapulse to keep displaying the few previous lyrics
rather than having them disappear right away.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
2022-12-12 13:39:44 +02:00
Sebastian Dröge d79edce517 webrtchttp: Fix documentation JSON
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
2022-12-12 13:39:21 +02:00
Sanchayan Maity 3fc0326084 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/1008>
2022-12-12 13:37:43 +02:00
Sanchayan Maity 2e529fa152 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/1008>
2022-12-12 13:37:13 +02:00
Seungha Yang 506c96e8aa 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/1008>
2022-12-12 13:36:43 +02:00
Jordan Petridis 507377c052 video/gtk4: Implement support for GLTextures when possible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1008>
2022-12-12 13:34:22 +02:00
Sebastian Dröge 516b561191 mp4: Add ONVIF non-fragmented MP4 muxer
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/987>
2022-11-28 10:47:02 +02:00
Sebastian Dröge e7f5e73e3f gst-plugin-mp4: Add new MP4 plugin with a non-fragmented MP4 muxer
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/987>
2022-11-28 10:47:02 +02:00
François Laignel 2e52fece61 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`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/977>
2022-11-12 15:52:50 +00:00
Sebastian Dröge 43ac186e69 fmp4mux: Make media/trak timescales configurable
And refactor a bit of code for easier extensibility.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/977>
2022-11-12 15:52:50 +00:00
Sebastian Dröge cb5a956ee7 fmp4mux: Add initial Opus support
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/239

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/977>
2022-11-12 15:52:49 +00:00
Sebastian Dröge d0228ed544 docs: Remove some stale entries of renamed elements
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/977>
2022-11-12 15:52:49 +00:00
Matthew Waters a54318fbb4 fmp4: add support for muxing VP9 streams in cmaf, dash and iso fmp4
As specified in https://www.webmproject.org/vp9/mp4/
2022-10-27 15:34:32 +03:00
Sebastian Dröge fe8e0a8bf8 Update docs 2022-10-23 21:29:14 +03:00
Rafael Caricio 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
Rafael Caricio 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