Commit graph

564 commits

Author SHA1 Message Date
Sebastian Dröge c3ced8c7e6 Update to AWS SDK 1.0 / 0.60 / 0.39
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1397>
2023-11-21 10:32:59 +02:00
Sebastian Dröge 1d9c89e3fe Update to AWS SDK 0.101 / 0.59 / 0.38 2023-11-20 10:13:13 +02:00
Sebastian Dröge 66c62d69b9 aws: Stop using deprecated aws_config function in the test 2023-11-18 10:16:24 +02:00
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 2d3d03b4d3 net/webrtc: rename WhipSignaller as WhipClientSignaller
remove generalized names to accommodate for the WhipServer
- name the Signaller for whipsink as WhipClient
- name the Settings for whipsink as WhipClientSettings
- name the State for whipsink as WhipClientState

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
Sebastian Dröge dee27e35b7 Update to latest AWS SDK
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1395>
2023-11-17 11:22:29 +02:00
Sebastian Dröge 58723f2a8c Update to AWS SDK 0.36
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1394>
2023-11-15 17:20:58 +02:00
François Laignel 9250c592a7 ndi: don't accumulate meta with audio only streams
Currently, only closed caption metadata are supported. When the next video
frame is received, pending meta are dequeued and parsed. If close captions
are found, they are attached to the video frame.

For audio only streams, it doesn't make sense to enqueue metadata. They would
accumulate in `pending_metadata` and would never be dequeued.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/460

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1392>
2023-11-13 19:26:23 +01:00
Sebastian Dröge 39155ef81c ndisrc: Implement zerocopy handling for the received frames if possible
Also move processing from the capture thread to the streaming thread.
The NDI SDK can cause frame drops if not reading fast enough from it.

All frame processing is now handled inside the ndisrcdemux.

Also use a buffer pool for video if copying is necessary.

Additionally, make sure to use different stream ids in the stream-start
event for the audio and video pad.

This plugin now requires GStreamer 1.16 or newer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 13:22:48 +02:00
Sebastian Dröge 2afffb39dd ndi: Don't mark private type as public
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 10:29:25 +02:00
Sebastian Dröge 99d7cce0d6 ndi: Refactor frame structs to have static lifetimes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 10:29:25 +02:00
Sebastian Dröge eb137ec6dc ndi: Remove wrong Clone impl on RecvInstance
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 10:29:25 +02:00
Arun Raghavan 771741c10c Revert "s3: tests: Remove emoji-based tests for now"
This reverts commit a49a5dcb11.

Now that hotdoc should work with emoji, let's bring the tests back.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1386>
2023-11-09 11:50:53 -05:00
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
Mathieu Duponchelle 5371eb52ad Port to AWS SDK 0.57/0.35
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
2023-11-03 15:13:45 +00:00
Sebastian Dröge f7745a336f aws: Update to test-with 0.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
2023-11-03 15:13:45 +00:00
Sebastian Dröge 16b917abb1 Update for gst::Rank API changes 2023-11-02 14:10:59 +02:00
Piotr Brzeziński 436b6d8efb gstwebrtc-api: Patch webrtc-adapter to fix Safari behaviour
There's currently a Safari-side bug causing webrtc-adapter to be unable to correctly shim the empty-candidate scenario
which we're using. This patch is very much a workaround and should be removed as soon as Safari and/or webrtc-adapter
fixes this on their side.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/439
https://github.com/webrtcHacks/adapter/issues/1140

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1377>
2023-10-30 16:36:11 +00:00
Sebastian Dröge 16c00ae3f5 Set sync=false in rsfilesink / s3sink
BaseSink defaults to sync=true and that doesn't make much sense for
these elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1376>
2023-10-30 17:38:46 +02:00
Sebastian Dröge 855b03a9ea Use let-else instead of match for weak reference upgrades
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1375>
2023-10-30 11:34:35 +02:00
Sebastian Dröge 557b249e11 Update to AWS SDK 0.34 and tracing-log 0.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1374>
2023-10-27 10:19:15 +03:00
Arun Raghavan d27a04e067 hlssink3: Close the playlist giostreamsink on stop if possible
This is a property that will be available from GStreamer 1.24, and will
ensure that we are able to flush the playlist during the READY->NULL
transition instead of when the element is freed.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/423
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1360>
2023-10-24 21:03:14 +00:00
Arun Raghavan a49a5dcb11 s3: tests: Remove emoji-based tests for now
These break hotdoc, which we need to fix first.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1333>
2023-10-24 12:52:12 -04:00
Arun Raghavan bb26e04a55 aws: s3: Properly percent-decode GstS3Url
We previously only percent-decoded the first fragment. This doesn't
necessarily harm anything, but for consistency we keep the structure
un-encoded, and encode when converting to a string representation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1333>
2023-10-24 12:52:12 -04:00
Arun Raghavan 51129febeb aws: s3sink: Fix handling of special characters in key
Properly URL-encode the string if needed, and add some tests for a
couple of cases.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/431
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1333>
2023-10-24 12:52:12 -04:00
Sebastian Dröge 829469d0fe rtpav1depay: Don't push stale temporal delimiters downstream
Only push them downstream once a complete OBU was assembled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1367>
2023-10-24 11:13:35 +00:00
Sebastian Dröge 1f5e9a9335 rtpav1depay: Skip unexpected leading fragments
If a packet is starting with a leading fragment but we do not expect to
receive one, then skip over it to the next OBU.

Not doing so would cause parsing of the middle of an OBU, which would
most likely fail and cause unnecessary warning messages about a
corrupted stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1367>
2023-10-24 11:13:35 +00:00
Sebastian Dröge 73ff822d24 Update to quick-xml 0.31
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1368>
2023-10-24 09:55:50 +03:00
Jordan Petridis a2d7f42138 Fix compilation after glib bindings changes
loggable_error! can now expand variables and we no longer need
the format! on our side.

https://github.com/gtk-rs/gtk-rs-core/pull/1210

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1366>
2023-10-22 01:20:56 +03:00
Sebastian Dröge 2ce04c6a78 webrtc: Update to livekit 0.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1293>
2023-10-18 10:30:59 +03:00
Sebastian Dröge d468e1e4a6 Clean up usage of pad probes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1361>
2023-10-17 08:44:06 +03:00
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
François Laignel 785c9557c8 net/webrtcsink: drop State lock before calling set-local-description
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1325>
2023-10-12 15:45:58 +00:00
François Laignel c021e2b69f net/webrtcsink: don't miss ice candidates
During `on_remote_description_set()` processing, current session is removed
from the sessions `HashMap`. If an ice candidate is submitted to `handle_ice()`
by that time, the session can't be found and the candidate is ignored.

This commit wraps the Session in the sessions `HashMap` so an entry is kept
while `on_remote_description_set()` is running. Incoming candidates received by
`handle_ice()` will be processed immediately or enqueued and handled when the
session is restored by `on_remote_description_set()`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1325>
2023-10-12 15:45:58 +00:00
Sebastian Dröge 42008fb895 aws: Update to test-with 0.11
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1358>
2023-10-12 06:57:28 +00:00
Lieven Paulissen 05aa9fa431 ndisrc: Assume input with more than 8 raw audio channels is unpositioned
gst_audio_channel_positions_from_mask() will otherwise print warnings
all the time.

Fixes #444

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1357>
2023-10-12 09:12:02 +03:00
François Laignel 022afa6375 ndi: use v210 encoding for cc and attach to video frame
The NDI closed captions specifications [1] define a variation where metadata is
attached to the video frame. This requires the AFD buffer to be v210 encoded.
This commit applies this strategy.

Another difference with previous version is that when an error occurs while
encoding or decoding a meta, next meta are also tried instead of failing
immediately.

Receiving closed captions as a standalone metadata is kept for interoperability
purposes. In this case, metadata is also expected to be v210 encoded.

[1]: http://www.sienna-tv.com/ndi/ndiclosedcaptions.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1356>
2023-10-11 21:25:29 +02:00
Maksym Khomenko 5b03f7d7b0 webrtcsrc: use @watch instead of @to-owned
@to-owned increases refcount of the element, which prevents the object from proper destruction, as the initial refcount with ElementFactory::make is larger than 1.

Instead, use @watch to create a weak reference and unbind the closure automatically if the object gets destroyed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1355>
2023-10-11 11:54:51 +03:00
Sebastian Dröge 3fc6220009 Update to AWS SDK 0.33
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1354>
2023-10-09 11:28:05 +03:00
Taruntej Kanakamalla 245185d2f6 net/webrtc/whip_signaller: Use the correct URL during redirect
Copy of 90e06dc3 for whipclientsink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1351>
2023-10-06 13:11:46 +00:00
Maksym Khomenko e4096b5157 webrtcsink: README: add documentation for custom signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1340>
2023-10-06 12:58:04 +03:00
Maksym Khomenko a9719cada2 webrtcsink: add custom signaller example
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1340>
2023-10-06 12:58:03 +03:00
Sebastian Dröge 1c4833bc5d Update to AWS SDK 0.32
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1352>
2023-10-06 09:11:17 +03:00
Sebastian Dröge 4569b7eca6 Fix various new 1.73 clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1347>
2023-10-03 17:47:30 +03:00
Sebastian Dröge 450ffbe452 Update for VideoFrame / GLVideoFrame API changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1345>
2023-10-02 13:25:25 +03:00
Piotr Brzeziński fe4273ca2a webrtc: Fix paths in README
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1342>
2023-09-29 17:05:29 +02:00
Sean DuBois 90e06dc37b net: webrtc/webrtchttp: Respect HTTP redirects
Properly follow redirect URL. Before new request would be made, but with
original URL again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1335>
2023-09-26 19:29:41 -04:00
Seungha Yang 22cc8c4986 hlssink3: Update README
Mention newly added hlscmafsink element and new properties

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-25 21:34:05 +09:00