mirrors/gst-plugins-rs
mirrors/gst-plugins-rs
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git synced 2024-11-29 15:01:07 +00:00
Code Issues Projects Releases Wiki Activity
2970 commits 14 branches 100 tags 46 MiB
Commit graph

594 commits

Author SHA1 Message Date
Taruntej Kanakamalla
50e905fe4b webrtc: conditional compile for features with 1_22 dependency 
Few features being used in webrtcsink like
the signal `request-aux-sender` are introduced
to webrtcbin in gstreamer release 1.22.

Rename the feature gst1_22 to v1_22 for uniformity.

Add v1_22 to default features.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1241>
 2024-02-01 15:08:11 +05:30
Sebastian Dröge
f2a7a34abf rtp: gcc: Use x += ... instead of x = x + ... 2024-01-31 18:46:55 +02:00
Sebastian Dröge
4ad101b53b Use once_cell crate directly again 
The glib crate does not depend on it anymore and also does not re-export
it anymore.

Also switch some usages of OnceCell to OnceLock from std.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
 2024-01-31 18:07:57 +02:00
Sebastian Dröge
451d928026 webrtc: Update AWS signaller to http 1 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
 2024-01-31 18:07:57 +02:00
Sanchayan Maity
95c007953c webrtchttp: Allow audio or video caps to be specified as None with WHEP 
We were setting audio and video caps by default even when the user
might have requested only video or audio. This would then result
in a `Could not reuse transceiver` error from the webrtcbin.

Fix this by allowing the user to specify audio or video caps as
None. This allows us to maintain the earlier behaviour for backward
compatibility while allowing the user to not request audio or video
as need be.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1433>
 2024-01-18 15:43:19 +05:30
Sebastian Dröge
764143d971 webrtc: Remove unnecessary manual Send+Sync implementations for signallers 
These are automatically implemented.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/483

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
 2024-01-18 10:01:25 +02:00
Sebastian Dröge
1af18f3028 webrtc: Require Send+Sync for signaller implementations 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
 2024-01-18 10:01:01 +02:00
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
773ebc7854 webrtcsrc: don't restrict RTP extensions to TWCC only 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1381>
 2024-01-17 07:34:01 +00:00
Sebastian Dröge
dfa95d8ed3 webrtc: Update to livekit-api / livekit-protocol 0.3 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
 2024-01-16 07:52:48 +00:00
Maksym Khomenko
fecbe01e06 webrtcsink: make 'extensions' property usage conditional 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1423>
 2024-01-16 07:13:56 +00:00
Sebastian Dröge
73a53e38c4 aws: s3: Disable remaining tests too for now 
They fail state changes, which cases `GstHarness` to abort.
 2024-01-16 09:13:41 +02:00
Arun Raghavan
fd3675aac0 aws: s3: Temporarily disable putobject tests 
Disabling while we figure out why it's failing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1429>
 2024-01-15 21:43:25 -05:00
Arun Raghavan
8b18ca15b5 Revert "aws: Disable putobjectsink tests for now" 
This reverts commit b128d127c2.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
 2024-01-11 15:38:36 -05:00
Arun Raghavan
06213714c5 aws: putobjectsink: Fix a couple of minor log typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
 2024-01-11 15:38:36 -05:00
Nirbheek Chauhan
2d85048925 webrtc/signalling: We get the address when accepting 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
 2023-12-29 13:28:48 +00:00
Nirbheek Chauhan
63b568f4a0 webrtc/signalling: Fix potential hang and FD leak 
If a peer connects via TCP and never initiates TLS, then the server
will get stuck in the accept loop. Spawn a task when accepting a TLS
connection, and timeout if it doesn't complete in 5 seconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
 2023-12-29 13:28:48 +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
b128d127c2 aws: Disable putobjectsink tests for now 
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
 2023-12-22 13:25:12 +02:00
Arun Raghavan
6d47045a60 aws: s3sink: Fix spelling of debug category 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
 2023-12-18 16:13:48 -05:00
Arun Raghavan
410d104ad6 aws: s3putobjectsink: Add a flush-on-error property 
Makes sure we can send out data even if the pipeline shutdown in error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
 2023-12-18 16:13:48 -05:00
Arun Raghavan
12dbf50ddc aws: s3putobjectsink: Add some thresholds for flushing 
Lets us connect when we perform a flush

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
 2023-12-18 16:13:48 -05:00
Arun Raghavan
a54b2dd39e aws: s3: Add a new awss3putobjectsink 
When streaming small amounts of data, using awss3sink might not be a
good idea, as we need to accumulate at least 5 MB of data for a
multipart upload (or we flush on EOS).

The alternative, while inefficient, is to do a complete PutObject of
_all_ the data periodically so as to not lose data in case of a pipeline
failure. This element makes a start on this idea by doing a PutObject
for every buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
 2023-12-18 10:39:23 -05:00
Sebastian Dröge
81dd45c814 webrtc: Downgrade aws-smithy-http to 0.60 
Version 0.61 was yanked from crates.io.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1407>
 2023-12-14 09:11:07 +02:00
Sebastian Dröge
2f2bf6ca8f webrtc: Update to aws-smithy-http 0.61 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
 2023-12-09 12:21:38 +02:00
Sebastian Dröge
0bae18fe0d rtp: Update to bitstream-io 2.0 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
 2023-12-09 12:17:51 +02:00
Sebastian Dröge
181bd13103 Update to async-tungstenite 0.24 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
 2023-12-09 12:17:11 +02:00
Guillaume Desmottes
6dfd1c1496 use new debug and parse API 
Changes from https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1355

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1403>
 2023-12-04 15:58:21 +01:00
Sebastian Dröge
f13574d8ed Update further AWS SDK crates to 1.0 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1400>
 2023-11-26 10:26:02 +02:00
Mathieu Duponchelle
cf1c7600a2 webrtcsink: don't panic on failure to request pad from webrtcbin 
webrtcbin will refuse pad requests for all sorts of reasons, and should
be logging an error when doing so, simply post an error message and let
the application deal with it, the reason for the refusal should
hopefully be available in the logs to the user.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1399>
 2023-11-24 19:53:38 +01:00
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