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Author SHA1 Message Date
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
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